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💩 Split webrtc tracks by stream id (need to clean this, stream ID must pass between the session descriptor and the webrtc flux transmit)

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This commit is contained in:
Yohann D'ANELLO
2020-10-05 22:00:08 +02:00
parent 76f009efe3
commit 022f6fb098
5 changed files with 87 additions and 30 deletions
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37
stream/webrtc/ingest.go
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@ -2,6 +2,7 @@ package webrtc
import (
"bufio"
"github.com/pion/webrtc/v3"
"io"
"log"
"net"
@ -18,10 +19,10 @@ func ingestFrom(inputChannel chan srt.Packet) {
for {
var err error = nil
packet := <-inputChannel
switch packet.PacketType {
srtPacket := <-inputChannel
switch srtPacket.PacketType {
case "register":
log.Printf("WebRTC RegisterStream %s", packet.StreamName)
log.Printf("WebRTC RegisterStream %s", srtPacket.StreamName)
// Open a UDP Listener for RTP Packets on port 5004
videoListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5004})
@ -74,13 +75,17 @@ func ingestFrom(inputChannel chan srt.Packet) {
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
log.Printf("Failed to unmarshal RTP packet: %s", err)
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
}
// Write RTP packet to all video tracks
if videoTracks[srtPacket.StreamName] == nil {
videoTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
}
// Write RTP srtPacket to all video tracks
// Adapt payload and SSRC to match destination
for _, videoTrack := range videoTracks {
for _, videoTrack := range videoTracks[srtPacket.StreamName] {
packet.Header.PayloadType = videoTrack.PayloadType()
packet.Header.SSRC = videoTrack.SSRC()
if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
@ -102,13 +107,17 @@ func ingestFrom(inputChannel chan srt.Packet) {
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
log.Printf("Failed to unmarshal RTP packet: %s", err)
log.Printf("Failed to unmarshal RTP srtPacket: %s", err)
continue
}
// Write RTP packet to all audio tracks
if audioTracks[srtPacket.StreamName] == nil {
audioTracks[srtPacket.StreamName] = make([]*webrtc.Track, 0)
}
// Write RTP srtPacket to all audio tracks
// Adapt payload and SSRC to match destination
for _, audioTrack := range audioTracks {
for _, audioTrack := range audioTracks[srtPacket.StreamName] {
packet.Header.PayloadType = audioTrack.PayloadType()
packet.Header.SSRC = audioTrack.SSRC()
if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
@ -127,20 +136,20 @@ func ingestFrom(inputChannel chan srt.Packet) {
}()
break
case "sendData":
// FIXME send to stream packet.StreamName
if _, err := ffmpegInput.Write(packet.Data); err != nil {
// FIXME send to stream srtPacket.StreamName
if _, err := ffmpegInput.Write(srtPacket.Data); err != nil {
log.Printf("Failed to write data to ffmpeg input: %s", err)
}
break
case "close":
log.Printf("WebRTC CloseConnection %s", packet.StreamName)
log.Printf("WebRTC CloseConnection %s", srtPacket.StreamName)
break
default:
log.Println("Unknown SRT packet type:", packet.PacketType)
log.Println("Unknown SRT srtPacket type:", srtPacket.PacketType)
break
}
if err != nil {
log.Printf("Error occured while receiving SRT packet of type %s: %s", packet.PacketType, err)
log.Printf("Error occured while receiving SRT srtPacket of type %s: %s", srtPacket.PacketType, err)
}
}
}