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ghostream
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Create first webrtc session like others

This commit is contained in:
Alexandre Iooss 2020-09-25 12:03:40 +02:00
parent 6a138d1b1a
commit ef760ae4cc
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GPG Key ID: 6C79278F3FCDCC02
1 changed files with 38 additions and 72 deletions
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110
stream/stream.go
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@ -21,19 +21,34 @@ const (
)
var (
peerConnectionConfig webrtc.Configuration
iceConnectedCtx, iceConnectedCtxCancel = context.WithCancel(context.Background())
)
// newPeerHandler is called when server receive a new session description
// this initiates a WebRTC connection and return server description
func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioTrack *webrtc.Track, videoTrack *webrtc.Track) webrtc.SessionDescription {
// Create a new PeerConnection
peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
})
if err != nil {
log.Println("Failed to initiate peer connection", err)
return webrtc.SessionDescription{}
}
// 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 connectionState == webrtc.ICEConnectionStateConnected {
iceConnectedCtxCancel()
}
})
// Add audio and video tracks
if _, err = peerConnection.AddTrack(audioTrack); err != nil {
log.Println("Failed to add audio track", err)
@ -77,48 +92,42 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
// Serve WebRTC media streaming server
func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
peerConnectionConfig = webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
}
// Assert that we have an audio or video file
_, err := os.Stat(videoFileName)
haveVideoFile := !os.IsNotExist(err)
_, err = os.Stat(audioFileName)
haveAudioFile := !os.IsNotExist(err)
if !haveAudioFile && !haveVideoFile {
if !haveAudioFile || !haveVideoFile {
panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
}
// We make our own mediaEngine so we can place the sender's codecs in it. This because we must use the
// dynamic media type from the sender in our answer. This is not required if we are the offerer
// Create media engine
// Only support VP8 and Opus
mediaEngine := webrtc.MediaEngine{}
offer := <-remoteSdpChan
if err = mediaEngine.PopulateFromSDP(offer); err != nil {
panic(err)
}
// Create a new RTCPeerConnection
// Create a new API object
api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
// Create video track
codec, payloadType := getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8")
videoTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "video", "pion", codec)
if err != nil {
panic(err)
}
iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
// Create a video track
videoTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8"), rand.Uint32(), "video", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(videoTrack); addTrackErr != nil {
panic(addTrackErr)
// Create audio track
codec, payloadType = getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus")
audioTrack, err := webrtc.NewTrack(payloadType, rand.Uint32(), "audio", "pion", codec)
if err != nil {
panic(err)
}
localSdpChan <- newPeerHandler(api, offer, audioTrack, videoTrack)
go func() {
// Open a IVF file and start reading using our IVFReader
file, ivfErr := os.Open(videoFileName)
@ -138,6 +147,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
sleepTime := time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000)
for {
// Need at least one client
frame, _, ivfErr := ivf.ParseNextFrame()
if ivfErr == io.EOF {
fmt.Printf("All video frames parsed and sent")
@ -150,20 +160,11 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
time.Sleep(sleepTime)
if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
panic(ivfErr)
log.Fatalln("Failed to write video stream:", ivfErr)
}
}
}()
// Create a audio track
audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), rand.Uint32(), "audio", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(audioTrack); addTrackErr != nil {
panic(addTrackErr)
}
go func() {
// Open a IVF file and start reading using our IVFReader
file, oggErr := os.Open(audioFileName)
@ -183,6 +184,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
// Keep track of last granule, the difference is the amount of samples in the buffer
var lastGranule uint64
for {
// Need at least one client
pageData, pageHeader, oggErr := ogg.ParseNextPage()
if oggErr == io.EOF {
fmt.Printf("All audio pages parsed and sent")
@ -198,7 +200,7 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
lastGranule = pageHeader.GranulePosition
if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
panic(oggErr)
log.Fatalln("Failed to write audio stream:", oggErr)
}
// Convert seconds to Milliseconds, Sleep doesn't accept floats
@ -206,42 +208,6 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
}
}()
// Set the handler for ICE connection state
// This will notify you when the peer has connected/disconnected
peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
fmt.Printf("Connection State has changed %s \n", connectionState.String())
if connectionState == webrtc.ICEConnectionStateConnected {
iceConnectedCtxCancel()
}
})
// Set the remote SessionDescription
if err = peerConnection.SetRemoteDescription(offer); err != nil {
panic(err)
}
// Create answer
answer, err := peerConnection.CreateAnswer(nil)
if err != nil {
panic(err)
}
// 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 {
panic(err)
}
// 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 answer
localSdpChan <- *peerConnection.LocalDescription()
// Handle new connections
for {
// Wait for incoming session description
@ -254,10 +220,10 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
// 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) uint8 {
func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) (*webrtc.RTPCodec, uint8) {
for _, codec := range m.GetCodecsByKind(codecType) {
if codec.Name == codecName {
return codec.PayloadType
return codec, codec.PayloadType
}
}
panic(fmt.Sprintf("Remote peer does not support %s", codecName))