适配引擎版本升级

client.go

@@ -18,13 +18,15 @@ import (
 
 	. "github.com/Monibuca/engine/v3"
 	. "github.com/Monibuca/utils/v3"
-	"github.com/pion/rtp"
 )
 
 // PullStream 从外部拉流
 func (rtsp *RTSP) PullStream(streamPath string, rtspUrl string) (err error) {
-	if result := rtsp.Publish(streamPath); result {
+	rtsp.Stream = &Stream{
-		rtsp.Stream.Type = "RTSP"
+		StreamPath: streamPath,
+		Type:       "RTSP Pull",
+	}
+	if result := rtsp.Publish(); result {
 		rtsp.TransType = TRANS_TYPE_TCP
 		rtsp.vRTPChannel = 0
 		rtsp.vRTPControlChannel = 1
@@ -217,14 +219,12 @@ func (client *RTSP) requestStream() (err error) {
 		switch t {
 		case "video":
 			if len(sdpInfo.SpropParameterSets) > 1 {
-				vt := NewVideoTrack()
+				client.RtpVideo = client.NewRTPVideo(7)
-				vt.CodecID = 7
+				client.RtpVideo.PushNalu(VideoPack{NALUs: sdpInfo.SpropParameterSets})
-				var pack VideoPack
+			} else if client.VSdp.Codec == "H264" {
-				pack.Payload = sdpInfo.SpropParameterSets[0]
+				client.RtpVideo = client.NewRTPVideo(7)
-				vt.Push(pack)
+			} else if client.VSdp.Codec == "H265" {
-				pack.Payload = sdpInfo.SpropParameterSets[1]
+				client.RtpVideo = client.NewRTPVideo(12)
-				vt.Push(pack)
-				client.SetOriginVT(vt)
 			}
 			if client.TransType == TRANS_TYPE_TCP {
 				headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.vRTPChannel, client.vRTPControlChannel)
@@ -238,13 +238,13 @@ func (client *RTSP) requestStream() (err error) {
 				client.Conn.timeout = 0 //	UDP ignore timeout
 			}
 		case "audio":
-			at := NewAudioTrack()
+			client.RtpAudio = client.NewRTPAudio(0)
-			if len(sdpInfo.Config) > 0 {
+			at := client.RtpAudio.AudioTrack
-				at.SetASC(sdpInfo.Config)
+			if len(client.ASdp.Control) > 0 {
+				at.SetASC(client.ASdp.Config)
 			} else {
 				client.setAudioFormat(at)
 			}
-			client.SetOriginAT(at)
 			if client.TransType == TRANS_TYPE_TCP {
 				headers["Transport"] = fmt.Sprintf("RTP/AVP/TCP;unicast;interleaved=%d-%d", client.aRTPChannel, client.aRTPControlChannel)
 			} else {
@@ -339,15 +339,14 @@ func (client *RTSP) startStream() {
 				Printf("io.ReadFull err:%v", err)
 				return
 			}
-			var pack rtp.Packet
+			
-			pack.Unmarshal(content)
 			switch channel {
 			case client.aRTPChannel:
-				client.OriginAudioTrack.PushRTP(pack)
+				client.RtpAudio.Push(content)
 			case client.aRTPControlChannel:
 
 			case client.vRTPChannel:
-				client.OriginVideoTrack.PushRTP(pack)
+				client.RtpVideo.Push(content)
 			case client.vRTPControlChannel:
 
 			default:

go.mod

@@ -3,11 +3,9 @@ module github.com/Monibuca/plugin-rtsp/v3
 go 1.16
 
 require (
-	github.com/Monibuca/engine/v3 v3.0.0-alpha6
+	github.com/Monibuca/engine/v3 v3.0.0-beta5
-	github.com/Monibuca/utils/v3 v3.0.0-alpha5
+	github.com/Monibuca/utils/v3 v3.0.0-beta
-	github.com/logrusorgru/aurora v2.0.3+incompatible // indirect
+	github.com/pion/rtp v1.6.5
-	github.com/pion/rtp v1.6.2
-	github.com/shirou/gopsutil v3.21.1+incompatible // indirect
 	github.com/teris-io/shortid v0.0.0-20201117134242-e59966efd125
-	golang.org/x/sys v0.0.0-20210220050731-9a76102bfb43 // indirect
+	golang.org/x/sys v0.0.0-20210611083646-a4fc73990273 // indirect
 )

go.sum

@@ -1,98 +1,43 @@
 github.com/BurntSushi/toml v0.3.1 h1:WXkYYl6Yr3qBf1K79EBnL4mak0OimBfB0XUf9Vl28OQ=
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+golang.org/x/sys v0.0.0-20210611083646-a4fc73990273 h1:faDu4veV+8pcThn4fewv6TVlNCezafGoC1gM/mxQLbQ=
-gopkg.in/check.v1 v0.0.0-20161208181325-20d25e280405/go.mod h1:Co6ibVJAznAaIkqp8huTwlJQCZ016jof/cbN4VW5Yz0=
+golang.org/x/sys v0.0.0-20210611083646-a4fc73990273/go.mod h1:oPkhp1MJrh7nUepCBck5+mAzfO9JrbApNNgaTdGDITg=
-gopkg.in/yaml.v2 v2.2.2 h1:ZCJp+EgiOT7lHqUV2J862kp8Qj64Jo6az82+3Td9dZw=
-gopkg.in/yaml.v2 v2.2.2/go.mod h1:hI93XBmqTisBFMUTm0b8Fm+jr3Dg1NNxqwp+5A1VGuI=

main.go

@@ -6,7 +6,6 @@ import (
 	"log"
 	"net"
 	"net/http"
-	"strings"
 	"sync"
 	"time"
 
@@ -75,12 +74,12 @@ func runPlugin() {
 	if config.ListenAddr != "" {
 		go log.Fatal(ListenRtsp(config.ListenAddr))
 	}
-	AddHook(HOOK_SUBSCRIBE, func(value interface{}) {
+	// AddHook(HOOK_SUBSCRIBE, func(value interface{}) {
-		s := value.(*Subscriber)
+	// 	s := value.(*Subscriber)
-		if config.AutoPull && s.Publisher == nil {
+	// 	if config.AutoPull && s.Publisher == nil {
-			new(RTSP).PullStream(s.StreamPath, strings.Replace(config.RemoteAddr, "${streamPath}", s.StreamPath, -1))
+	// 		new(RTSP).PullStream(s.StreamPath, strings.Replace(config.RemoteAddr, "${streamPath}", s.StreamPath, -1))
-		}
+	// 	}
-	})
+	// })
 }
 
 func ListenRtsp(addr string) error {
@@ -129,7 +128,7 @@ func ListenRtsp(addr string) error {
 }
 
 type RTSP struct {
-	Publisher
+	*Stream
 	URL      string
 	SDPRaw   string
 	InBytes  int
@@ -159,21 +158,27 @@ type RTSP struct {
 	Auth               func(string) string `json:"-"`
 	HasVideo           bool
 	HasAudio           bool
+	RtpAudio           *RTPAudio
+	RtpVideo           *RTPVideo
 }
 
 func (rtsp *RTSP) setAudioFormat(at *AudioTrack) {
 	switch rtsp.ASdp.Codec {
 	case "aac":
-		at.SoundFormat = 10
+		at.CodecID = 10
 	case "pcma":
-		at.SoundFormat = 7
+		at.CodecID = 7
 		at.SoundRate = rtsp.ASdp.TimeScale
 		at.SoundSize = 16
 	case "pcmu":
-		at.SoundFormat = 8
+		at.CodecID = 8
 		at.SoundRate = rtsp.ASdp.TimeScale
 		at.SoundSize = 16
+	default:
+		Printf("rtsp audio codec not support:%s", rtsp.ASdp.Codec)
+		return
 	}
+	rtsp.AudioTracks.AddTrack(rtsp.ASdp.Codec, at)
 }
 
 type RTSPClientInfo struct {

session.go

@@ -114,43 +114,42 @@ func (session *RTSP) AcceptPush() {
 			}
 			channel := int(buf1)
 			rtpLen := int(binary.BigEndian.Uint16(buf2))
-			pack := new(RTPPack)
 			rtpBytes := make([]byte, rtpLen)
 			if _, err := io.ReadFull(session.connRW, rtpBytes); err != nil {
 				Println(err)
 				return
 			}
-			if err = pack.Unmarshal(rtpBytes); err != nil {
+
-				Println(err)
+			// t := pack.Timestamp / 90
-				return
-			}
-			t := pack.Timestamp / 90
-			Println(t,pack.SequenceNumber)
 			switch channel {
 			case session.aRTPChannel:
-				pack.Type = RTP_TYPE_AUDIO
+				// pack.Type = RTP_TYPE_AUDIO
-				elapsed := time.Now().Sub(timer)
+				elapsed := time.Since(timer)
 				if elapsed >= 30*time.Second {
 					Println("Recv an audio RTP package")
 					timer = time.Now()
 				}
-				for _, payload := range codec.ParseRTPAAC(pack.Payload) {
+				if session.RtpAudio.CodecID == 10 {
-					session.OriginAudioTrack.Push(t, payload)
+					for _, payload := range codec.ParseRTPAAC(rtpBytes) {
+						session.RtpAudio.Push(payload)
+					}
+				} else {
+					session.RtpAudio.Push(rtpBytes)
 				}
 			case session.aRTPControlChannel:
-				pack.Type = RTP_TYPE_AUDIOCONTROL
+				// pack.Type = RTP_TYPE_AUDIOCONTROL
 			case session.vRTPChannel:
-				pack.Type = RTP_TYPE_VIDEO
+				// pack.Type = RTP_TYPE_VIDEO
-				elapsed := time.Now().Sub(timer)
+				elapsed := time.Since(timer)
 				if elapsed >= 30*time.Second {
 					Println("Recv an video RTP package")
 					timer = time.Now()
 				}
-				session.OriginVideoTrack.Push(VideoPack{Timestamp: t, Payload: pack.Payload})
+				session.RtpVideo.Push(rtpBytes)
 			case session.vRTPControlChannel:
-				pack.Type = RTP_TYPE_VIDEOCONTROL
+				// pack.Type = RTP_TYPE_VIDEOCONTROL
 			default:
-				Printf("unknow rtp pack type, %v", pack.Type)
+				//	Printf("unknow rtp pack type, %v", pack.Type)
 				continue
 			}
 			session.InBytes += rtpLen + 4
@@ -335,28 +334,30 @@ func (session *RTSP) handleRequest(req *Request) {
 
 		session.SDPRaw = req.Body
 		session.SDPMap = ParseSDP(req.Body)
-		if session.Publish(streamPath) {
+		stream := &Stream{
+			StreamPath: streamPath,
+			Type:       "RTSP",
+		}
+		session.Stream = stream
+		if session.Publish() {
 			if session.ASdp, session.HasAudio = session.SDPMap["audio"]; session.HasAudio {
-				at := NewAudioTrack()
+				session.RtpAudio = stream.NewRTPAudio(0)
+				at := session.RtpAudio.AudioTrack
 				if len(session.ASdp.Control) > 0 {
 					at.SetASC(session.ASdp.Config)
 				} else {
 					session.setAudioFormat(at)
 				}
-				session.SetOriginAT(at)
 				Printf("audio codec[%s]\n", session.ASdp.Codec)
 			}
 			if session.VSdp, session.HasVideo = session.SDPMap["video"]; session.HasVideo {
 				if len(session.VSdp.SpropParameterSets) > 1 {
-					vt := NewVideoTrack()
+					session.RtpVideo = stream.NewRTPVideo(7)
-					vt.CodecID = 7
+					session.RtpVideo.PushNalu(VideoPack{NALUs: session.VSdp.SpropParameterSets})
-					var pack VideoPack
+				} else if session.VSdp.Codec == "H264" {
-					pack.Payload = session.VSdp.SpropParameterSets[0]
+					session.RtpVideo = stream.NewRTPVideo(7)
-					vt.Push(pack)
+				} else if session.VSdp.Codec == "H265" {
-					pack.Payload = session.VSdp.SpropParameterSets[1]
+					session.RtpVideo = stream.NewRTPVideo(12)
-					vt.Push(pack)
-					vt.SPSInfo, err = codec.ParseSPS(vt.SPS)
-					session.SetOriginVT(vt)
 				}
 				Printf("video codec[%s]\n", session.VSdp.Codec)
 			}

udp-server.go

@@ -25,19 +25,6 @@ func (s *UDPServer) AddInputBytes(bytes int) {
 	panic(fmt.Errorf("session and RTSPClient both nil"))
 }
 
-func (s *UDPServer) HandleRTP(pack RTPPack) {
-	if s.Session != nil {
-		s.Lock()
-		defer s.Unlock()
-		switch pack.Type {
-		case RTP_TYPE_AUDIO:
-			s.Session.OriginAudioTrack.PushRTP(pack.Packet)
-		case RTP_TYPE_VIDEO:
-			s.Session.OriginVideoTrack.PushRTP(pack.Packet)
-		}
-	}
-}
-
 func (s *UDPServer) Stop() {
 	if s.Stoped {
 		return
@@ -96,11 +83,7 @@ func (s *UDPServer) SetupAudio() (err error) {
 					timer = time.Now()
 				}
 				s.AddInputBytes(n)
-				pack := RTPPack{
+				s.Session.RtpAudio.Push(bufUDP[:n])
-					Type: RTP_TYPE_AUDIO,
-				}
-				pack.Unmarshal(bufUDP[:n])
-				s.HandleRTP(pack)
 			} else {
 				Println("udp server read audio pack error", err)
 				continue
@@ -135,11 +118,11 @@ func (s *UDPServer) SetupAudio() (err error) {
 			if n, _, err := s.AControlConn.ReadFromUDP(bufUDP); err == nil {
 				//Printf("Package recv from AControlConn.len:%d\n", n)
 				s.AddInputBytes(n)
-				pack := RTPPack{
+				// pack := RTPPack{
-					Type: RTP_TYPE_AUDIOCONTROL,
+				// 	Type: RTP_TYPE_AUDIOCONTROL,
-				}
+				// }
-				pack.Unmarshal(bufUDP[:n])
+				// pack.Unmarshal(bufUDP[:n])
-				s.HandleRTP(pack)
+				// s.HandleRTP(pack)
 			} else {
 				Println("udp server read audio control pack error", err)
 				continue
@@ -184,11 +167,7 @@ func (s *UDPServer) SetupVideo() (err error) {
 					timer = time.Now()
 				}
 				s.AddInputBytes(n)
-				pack := RTPPack{
+				s.Session.RtpVideo.Push(bufUDP[:n])
-					Type: RTP_TYPE_VIDEO,
-				}
-				pack.Unmarshal(bufUDP[:n])
-				s.HandleRTP(pack)
 			} else {
 				Println("udp server read video pack error", err)
 				continue
@@ -224,11 +203,11 @@ func (s *UDPServer) SetupVideo() (err error) {
 			if n, _, err := s.VControlConn.ReadFromUDP(bufUDP); err == nil {
 				//Printf("Package recv from VControlConn.len:%d\n", n)
 				s.AddInputBytes(n)
-				pack := RTPPack{
+				// pack := RTPPack{
-					Type: RTP_TYPE_VIDEOCONTROL,
+				// 	Type: RTP_TYPE_VIDEOCONTROL,
-				}
+				// }
-				pack.Unmarshal(bufUDP[:n])
+				// pack.Unmarshal(bufUDP[:n])
-				s.HandleRTP(pack)
+				// s.HandleRTP(pack)
 			} else {
 				Println("udp server read video control pack error", err)
 				continue