Files
webrtc/examples/sfu-ws/room.go
Sean DuBois 1202dbaa06 Migrate SDP generation to Unified Plan
This commit has breaking changes. This API change means we
can no longer support an arbitrary number of receivers. For every track
you want to receive you MUST call PeerConnection.AddTransceiver

We do now support sending an multiple audio/video feeds. You can see
this behavior via gstreamer-receive and gstreamer-send currently.

Resolves #54
2019-04-04 12:55:36 -07:00

211 lines
5.0 KiB
Go

package main
import (
"io"
"net/http"
"sync"
"sync/atomic"
"time"
"github.com/gorilla/websocket"
"github.com/pions/rtcp"
"github.com/pions/webrtc"
)
// Peer config
var peerConnectionConfig = webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
}
var (
// Media engine
m webrtc.MediaEngine
// API object
api *webrtc.API
// Publisher Peer
pubCount int32
pubReceiver *webrtc.PeerConnection
// Local track
videoTrack *webrtc.Track
audioTrack *webrtc.Track
videoTrackLock = sync.RWMutex{}
audioTrackLock = sync.RWMutex{}
// Websocket upgrader
upgrader = websocket.Upgrader{}
// Broadcast channels
broadcastHub = newHub()
)
const (
rtcpPLIInterval = time.Second * 3
)
func room(w http.ResponseWriter, r *http.Request) {
// Websocket client
c, err := upgrader.Upgrade(w, r, nil)
checkError(err)
defer func() {
checkError(c.Close())
}()
// Read sdp from websocket
mt, msg, err := c.ReadMessage()
checkError(err)
if atomic.LoadInt32(&pubCount) == 0 {
atomic.AddInt32(&pubCount, 1)
// Create a new RTCPeerConnection
pubReceiver, err = api.NewPeerConnection(peerConnectionConfig)
checkError(err)
_, err = pubReceiver.AddTransceiver(webrtc.RTPCodecTypeAudio)
checkError(err)
_, err = pubReceiver.AddTransceiver(webrtc.RTPCodecTypeVideo)
checkError(err)
pubReceiver.OnTrack(func(remoteTrack *webrtc.Track, receiver *webrtc.RTPReceiver) {
if remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeVP8 || remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeVP9 || remoteTrack.PayloadType() == webrtc.DefaultPayloadTypeH264 {
// Create a local video track, all our SFU clients will be fed via this track
var err error
videoTrackLock.Lock()
videoTrack, err = pubReceiver.NewTrack(remoteTrack.PayloadType(), remoteTrack.SSRC(), "video", "pion")
videoTrackLock.Unlock()
checkError(err)
// Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
go func() {
ticker := time.NewTicker(rtcpPLIInterval)
for range ticker.C {
checkError(pubReceiver.WriteRTCP(&rtcp.PictureLossIndication{MediaSSRC: videoTrack.SSRC()}))
}
}()
rtpBuf := make([]byte, 1400)
for {
i, err := remoteTrack.Read(rtpBuf)
checkError(err)
videoTrackLock.RLock()
_, err = videoTrack.Write(rtpBuf[:i])
videoTrackLock.RUnlock()
if err != io.ErrClosedPipe {
checkError(err)
}
}
} else {
// Create a local audio track, all our SFU clients will be fed via this track
var err error
audioTrackLock.Lock()
audioTrack, err = pubReceiver.NewTrack(remoteTrack.PayloadType(), remoteTrack.SSRC(), "audio", "pion")
audioTrackLock.Unlock()
checkError(err)
rtpBuf := make([]byte, 1400)
for {
i, err := remoteTrack.Read(rtpBuf)
checkError(err)
audioTrackLock.RLock()
_, err = audioTrack.Write(rtpBuf[:i])
audioTrackLock.RUnlock()
if err != io.ErrClosedPipe {
checkError(err)
}
}
}
})
// Set the remote SessionDescription
checkError(pubReceiver.SetRemoteDescription(
webrtc.SessionDescription{
SDP: string(msg),
Type: webrtc.SDPTypeOffer,
}))
// Create answer
answer, err := pubReceiver.CreateAnswer(nil)
checkError(err)
// Sets the LocalDescription, and starts our UDP listeners
checkError(pubReceiver.SetLocalDescription(answer))
// Send server sdp to publisher
checkError(c.WriteMessage(mt, []byte(answer.SDP)))
// Register incoming channel
pubReceiver.OnDataChannel(func(d *webrtc.DataChannel) {
d.OnMessage(func(msg webrtc.DataChannelMessage) {
// Broadcast the data to subSenders
broadcastHub.broadcastChannel <- msg.Data
})
})
} else {
// Create a new PeerConnection
subSender, err := api.NewPeerConnection(peerConnectionConfig)
checkError(err)
// Register data channel creation handling
subSender.OnDataChannel(func(d *webrtc.DataChannel) {
broadcastHub.addListener(d)
})
// Waiting for publisher track finish
for {
videoTrackLock.RLock()
if videoTrack == nil {
videoTrackLock.RUnlock()
//if videoTrack == nil, waiting..
time.Sleep(100 * time.Millisecond)
} else {
videoTrackLock.RUnlock()
break
}
}
// Add local video track
videoTrackLock.RLock()
_, err = subSender.AddTrack(videoTrack)
videoTrackLock.RUnlock()
checkError(err)
// Add local audio track
audioTrackLock.RLock()
_, err = subSender.AddTrack(audioTrack)
audioTrackLock.RUnlock()
checkError(err)
// Set the remote SessionDescription
checkError(subSender.SetRemoteDescription(
webrtc.SessionDescription{
SDP: string(msg),
Type: webrtc.SDPTypeOffer,
}))
// Create answer
answer, err := subSender.CreateAnswer(nil)
checkError(err)
// Sets the LocalDescription, and starts our UDP listeners
checkError(subSender.SetLocalDescription(answer))
// Send server sdp to subscriber
checkError(c.WriteMessage(mt, []byte(answer.SDP)))
}
}