install go2rtc on bob

2026-04-04 19:36:14 +02:00
parent f0b56e63d1
commit ccf88187b8
537 changed files with 69213 additions and 0 deletions
@@ -0,0 +1,204 @@
+package homekit
+
+import (
+	"fmt"
+	"io"
+	"math/rand"
+	"net"
+	"time"
+
+	"github.com/AlexxIT/go2rtc/pkg/core"
+	"github.com/AlexxIT/go2rtc/pkg/h264"
+	"github.com/AlexxIT/go2rtc/pkg/hap/camera"
+	"github.com/AlexxIT/go2rtc/pkg/opus"
+	"github.com/AlexxIT/go2rtc/pkg/srtp"
+	"github.com/pion/rtp"
+)
+
+type Consumer struct {
+	core.Connection
+	conn net.Conn
+	srtp *srtp.Server
+
+	deadline *time.Timer
+
+	sessionID    string
+	videoSession *srtp.Session
+	audioSession *srtp.Session
+	audioRTPTime byte
+}
+
+func NewConsumer(conn net.Conn, server *srtp.Server) *Consumer {
+	medias := []*core.Media{
+		{
+			Kind:      core.KindVideo,
+			Direction: core.DirectionSendonly,
+			Codecs: []*core.Codec{
+				{Name: core.CodecH264},
+			},
+		},
+		{
+			Kind:      core.KindAudio,
+			Direction: core.DirectionSendonly,
+			Codecs: []*core.Codec{
+				{Name: core.CodecOpus},
+			},
+		},
+	}
+	return &Consumer{
+		Connection: core.Connection{
+			ID:         core.NewID(),
+			FormatName: "homekit",
+			Protocol:   "rtp",
+			RemoteAddr: conn.RemoteAddr().String(),
+			Medias:     medias,
+			Transport:  conn,
+		},
+		conn: conn,
+		srtp: server,
+	}
+}
+
+func (c *Consumer) SessionID() string {
+	return c.sessionID
+}
+
+func (c *Consumer) SetOffer(offer *camera.SetupEndpointsRequest) {
+	c.sessionID = offer.SessionID
+	c.videoSession = &srtp.Session{
+		Remote: &srtp.Endpoint{
+			Addr:       offer.Address.IPAddr,
+			Port:       offer.Address.VideoRTPPort,
+			MasterKey:  []byte(offer.VideoCrypto.MasterKey),
+			MasterSalt: []byte(offer.VideoCrypto.MasterSalt),
+		},
+	}
+	c.audioSession = &srtp.Session{
+		Remote: &srtp.Endpoint{
+			Addr:       offer.Address.IPAddr,
+			Port:       offer.Address.AudioRTPPort,
+			MasterKey:  []byte(offer.AudioCrypto.MasterKey),
+			MasterSalt: []byte(offer.AudioCrypto.MasterSalt),
+		},
+	}
+}
+
+func (c *Consumer) GetAnswer() *camera.SetupEndpointsResponse {
+	c.videoSession.Local = c.srtpEndpoint()
+	c.audioSession.Local = c.srtpEndpoint()
+
+	return &camera.SetupEndpointsResponse{
+		SessionID: c.sessionID,
+		Status:    camera.StreamingStatusAvailable,
+		Address: camera.Address{
+			IPAddr:       c.videoSession.Local.Addr,
+			VideoRTPPort: c.videoSession.Local.Port,
+			AudioRTPPort: c.audioSession.Local.Port,
+		},
+		VideoCrypto: camera.SRTPCryptoSuite{
+			MasterKey:  string(c.videoSession.Local.MasterKey),
+			MasterSalt: string(c.videoSession.Local.MasterSalt),
+		},
+		AudioCrypto: camera.SRTPCryptoSuite{
+			MasterKey:  string(c.audioSession.Local.MasterKey),
+			MasterSalt: string(c.audioSession.Local.MasterSalt),
+		},
+		VideoSSRC: c.videoSession.Local.SSRC,
+		AudioSSRC: c.audioSession.Local.SSRC,
+	}
+}
+
+func (c *Consumer) SetConfig(conf *camera.SelectedStreamConfiguration) bool {
+	if c.sessionID != conf.Control.SessionID {
+		return false
+	}
+
+	c.SDP = fmt.Sprintf("%+v\n%+v", conf.VideoCodec, conf.AudioCodec)
+
+	c.videoSession.Remote.SSRC = conf.VideoCodec.RTPParams[0].SSRC
+	c.videoSession.PayloadType = conf.VideoCodec.RTPParams[0].PayloadType
+	c.videoSession.RTCPInterval = toDuration(conf.VideoCodec.RTPParams[0].RTCPInterval)
+
+	c.audioSession.Remote.SSRC = conf.AudioCodec.RTPParams[0].SSRC
+	c.audioSession.PayloadType = conf.AudioCodec.RTPParams[0].PayloadType
+	c.audioSession.RTCPInterval = toDuration(conf.AudioCodec.RTPParams[0].RTCPInterval)
+	c.audioRTPTime = conf.AudioCodec.CodecParams[0].RTPTime[0]
+
+	c.srtp.AddSession(c.videoSession)
+	c.srtp.AddSession(c.audioSession)
+
+	return true
+}
+
+func (c *Consumer) AddTrack(media *core.Media, codec *core.Codec, track *core.Receiver) error {
+	var session *srtp.Session
+	if codec.Kind() == core.KindVideo {
+		session = c.videoSession
+	} else {
+		session = c.audioSession
+	}
+
+	sender := core.NewSender(media, track.Codec)
+
+	if c.deadline == nil {
+		c.deadline = time.NewTimer(time.Second * 30)
+
+		sender.Handler = func(packet *rtp.Packet) {
+			c.deadline.Reset(core.ConnDeadline)
+			if n, err := session.WriteRTP(packet); err == nil {
+				c.Send += n
+			}
+		}
+	} else {
+		sender.Handler = func(packet *rtp.Packet) {
+			if n, err := session.WriteRTP(packet); err == nil {
+				c.Send += n
+			}
+		}
+	}
+
+	switch codec.Name {
+	case core.CodecH264:
+		sender.Handler = h264.RTPPay(1378, sender.Handler)
+		if track.Codec.IsRTP() {
+			sender.Handler = h264.RTPDepay(track.Codec, sender.Handler)
+		} else {
+			sender.Handler = h264.RepairAVCC(track.Codec, sender.Handler)
+		}
+	case core.CodecOpus:
+		sender.Handler = opus.RepackToHAP(c.audioRTPTime, sender.Handler)
+	}
+
+	sender.HandleRTP(track)
+	c.Senders = append(c.Senders, sender)
+	return nil
+}
+
+func (c *Consumer) WriteTo(io.Writer) (int64, error) {
+	if c.deadline != nil {
+		<-c.deadline.C
+	}
+	return 0, nil
+}
+
+func (c *Consumer) Stop() error {
+	if c.deadline != nil {
+		c.deadline.Reset(0)
+	}
+	return c.Connection.Stop()
+}
+
+func (c *Consumer) srtpEndpoint() *srtp.Endpoint {
+	addr := c.conn.LocalAddr().(*net.TCPAddr)
+	return &srtp.Endpoint{
+		Addr:       addr.IP.To4().String(),
+		Port:       uint16(c.srtp.Port()),
+		MasterKey:  []byte(core.RandString(16, 0)),
+		MasterSalt: []byte(core.RandString(14, 0)),
+		SSRC:       rand.Uint32(),
+	}
+}
+
+func toDuration(seconds float32) time.Duration {
+	return time.Duration(seconds * float32(time.Second))
+}