forked from bluenviron/mediamtx
/
webrtc_incoming_track.go
146 lines (122 loc) · 2.96 KB
/
webrtc_incoming_track.go
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package core
import (
"fmt"
"strings"
"time"
"github.com/bluenviron/gortsplib/v3/pkg/formats"
"github.com/bluenviron/gortsplib/v3/pkg/media"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
"github.com/bluenviron/mediamtx/internal/stream"
)
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 strings.ToLower(track.Codec().MimeType) {
case strings.ToLower(webrtc.MimeTypeAV1):
t.mediaType = media.TypeVideo
t.format = &formats.AV1{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeVP9):
t.mediaType = media.TypeVideo
t.format = &formats.VP9{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeVP8):
t.mediaType = media.TypeVideo
t.format = &formats.VP8{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeH264):
t.mediaType = media.TypeVideo
t.format = &formats.H264{
PayloadTyp: uint8(track.PayloadType()),
PacketizationMode: 1,
}
case strings.ToLower(webrtc.MimeTypeOpus):
t.mediaType = media.TypeAudio
t.format = &formats.Opus{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeG722):
t.mediaType = media.TypeAudio
t.format = &formats.G722{}
case strings.ToLower(webrtc.MimeTypePCMU):
t.mediaType = media.TypeAudio
t.format = &formats.G711{
MULaw: true,
}
case strings.ToLower(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.Stream) {
go func() {
for {
pkt, _, err := t.track.ReadRTP()
if err != nil {
return
}
// sometimes Chrome sends empty RTP packets. ignore them.
if len(pkt.Payload) == 0 {
continue
}
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
}
}
}()
}
}