Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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package core
import (
"fmt"
"time"
"github.com/bluenviron/gortsplib/v3/pkg/formats"
"github.com/bluenviron/gortsplib/v3/pkg/media"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
)
const (
keyFrameInterval = 2 * time.Second
)
type webRTCIncomingTrack struct {
track *webrtc.TrackRemote
receiver *webrtc.RTPReceiver
writeRTCP func([]rtcp.Packet) error
mediaType media.Type
format formats.Format
media *media.Media
}
func newWebRTCIncomingTrack(
track *webrtc.TrackRemote,
receiver *webrtc.RTPReceiver,
writeRTCP func([]rtcp.Packet) error,
) (*webRTCIncomingTrack, error) {
t := &webRTCIncomingTrack{
track: track,
receiver: receiver,
writeRTCP: writeRTCP,
}
switch track.Codec().MimeType {
case webrtc.MimeTypeAV1:
t.mediaType = media.TypeVideo
t.format = &formats.AV1{
PayloadTyp: uint8(track.PayloadType()),
}
case webrtc.MimeTypeVP9:
t.mediaType = media.TypeVideo
t.format = &formats.VP9{
PayloadTyp: uint8(track.PayloadType()),
}
case webrtc.MimeTypeVP8:
t.mediaType = media.TypeVideo
t.format = &formats.VP8{
PayloadTyp: uint8(track.PayloadType()),
}
case webrtc.MimeTypeH264:
t.mediaType = media.TypeVideo
t.format = &formats.H264{
PayloadTyp: uint8(track.PayloadType()),
PacketizationMode: 1,
}
case webrtc.MimeTypeOpus:
t.mediaType = media.TypeAudio
t.format = &formats.Opus{
PayloadTyp: uint8(track.PayloadType()),
}
case webrtc.MimeTypeG722:
t.mediaType = media.TypeAudio
t.format = &formats.G722{}
case webrtc.MimeTypePCMU:
t.mediaType = media.TypeAudio
t.format = &formats.G711{MULaw: true}
case webrtc.MimeTypePCMA:
t.mediaType = media.TypeAudio
t.format = &formats.G711{MULaw: false}
default:
return nil, fmt.Errorf("unsupported codec: %v", track.Codec())
}
t.media = &media.Media{
Type: t.mediaType,
Formats: []formats.Format{t.format},
}
return t, nil
}
func (t *webRTCIncomingTrack) start(stream *stream) {
go func() {
for {
pkt, _, err := t.track.ReadRTP()
if err != nil {
return
}
stream.writeRTPPacket(t.media, t.format, pkt, time.Now())
}
}()
// read incoming RTCP packets to make interceptors work
go func() {
buf := make([]byte, 1500)
for {
_, _, err := t.receiver.Read(buf)
if err != nil {
return
}
}
}()
if t.mediaType == media.TypeVideo {
go func() {
keyframeTicker := time.NewTicker(keyFrameInterval)
defer keyframeTicker.Stop()
for range keyframeTicker.C {
err := t.writeRTCP([]rtcp.Packet{
&rtcp.PictureLossIndication{
MediaSSRC: uint32(t.track.SSRC()),
},
})
if err != nil {
return
}
}
}()
}
}