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Code Issues Releases Wiki Activity

Copy Pion RTP -> WebRTC example, it does not work but does not crash

This commit is contained in:
Yohann D'ANELLO 2020-10-05 00:45:22 +02:00
parent 4e066d6c33
commit 3a9568e764
1 changed files with 108 additions and 5 deletions
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113
stream/webrtc/webrtc.go
View File

@ -1,11 +1,15 @@
package webrtc
import (
"bufio"
"fmt"
"github.com/pion/rtp"
"io"
"log"
"math/rand"
"net"
"os"
"os/exec"
"time"
"github.com/pion/webrtc/v3"
@ -30,7 +34,7 @@ type SessionDescription = webrtc.SessionDescription
const (
audioFileName = "output.ogg"
videoFileName = "output.ivf"
videoFileName = "toto.ivf"
)
var (
@ -244,16 +248,115 @@ func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecNa
}
func waitForPackets(inputChannel chan srt.Packet) {
// FIXME Clean code
var ffmpeg *exec.Cmd
var ffmpegInput io.WriteCloser
for {
var err error = nil
packet := <-inputChannel
switch packet.PacketType {
case "register":
log.Printf("WebRTC RegisterStream %s", packet.StreamName)
// Copied from https://github.com/pion/webrtc/blob/master/examples/rtp-to-webrtc/main.go
// 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})
if err != nil {
panic(err)
}
audioListener, err := net.ListenUDP("udp", &net.UDPAddr{IP: net.ParseIP("127.0.0.1"), Port: 5005})
if err != nil {
panic(err)
}
defer func() {
if err = videoListener.Close(); err != nil {
panic(err)
}
if err = audioListener.Close(); err != nil {
panic(err)
}
}()
ffmpeg = exec.Command("ffmpeg", "-re", "-i", "pipe:0",
"-an", "-vcodec", "libvpx", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5004",
"-vn", "-acodec", "libopus", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5005")
fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")
input, err := ffmpeg.StdinPipe()
if err != nil {
panic(err)
}
ffmpegInput = input
errOutput, err := ffmpeg.StderrPipe()
if err != nil {
panic(err)
}
if err := ffmpeg.Start(); err != nil {
panic(err)
}
// Receive video
go func() {
for {
// Listen for a single RTP Packet, we need this to determine the SSRC
inboundRTPPacket := make([]byte, 1500) // UDP MTU
n, _, err := videoListener.ReadFromUDP(inboundRTPPacket)
if err != nil {
panic(err)
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
panic(err)
}
log.Printf("[Video] %s", packet)
for _, videoTrack := range videoTracks {
if writeErr := videoTrack.WriteRTP(packet); writeErr != nil {
panic(err)
}
}
}
}()
// Receive audio
go func() {
for {
// Listen for a single RTP Packet, we need this to determine the SSRC
inboundRTPPacket := make([]byte, 1500) // UDP MTU
n, _, err := audioListener.ReadFromUDP(inboundRTPPacket)
if err != nil {
panic(err)
}
packet := &rtp.Packet{}
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
panic(err)
}
log.Printf("[Audio] %s", packet)
for _, audioTrack := range audioTracks {
if writeErr := audioTrack.WriteRTP(packet); writeErr != nil {
panic(err)
}
}
}
}()
go func() {
scanner := bufio.NewScanner(errOutput)
for scanner.Scan() {
log.Printf("[WEBRTC FFMPEG %s] %s", "demo", scanner.Text())
}
}()
break
case "sendData":
log.Printf("WebRTC SendPacket %s", packet.StreamName)
// packet.Data
// log.Printf("WebRTC SendPacket %s", packet.StreamName)
_, err := ffmpegInput.Write(packet.Data)
if err != nil {
panic(err)
}
break
case "close":
log.Printf("WebRTC CloseConnection %s", packet.StreamName)
@ -274,8 +377,8 @@ func Serve(remoteSdpChan, localSdpChan chan webrtc.SessionDescription, inputChan
// FIXME: use data from inputChannel
go waitForPackets(inputChannel)
go playVideo()
go playAudio()
// go playVideo()
// go playAudio()
// Handle new connections
for {