mirror of
https://gitlab.crans.org/nounous/ghostream.git
synced 2024-12-22 10:22:19 +00:00
Proper multi client WebRTC
This commit is contained in:
Alexandre Iooss
parent
ef760ae4cc
commit
6b5fea66e6
237
stream/stream.go
237
stream/stream.go
@ -1,7 +1,6 @@
|
||||
package stream
|
||||
|
||||
import (
|
||||
"context"
|
||||
"fmt"
|
||||
"io"
|
||||
"log"
|
||||
@ -21,13 +20,32 @@ const (
|
||||
)
|
||||
|
||||
var (
|
||||
iceConnectedCtx, iceConnectedCtxCancel = context.WithCancel(context.Background())
|
||||
videoTracks []*webrtc.Track
|
||||
audioTracks []*webrtc.Track
|
||||
)
|
||||
|
||||
// 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
|
||||
}
|
||||
|
||||
// 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 {
|
||||
func newPeerHandler(remoteSdp webrtc.SessionDescription) webrtc.SessionDescription {
|
||||
// Create media engine using client SDP
|
||||
mediaEngine := webrtc.MediaEngine{}
|
||||
if err := mediaEngine.PopulateFromSDP(remoteSdp); err != nil {
|
||||
log.Println("Failed to create new media engine", err)
|
||||
return webrtc.SessionDescription{}
|
||||
}
|
||||
|
||||
// Create a new PeerConnection
|
||||
api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
|
||||
peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
|
||||
ICEServers: []webrtc.ICEServer{
|
||||
{
|
||||
@ -40,18 +58,11 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
|
||||
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)
|
||||
// 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)
|
||||
return webrtc.SessionDescription{}
|
||||
}
|
||||
if _, err = peerConnection.AddTrack(videoTrack); err != nil {
|
||||
@ -59,6 +70,18 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
|
||||
return webrtc.SessionDescription{}
|
||||
}
|
||||
|
||||
// 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{}
|
||||
}
|
||||
|
||||
// Set the remote SessionDescription
|
||||
if err = peerConnection.SetRemoteDescription(remoteSdp); err != nil {
|
||||
log.Println("Failed to set remote description", err)
|
||||
@ -81,6 +104,21 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
|
||||
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 {
|
||||
// Register tracks
|
||||
videoTracks = append(videoTracks, videoTrack)
|
||||
audioTracks = append(audioTracks, audioTrack)
|
||||
} else if connectionState == webrtc.ICEConnectionStateDisconnected {
|
||||
// Unregister tracks
|
||||
videoTracks = removeTrack(videoTracks, videoTrack)
|
||||
audioTracks = removeTrack(audioTracks, audioTrack)
|
||||
}
|
||||
})
|
||||
|
||||
// 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
|
||||
@ -90,130 +128,81 @@ func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioT
|
||||
return *peerConnection.LocalDescription()
|
||||
}
|
||||
|
||||
// Serve WebRTC media streaming server
|
||||
func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
|
||||
// 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 {
|
||||
panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
|
||||
func playVideo() {
|
||||
// Open a IVF file and start reading using our IVFReader
|
||||
file, ivfErr := os.Open(videoFileName)
|
||||
if ivfErr != nil {
|
||||
panic(ivfErr)
|
||||
}
|
||||
|
||||
// Create media engine
|
||||
// Only support VP8 and Opus
|
||||
mediaEngine := webrtc.MediaEngine{}
|
||||
offer := <-remoteSdpChan
|
||||
if err = mediaEngine.PopulateFromSDP(offer); err != nil {
|
||||
panic(err)
|
||||
ivf, header, ivfErr := ivfreader.NewWith(file)
|
||||
if ivfErr != nil {
|
||||
panic(ivfErr)
|
||||
}
|
||||
|
||||
// Create a new API object
|
||||
api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
|
||||
// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
|
||||
// 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")
|
||||
os.Exit(0)
|
||||
}
|
||||
|
||||
// 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)
|
||||
}
|
||||
|
||||
// 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)
|
||||
if ivfErr != nil {
|
||||
panic(ivfErr)
|
||||
}
|
||||
|
||||
ivf, header, ivfErr := ivfreader.NewWith(file)
|
||||
if ivfErr != nil {
|
||||
panic(ivfErr)
|
||||
}
|
||||
|
||||
// Wait for connection established
|
||||
<-iceConnectedCtx.Done()
|
||||
|
||||
// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
|
||||
// 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")
|
||||
os.Exit(0)
|
||||
}
|
||||
|
||||
if ivfErr != nil {
|
||||
panic(ivfErr)
|
||||
}
|
||||
|
||||
time.Sleep(sleepTime)
|
||||
if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
|
||||
time.Sleep(sleepTime)
|
||||
for _, t := range videoTracks {
|
||||
if ivfErr = t.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
|
||||
log.Fatalln("Failed to write video stream:", ivfErr)
|
||||
}
|
||||
}
|
||||
}()
|
||||
}
|
||||
}
|
||||
|
||||
func playAudio() {
|
||||
// Open a IVF file and start reading using our IVFReader
|
||||
file, oggErr := os.Open(audioFileName)
|
||||
if oggErr != nil {
|
||||
panic(oggErr)
|
||||
}
|
||||
|
||||
// Open on oggfile in non-checksum mode.
|
||||
ogg, _, oggErr := oggreader.NewWith(file)
|
||||
if oggErr != nil {
|
||||
panic(oggErr)
|
||||
}
|
||||
|
||||
// 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")
|
||||
os.Exit(0)
|
||||
}
|
||||
|
||||
go func() {
|
||||
// Open a IVF file and start reading using our IVFReader
|
||||
file, oggErr := os.Open(audioFileName)
|
||||
if oggErr != nil {
|
||||
panic(oggErr)
|
||||
}
|
||||
|
||||
// Open on oggfile in non-checksum mode.
|
||||
ogg, _, oggErr := oggreader.NewWith(file)
|
||||
if oggErr != nil {
|
||||
panic(oggErr)
|
||||
}
|
||||
// The amount of samples is the difference between the last and current timestamp
|
||||
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
|
||||
lastGranule = pageHeader.GranulePosition
|
||||
|
||||
// Wait for connection established
|
||||
<-iceConnectedCtx.Done()
|
||||
|
||||
// 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")
|
||||
os.Exit(0)
|
||||
}
|
||||
|
||||
if oggErr != nil {
|
||||
panic(oggErr)
|
||||
}
|
||||
|
||||
// The amount of samples is the difference between the last and current timestamp
|
||||
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
|
||||
lastGranule = pageHeader.GranulePosition
|
||||
|
||||
if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
|
||||
for _, t := range audioTracks {
|
||||
if oggErr = t.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
|
||||
log.Fatalln("Failed to write audio stream:", oggErr)
|
||||
}
|
||||
|
||||
// Convert seconds to Milliseconds, Sleep doesn't accept floats
|
||||
time.Sleep(time.Duration((sampleCount/48000)*1000) * time.Millisecond)
|
||||
}
|
||||
}()
|
||||
|
||||
// Handle new connections
|
||||
for {
|
||||
// Wait for incoming session description
|
||||
// then send the local description to browser
|
||||
offer := <-remoteSdpChan
|
||||
localSdpChan <- newPeerHandler(api, offer, audioTrack, videoTrack)
|
||||
// Convert seconds to Milliseconds, Sleep doesn't accept floats
|
||||
time.Sleep(time.Duration((sampleCount/48000)*1000) * time.Millisecond)
|
||||
}
|
||||
}
|
||||
|
||||
@ -228,3 +217,17 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
|
||||
}
|
||||
panic(fmt.Sprintf("Remote peer does not support %s", codecName))
|
||||
}
|
||||
|
||||
// Serve WebRTC media streaming server
|
||||
func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
|
||||
go playVideo()
|
||||
go playAudio()
|
||||
|
||||
// Handle new connections
|
||||
for {
|
||||
// Wait for incoming session description
|
||||
// then send the local description to browser
|
||||
offer := <-remoteSdpChan
|
||||
localSdpChan <- newPeerHandler(offer)
|
||||
}
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user