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- // SPDX-FileCopyrightText: 2023 The Pion community <https://pion.ly>
- // SPDX-License-Identifier: MIT
- //go:build !js
- // +build !js
- // play-from-disk demonstrates how to send video and/or audio to your browser from files saved to disk.
- package main
- import (
- "context"
- "errors"
- "fmt"
- "io"
- "os"
- "time"
- "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"
- oggPageDuration = time.Millisecond * 20
- )
- // nolint:gocognit
- 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 + "`")
- }
- // Create a new RTCPeerConnection
- peerConnection, err := webrtc.NewPeerConnection(webrtc.Configuration{
- ICEServers: []webrtc.ICEServer{
- {
- URLs: []string{"stun:stun.l.google.com:19302"},
- },
- },
- })
- if err != nil {
- panic(err)
- }
- defer func() {
- if cErr := peerConnection.Close(); cErr != nil {
- fmt.Printf("cannot close peerConnection: %v\n", cErr)
- }
- }()
- iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
- if haveVideoFile {
- file, openErr := os.Open(videoFileName)
- if openErr != nil {
- panic(openErr)
- }
- _, header, openErr := ivfreader.NewWith(file)
- if openErr != nil {
- panic(openErr)
- }
- // Determine video codec
- var trackCodec string
- switch header.FourCC {
- case "AV01":
- trackCodec = webrtc.MimeTypeAV1
- case "VP90":
- trackCodec = webrtc.MimeTypeVP9
- case "VP80":
- trackCodec = webrtc.MimeTypeVP8
- default:
- panic(fmt.Sprintf("Unable to handle FourCC %s", header.FourCC))
- }
- // Create a video track
- videoTrack, videoTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: trackCodec}, "video", "pion")
- if videoTrackErr != nil {
- panic(videoTrackErr)
- }
- rtpSender, videoTrackErr := peerConnection.AddTrack(videoTrack)
- if videoTrackErr != nil {
- panic(videoTrackErr)
- }
- // Read incoming RTCP packets
- // Before these packets are returned they are processed by interceptors. For things
- // like NACK this needs to be called.
- go func() {
- rtcpBuf := make([]byte, 1500)
- for {
- if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
- return
- }
- }
- }()
- 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.
- //
- // It is important to use a time.Ticker instead of time.Sleep because
- // * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
- // * works around latency issues with Sleep (see https://github.com/golang/go/issues/44343)
- ticker := time.NewTicker(time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000))
- for ; true; <-ticker.C {
- frame, _, ivfErr := ivf.ParseNextFrame()
- if errors.Is(ivfErr, io.EOF) {
- fmt.Printf("All video frames parsed and sent")
- os.Exit(0)
- }
- if ivfErr != nil {
- panic(ivfErr)
- }
- if ivfErr = videoTrack.WriteSample(media.Sample{Data: frame, Duration: time.Second}); ivfErr != nil {
- panic(ivfErr)
- }
- }
- }()
- }
- if haveAudioFile {
- // Create a audio track
- audioTrack, audioTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "pion")
- if audioTrackErr != nil {
- panic(audioTrackErr)
- }
- rtpSender, audioTrackErr := peerConnection.AddTrack(audioTrack)
- if audioTrackErr != nil {
- panic(audioTrackErr)
- }
- // Read incoming RTCP packets
- // Before these packets are returned they are processed by interceptors. For things
- // like NACK this needs to be called.
- go func() {
- rtcpBuf := make([]byte, 1500)
- for {
- if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
- return
- }
- }
- }()
- go func() {
- // Open a OGG file and start reading using our OGGReader
- 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
- // It is important to use a time.Ticker instead of time.Sleep because
- // * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
- // * works around latency issues with Sleep (see https://github.com/golang/go/issues/44343)
- ticker := time.NewTicker(oggPageDuration)
- for ; true; <-ticker.C {
- pageData, pageHeader, oggErr := ogg.ParseNextPage()
- if errors.Is(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
- sampleDuration := time.Duration((sampleCount/48000)*1000) * time.Millisecond
- if oggErr = audioTrack.WriteSample(media.Sample{Data: pageData, Duration: sampleDuration}); oggErr != nil {
- panic(oggErr)
- }
- }
- }()
- }
- // 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 handler for Peer connection state
- // This will notify you when the peer has connected/disconnected
- peerConnection.OnConnectionStateChange(func(s webrtc.PeerConnectionState) {
- fmt.Printf("Peer Connection State has changed: %s\n", s.String())
- if s == webrtc.PeerConnectionStateFailed {
- // Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
- // Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
- // Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
- fmt.Println("Peer Connection has gone to failed exiting")
- os.Exit(0)
- }
- })
- // Wait for the offer to be pasted
- offer := webrtc.SessionDescription{}
- signal.Decode(signal.MustReadStdin(), &offer)
- // 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 {}
- }
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