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package webrtc
import (
"net"
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"github.com/pion/rtcp"
. "github.com/pion/webrtc/v3"
. "m7s.live/engine/v4"
"m7s.live/engine/v4/codec"
"m7s.live/engine/v4/track"
"m7s.live/engine/v4/util"
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)
type WebRTCSubscriber struct {
Subscriber
WebRTCIO
videoTrack *TrackLocalStaticRTP
audioTrack *TrackLocalStaticRTP
videoSender *RTPSender
audioSender *RTPSender
DC *DataChannel
flvHeadCache []byte
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}
func (suber *WebRTCSubscriber) OnEvent(event any) {
switch v := event.(type) {
case *track.Video:
switch v.CodecID {
case codec.CodecID_H264:
pli := "420028"
// pli = fmt.Sprintf("%x", v.GetDecoderConfiguration().Raw[0][1:4])
// if !strings.Contains(suber.SDP, pli) {
// list := reg_level.FindAllStringSubmatch(suber.SDP, -1)
// if len(list) > 0 {
// pli = list[0][1]
// }
// }
suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH264, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, v.Name, suber.Subscriber.Stream.Path)
case codec.CodecID_H265:
// suber.videoTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: MimeTypeH265, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=" + pli}, "video", suber.Subscriber.Stream.Path)
default:
return
}
if suber.videoTrack == nil {
suber.DC, _ = suber.PeerConnection.CreateDataChannel(suber.Subscriber.Stream.Path, nil)
} else {
suber.videoSender, _ = suber.PeerConnection.AddTrack(suber.videoTrack)
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go func() {
rtcpBuf := make([]byte, 1500)
for {
if n, _, rtcpErr := suber.videoSender.Read(rtcpBuf); rtcpErr != nil {
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return
} else {
if p, err := rtcp.Unmarshal(rtcpBuf[:n]); err == nil {
for _, pp := range p {
switch pp.(type) {
case *rtcp.PictureLossIndication:
// fmt.Println("PictureLossIndication")
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}
}
}
}
}
}()
}
suber.Subscriber.AddTrack(v) //接受这个track
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case *track.Audio:
audioMimeType := MimeTypePCMA
if v.CodecID == codec.CodecID_PCMU {
audioMimeType = MimeTypePCMU
}
if v.CodecID == codec.CodecID_PCMA || v.CodecID == codec.CodecID_PCMU {
suber.audioTrack, _ = NewTrackLocalStaticRTP(RTPCodecCapability{MimeType: audioMimeType}, v.Name, suber.Subscriber.Stream.Path)
suber.audioSender, _ = suber.PeerConnection.AddTrack(suber.audioTrack)
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suber.Subscriber.AddTrack(v) //接受这个track
}
case VideoDeConf:
if suber.DC != nil {
if suber.flvHeadCache == nil {
suber.flvHeadCache = make([]byte, 15)
suber.flvHeadCache[0] = 9
suber.DC.Send(codec.FLVHeader)
}
suber.DC.Send(util.ConcatBuffers(codec.VideoAVCC2FLV(0, v)))
}
case VideoRTP:
if suber.videoTrack != nil {
suber.videoTrack.WriteRTP(&v.Packet)
} else if suber.DC != nil {
frame := suber.VideoReader.Frame
dataSize := uint32(frame.AVCC.ByteLength)
result := net.Buffers{suber.flvHeadCache[:11]}
result = append(result, frame.AVCC.ToBuffers()...)
ts := suber.VideoReader.AbsTime
util.PutBE(suber.flvHeadCache[1:4], dataSize)
util.PutBE(suber.flvHeadCache[4:7], ts)
suber.flvHeadCache[7] = byte(ts >> 24)
result = append(result, util.PutBE(suber.flvHeadCache[11:15], dataSize+11))
for _, data := range util.SplitBuffers(result, 65535) {
for _, d := range data {
suber.DC.Send(d)
}
}
}
case AudioRTP:
suber.audioTrack.WriteRTP(&v.Packet)
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case ISubscriber:
suber.OnConnectionStateChange(func(pcs PeerConnectionState) {
suber.Info("Connection State has changed:" + pcs.String())
switch pcs {
case PeerConnectionStateConnected:
go suber.PlayRTP()
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case PeerConnectionStateDisconnected, PeerConnectionStateFailed:
suber.Stop()
suber.PeerConnection.Close()
}
})
default:
suber.Subscriber.OnEvent(event)
}
}
type WebRTCBatchSubscriber struct {
WebRTCSubscriber
}
func (suber *WebRTCBatchSubscriber) OnEvent(event any) {
switch event.(type) {
case ISubscriber:
default:
suber.WebRTCSubscriber.OnEvent(event)
}
}