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main.go
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package main
import (
"context"
"fmt"
"io"
"os"
"time"
"github.com/pion/randutil"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/examples/internal/signal"
"github.com/pion/webrtc/v3/pkg/media"
"github.com/pion/webrtc/v3/pkg/media/ivfreader"
"github.com/pion/webrtc/v3/pkg/media/oggreader"
)
const (
audioFileName = "output.ogg"
videoFileName = "output.ivf"
)
func main() {
// Assert that we have an audio or video file
_, err := os.Stat(videoFileName)
haveVideoFile := !os.IsNotExist(err)
_, err = os.Stat(audioFileName)
haveAudioFile := !os.IsNotExist(err)
if !haveAudioFile && !haveVideoFile {
panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
}
// Wait for the offer to be pasted
offer := webrtc.SessionDescription{}
signal.Decode(signal.MustReadStdin(), &offer)
// We make our own mediaEngine so we can place the sender's codecs in it. This because we must use the
// dynamic media type from the sender in our answer. This is not required if we are the offerer
mediaEngine := webrtc.MediaEngine{}
if err = mediaEngine.PopulateFromSDP(offer); err != nil {
panic(err)
}
// Create a new RTCPeerConnection
api := webrtc.NewAPI(webrtc.WithMediaEngine(mediaEngine))
peerConnection, err := api.NewPeerConnection(webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
})
if err != nil {
panic(err)
}
iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
if haveVideoFile {
// Create a video track
videoTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8"), randutil.NewMathRandomGenerator().Uint32(), "video", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(videoTrack); addTrackErr != nil {
panic(addTrackErr)
}
go func() {
// Open a IVF file and start reading using our IVFReader
file, ivfErr := os.Open(videoFileName)
if ivfErr != nil {
panic(ivfErr)
}
ivf, header, ivfErr := ivfreader.NewWith(file)
if ivfErr != nil {
panic(ivfErr)
}
// Wait for connection established
<-iceConnectedCtx.Done()
// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
sleepTime := time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000)
for {
frame, _, ivfErr := ivf.ParseNextFrame()
if ivfErr == io.EOF {
fmt.Printf("All video frames parsed and sent")
os.Exit(0)
}
if ivfErr != nil {
panic(ivfErr)
}
time.Sleep(sleepTime)
if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); ivfErr != nil {
panic(ivfErr)
}
}
}()
}
if haveAudioFile {
// Create a audio track
audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), randutil.NewMathRandomGenerator().Uint32(), "audio", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(audioTrack); addTrackErr != nil {
panic(addTrackErr)
}
go func() {
// Open a IVF file and start reading using our IVFReader
file, oggErr := os.Open(audioFileName)
if oggErr != nil {
panic(oggErr)
}
// Open on oggfile in non-checksum mode.
ogg, _, oggErr := oggreader.NewWith(file)
if oggErr != nil {
panic(oggErr)
}
// Wait for connection established
<-iceConnectedCtx.Done()
// Keep track of last granule, the difference is the amount of samples in the buffer
var lastGranule uint64
for {
pageData, pageHeader, oggErr := ogg.ParseNextPage()
if oggErr == io.EOF {
fmt.Printf("All audio pages parsed and sent")
os.Exit(0)
}
if oggErr != nil {
panic(oggErr)
}
// The amount of samples is the difference between the last and current timestamp
sampleCount := float64(pageHeader.GranulePosition - lastGranule)
lastGranule = pageHeader.GranulePosition
if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Samples: uint32(sampleCount)}); oggErr != nil {
panic(oggErr)
}
// Convert seconds to Milliseconds, Sleep doesn't accept floats
time.Sleep(time.Duration((sampleCount/48000)*1000) * time.Millisecond)
}
}()
}
// 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 {
iceConnectedCtxCancel()
}
})
// Set the remote SessionDescription
if err = peerConnection.SetRemoteDescription(offer); err != nil {
panic(err)
}
// Create 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
if err = peerConnection.SetLocalDescription(answer); 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(signal.Encode(*peerConnection.LocalDescription()))
// Block forever
select {}
}
// Search for Codec PayloadType
//
// Since we are answering we need to match the remote PayloadType
func getPayloadType(m webrtc.MediaEngine, codecType webrtc.RTPCodecType, codecName string) uint8 {
for _, codec := range m.GetCodecsByKind(codecType) {
if codec.Name == codecName {
return codec.PayloadType
}
}
panic(fmt.Sprintf("Remote peer does not support %s", codecName))
}