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@kalradivyanshu
Created June 21, 2020 22:52
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package main
import (
"fmt"
"os"
"time"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
"github.com/pion/webrtc/v2/pkg/media/h264writer"
"github.com/pion/webrtc/v2/pkg/media/oggwriter"
"github.com/pion/webrtc/v2/examples/internal/signal"
)
func saveToDisk(i media.Writer, track *webrtc.Track) {
defer func() {
if err := i.Close(); err != nil {
panic(err)
}
}()
for {
rtpPacket, err := track.ReadRTP()
if err != nil {
panic(err)
}
if err := i.WriteRTP(rtpPacket); err != nil {
panic(err)
}
}
}
func main() {
// Create a MediaEngine object to configure the supported codec
m := webrtc.MediaEngine{}
// Setup the codecs you want to use.
// We'll use a VP8 codec but you can also define your own
m.RegisterCodec(webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000))
m.RegisterCodec(webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000))
// Create the API object with the MediaEngine
api := webrtc.NewAPI(webrtc.WithMediaEngine(m))
// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
// Prepare the configuration
config := webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
}
// Create a new RTCPeerConnection
peerConnection, err := api.NewPeerConnection(config)
if err != nil {
panic(err)
}
// Allow us to receive 1 audio track, and 1 video track
if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeAudio); err != nil {
panic(err)
} else if _, err = peerConnection.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo); err != nil {
panic(err)
}
oggFile, err := oggwriter.New("output.ogg", 48000, 2)
if err != nil {
panic(err)
}
h264File, err := h264writer.New("output.mp4")
if err != nil {
panic(err)
}
// Set a handler for when a new remote track starts, this handler saves buffers to disk as
// an ivf file, since we could have multiple video tracks we provide a counter.
// In your application this is where you would handle/process video
peerConnection.OnTrack(func(track *webrtc.Track, 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 {
errSend := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}})
if errSend != nil {
fmt.Println(errSend)
}
}
}()
codec := track.Codec()
if codec.Name == webrtc.H264 {
fmt.Println("Got H264 track, saving to disk as output.mp4")
saveToDisk(h264File, track)
}
if codec.Name == webrtc.Opus {
fmt.Println("Got Opus track, saving to disk as output.opus (48 kHz, 2 channels)")
saveToDisk(oggFile, track)
}
})
// 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())
if connectionState == webrtc.ICEConnectionStateConnected {
fmt.Println("Ctrl+C the remote client to stop the demo")
} else if connectionState == webrtc.ICEConnectionStateFailed ||
connectionState == webrtc.ICEConnectionStateDisconnected {
closeErr := oggFile.Close()
if closeErr != nil {
panic(closeErr)
}
closeErr = h264File.Close()
if closeErr != nil {
panic(closeErr)
}
fmt.Println("Done writing media files")
os.Exit(0)
}
})
// Wait for the offer to be pasted
offer := webrtc.SessionDescription{}
signal.Decode(signal.MustReadStdin(), &offer)
// Set the remote SessionDescription
err = peerConnection.SetRemoteDescription(offer)
if err != nil {
panic(err)
}
// Create answer
answer, err := peerConnection.CreateAnswer(nil)
if err != nil {
panic(err)
}
// Sets the LocalDescription, and starts our UDP listeners
err = peerConnection.SetLocalDescription(answer)
if err != nil {
panic(err)
}
// Output the answer in base64 so we can paste it in browser
fmt.Println(signal.Encode(answer))
// Block forever
select {}
}
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