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ghostream
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WebRTC for the curious reference

This commit is contained in:
Alexandre Iooss 2020-09-24 19:01:26 +02:00
parent eafb4456c4
commit 6a138d1b1a
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GPG Key ID: 6C79278F3FCDCC02
2 changed files with 149 additions and 91 deletions
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1
README.md
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@ -20,3 +20,4 @@ It uses Traefik reverse proxy.
- Phil Cluff (2019), *[Streaming video on the internet without MPEG.](https://mux.com/blog/streaming-video-on-the-internet-without-mpeg/)*
- MDN web docs, *[Signaling and video calling.](https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling)*
- [WebRTC For The Curious](https://webrtcforthecurious.com/)

239
stream/stream.go
View File

@ -4,6 +4,7 @@ import (
"context"
"fmt"
"io"
"log"
"math/rand"
"os"
"time"
@ -19,8 +20,71 @@ const (
videoFileName = "output.ivf"
)
var (
peerConnectionConfig webrtc.Configuration
)
// newPeerHandler is called when server receive a new session description
// this initiates a WebRTC connection and return server description
func newPeerHandler(api *webrtc.API, remoteSdp webrtc.SessionDescription, audioTrack *webrtc.Track, videoTrack *webrtc.Track) webrtc.SessionDescription {
// Create a new PeerConnection
peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
if err != nil {
log.Println("Failed to initiate peer connection", err)
return webrtc.SessionDescription{}
}
// Add audio and video tracks
if _, err = peerConnection.AddTrack(audioTrack); err != nil {
log.Println("Failed to add audio track", err)
return webrtc.SessionDescription{}
}
if _, err = peerConnection.AddTrack(videoTrack); err != nil {
log.Println("Failed to add video track", err)
return webrtc.SessionDescription{}
}
// Set the remote SessionDescription
if err = peerConnection.SetRemoteDescription(remoteSdp); err != nil {
log.Println("Failed to set remote description", err)
return webrtc.SessionDescription{}
}
// Create answer
answer, err := peerConnection.CreateAnswer(nil)
if err != nil {
log.Println("Failed to create answer", err)
return webrtc.SessionDescription{}
}
// 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 {
log.Println("Failed to set local description", err)
return webrtc.SessionDescription{}
}
// 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 local description and send it to browser
return *peerConnection.LocalDescription()
}
// Serve WebRTC media streaming server
func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrtc.SessionDescription) {
peerConnectionConfig = webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
}
// Assert that we have an audio or video file
_, err := os.Stat(videoFileName)
haveVideoFile := !os.IsNotExist(err)
@ -30,129 +94,117 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
panic("Could not find `" + audioFileName + "` or `" + videoFileName + "`")
}
// Passing client offer
offer := <-remoteSdpChan
// 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{}
offer := <-remoteSdpChan
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"},
},
},
})
peerConnection, err := api.NewPeerConnection(peerConnectionConfig)
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"), rand.Uint32(), "video", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(videoTrack); addTrackErr != nil {
panic(addTrackErr)
// Create a video track
videoTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeVideo, "VP8"), rand.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)
}
go func() {
// Open a IVF file and start reading using our IVFReader
file, ivfErr := os.Open(videoFileName)
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)
}
ivf, header, ivfErr := ivfreader.NewWith(file)
if ivfErr != nil {
time.Sleep(sleepTime)
if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Samples: 90000}); 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)
}
}
}()
// Create a audio track
audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), rand.Uint32(), "audio", "pion")
if addTrackErr != nil {
panic(addTrackErr)
}
if _, addTrackErr = peerConnection.AddTrack(audioTrack); addTrackErr != nil {
panic(addTrackErr)
}
if haveAudioFile {
// Create a audio track
audioTrack, addTrackErr := peerConnection.NewTrack(getPayloadType(mediaEngine, webrtc.RTPCodecTypeAudio, "opus"), rand.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)
}
go func() {
// Open a IVF file and start reading using our IVFReader
file, oggErr := os.Open(audioFileName)
// 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)
}
// Open on oggfile in non-checksum mode.
ogg, _, oggErr := oggreader.NewWith(file)
if oggErr != nil {
// 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)
}
// 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)
}
}()
}
// 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
@ -190,8 +242,13 @@ func Serve(remoteSdpChan chan webrtc.SessionDescription, localSdpChan chan webrt
// Output the answer
localSdpChan <- *peerConnection.LocalDescription()
// Block forever
select {}
// Handle new connections
for {
// Wait for incoming session description
// then send the local description to browser
offer := <-remoteSdpChan
localSdpChan <- newPeerHandler(api, offer, audioTrack, videoTrack)
}
}
// Search for Codec PayloadType