mirror of
https://gitlab.crans.org/nounous/ghostream.git
synced 2024-12-22 16:12:19 +00:00
Create first webrtc session like others
This commit is contained in:
Alexandre Iooss
parent
6a138d1b1a
commit
ef760ae4cc
110
stream/stream.go
110
stream/stream.go
@ -21,19 +21,34 @@ const (
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)
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var (
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peerConnectionConfig webrtc.Configuration
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iceConnectedCtx, iceConnectedCtxCancel = context.WithCancel(context.Background())
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)
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// newPeerHandler is called when server receive a new session description
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// this initiates a WebRTC connection and return server description
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func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioTrack *webrtc.Track, videoTrack *webrtc.Track) webrtc.SessionDescription {
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// Create a new PeerConnection
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peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
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peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{
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{
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URLs: []string{"stun:stun.l.google.com:19302"},
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},
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},
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})
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if err != nil {
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log.Println("Failed to initiate peer connection", err)
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return webrtc.SessionDescription{}
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}
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// Set the handler for ICE connection state
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// This will notify you when the peer has connected/disconnected
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peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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log.Printf("Connection State has changed %s \n", connectionState.String())
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if connectionState == webrtc.ICEConnectionStateConnected {
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iceConnectedCtxCancel()
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}
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})
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// Add audio and video tracks
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if _, err = peerConnection.AddTrack(audioTrack); err != nil {
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log.Println("Failed to add audio track", err)
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@ -77,48 +92,42 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
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// Serve WebRTC media streaming server
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func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
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peerConnectionConfig = webrtc.Configuration{
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ICEServers: []webrtc.ICEServer{
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{
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URLs: []string{"stun:stun.l.google.com:19302"},
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},
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},
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}
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// Assert that we have an audio or video file
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_, err := os.Stat(videoFileName)
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haveVideoFile := !os.IsNotExist(err)
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_, err = os.Stat(audioFileName)
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haveAudioFile := !os.IsNotExist(err)
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if !haveAudioFile && !haveVideoFile {
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if !haveAudioFile || !haveVideoFile {
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panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
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}
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// We make our own mediaEngine so we can place the sender's codecs in it. This because we must use the
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// dynamic media type from the sender in our answer. This is not required if we are the offerer
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// Create media engine
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// Only support VP8 and Opus
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mediaEngine := webrtc.MediaEngine{}
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offer := <-remoteSdpChan
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if err = mediaEngine.PopulateFromSDP(offer); err != nil {
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panic(err)
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}
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// Create a new RTCPeerConnection
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// Create a new API object
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api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
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peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
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// Create video track
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codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
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videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
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if err != nil {
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panic(err)
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}
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iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
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// Create a video track
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videoTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8"), rand.Uint32(), "video", "pion")
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if addTrackErr != nil {
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panic(addTrackErr)
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}
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if _, addTrackErr = peerConnection.AddTrack(videoTrack); addTrackErr != nil {
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panic(addTrackErr)
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// Create audio track
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codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
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audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
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if err != nil {
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panic(err)
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}
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localSdpChan <- newPeerHandler(api, offer, audioTrack, videoTrack)
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go func() {
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// Open a IVF file and start reading using our IVFReader
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file, ivfErr := os.Open(videoFileName)
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@ -138,6 +147,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
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sleepTime := time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000)
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for {
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// Need at least one client
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frame, _, ivfErr := ivf.ParseNextFrame()
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if ivfErr == io.EOF {
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fmt.Printf("All video frames parsed and sent")
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@ -150,20 +160,11 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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time.Sleep(sleepTime)
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if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
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panic(ivfErr)
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log.Fatalln("Failed to write video stream:", ivfErr)
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}
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}
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}()
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// Create a audio track
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audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), rand.Uint32(), "audio", "pion")
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if addTrackErr != nil {
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panic(addTrackErr)
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}
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if _, addTrackErr = peerConnection.AddTrack(audioTrack); addTrackErr != nil {
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panic(addTrackErr)
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}
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go func() {
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// Open a IVF file and start reading using our IVFReader
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file, oggErr := os.Open(audioFileName)
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@ -183,6 +184,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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// Keep track of last granule, the difference is the amount of samples in the buffer
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var lastGranule uint64
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for {
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// Need at least one client
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pageData, pageHeader, oggErr := ogg.ParseNextPage()
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if oggErr == io.EOF {
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fmt.Printf("All audio pages parsed and sent")
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@ -198,7 +200,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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lastGranule = pageHeader.GranulePosition
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if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
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panic(oggErr)
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log.Fatalln("Failed to write audio stream:", oggErr)
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}
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// Convert seconds to Milliseconds, Sleep doesn't accept floats
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@ -206,42 +208,6 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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}
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}()
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// Set the handler for ICE connection state
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// This will notify you when the peer has connected/disconnected
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peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
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fmt.Printf("Connection State has changed %s \n", connectionState.String())
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if connectionState == webrtc.ICEConnectionStateConnected {
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iceConnectedCtxCancel()
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}
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})
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// Set the remote SessionDescription
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if err = peerConnection.SetRemoteDescription(offer); err != nil {
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panic(err)
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}
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// Create answer
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answer, err := peerConnection.CreateAnswer(nil)
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if err != nil {
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panic(err)
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}
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// Create channel that is blocked until ICE Gathering is complete
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gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
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// Sets the LocalDescription, and starts our UDP listeners
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if err = peerConnection.SetLocalDescription(answer); err != nil {
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panic(err)
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}
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// Block until ICE Gathering is complete, disabling trickle ICE
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// we do this because we only can exchange one signaling message
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// in a production application you should exchange ICE Candidates via OnICECandidate
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<-gatherComplete
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// Output the answer
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localSdpChan <- *peerConnection.LocalDescription()
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// Handle new connections
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for {
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// Wait for incoming session description
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@ -254,10 +220,10 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
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// Search for Codec PayloadType
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//
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// Since we are answering we need to match the remote PayloadType
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func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) uint8 {
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func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) (*webrtc.RTPCodec, uint8) {
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for _, codec := range m.GetCodecsByKind(codecType) {
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if codec.Name == codecName {
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return codec.PayloadType
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return codec, codec.PayloadType
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}
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}
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panic(fmt.Sprintf("Remote peer does not support %s", codecName))
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