ashellunts / ffmpeg-to-webrtc Goto Github PK

package main

import (
	"fmt"
	"time"
	"os"
    "os/exec"

	"github.com/pion/rtcp"
	"github.com/pion/webrtc/v3"
	//gst "gstreamer-sink"
)

func check(err error) {
	if err != nil {
		panic(err)
	}
}

func main() {
	// Prepare the configuration
	config := webrtc.Configuration{
		ICEServers: []webrtc.ICEServer{
			{
				URLs: []string{"stun:stun.l.google.com:19302"},
			},
		},
	}

	// Create a new RTCPeerConnection
	peerConnection, err := webrtc.NewPeerConnection(config)
	if err != nil {
		panic(err)
	}

	// Set a handler for when a new remote track starts, this handler creates a gstreamer pipeline
	// for the given codec
	peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {

		// Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
		go func() {
			ticker := time.NewTicker(time.Second * 3)
			for range ticker.C {
				rtcpSendErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())}})
				if rtcpSendErr != nil {
					fmt.Println(rtcpSendErr)
				}				
			}
		}()

		//codecName := strings.Split(track.Codec().RTPCodecCapability.MimeType, "/")[1]
		//fmt.Printf("Track has started, of type %d: %s \n", track.PayloadType(), codecName)
		//pipeline := gst.CreatePipeline(track.PayloadType(), strings.ToLower(codecName))
		//pipeline.Start()
		
		buf := make([]byte, 1400)
		chBuff := make(chan []byte, 1400)

		go playTrack(chBuff)

		for {
			i, _, readErr := track.Read(buf)
			if readErr != nil {
				panic(err)
			}
			chBuff <- buf[:i]
			//pipeline.Push(buf[:i])
			//fmt.Printf("%x", buf[:i])
			//fmt.Println(track.PayloadType())			
		}
	})

	// 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())
	})

	// Wait for the offer to be pasted
	offer := webrtc.SessionDescription{}
	Decode(MustReadStdin(), &offer)

	// Set the remote SessionDescription
	err = peerConnection.SetRemoteDescription(offer)
	if err != nil {
		panic(err)
	}

	// Create an 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
	err = peerConnection.SetLocalDescription(answer)
	if 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 in base64 so we can paste it in browser
	fmt.Println(Encode(*peerConnection.LocalDescription()))

	// Block forever
	select {}
}

func playTrack(ch <-chan []byte){
	//cmd := exec.Command("ffmpeg", "-i", "pipe:0", "-f", "alsa", "default")
	cmd:= exec.Command("ffmpeg", "-i", "pipe:0", "-c:a", "copy", "-sample_fmt", "s16p", "-ssrc", "1", "-payload_type", "111",  "-b", "96k", "-f", "alsa", "default")

    cmd.Stderr = os.Stderr // bind log stream to stderr
    //cmd.Stdout = resultBuffer // stdout result will be written here

    stdin, err := cmd.StdinPipe() // Open stdin pipe
    check(err)

    err = cmd.Start() // Start a process on another goroutine
    check(err)

	for {
    	_, err = stdin.Write(<-ch) // pump audio data to stdin pipe
    	check(err)
	}

    err = stdin.Close() // close the stdin, or ffmpeg will wait forever
    check(err)

    err = cmd.Wait() // wait until ffmpeg finish
    check(err)
}