apps/plugin-rtsp
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初步改造完成

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This commit is contained in:
dexter
2021-11-16 19:06:24 +08:00
parent ac8aa96350
commit a900613c70
15 changed files with 583 additions and 2123 deletions
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193
main.go
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@@ -1,25 +1,25 @@
package rtsp
import (
"bufio"
"encoding/json"
"fmt"
"log"
"net"
"net/http"
"sync"
"time"
. "github.com/Monibuca/engine/v3"
. "github.com/Monibuca/utils/v3"
"github.com/teris-io/shortid"
"github.com/aler9/gortsplib"
)
var config = struct {
ListenAddr string
UDPAddr string
RTCPAddr string
Timeout int
Reconnect bool
AutoPullList map[string]string
}{":554", 0, false, nil}
}{":554", ":8000", ":8001", 0, false, nil}
func init() {
InstallPlugin(&PluginConfig{
@@ -28,25 +28,36 @@ func init() {
Run: runPlugin,
})
}
func getRtspList() (info []*RTSPublisher) {
for _, s := range Streams.ToList() {
if rtsp, ok := s.ExtraProp.(*RTSPublisher); ok {
info = append(info, rtsp)
}
}
return
}
func runPlugin() {
http.HandleFunc("/api/rtsp/list", func(w http.ResponseWriter, r *http.Request) {
CORS(w, r)
if r.URL.Query().Get("json") != "" {
if jsonData, err := json.Marshal(getRtspList()); err == nil {
w.Write(jsonData)
} else {
w.WriteHeader(500)
}
return
}
sse := NewSSE(w, r.Context())
var err error
for tick := time.NewTicker(time.Second); err == nil; <-tick.C {
var info []*RTSP
for _, s := range Streams.ToList() {
if rtsp, ok := s.ExtraProp.(*RTSP); ok {
info = append(info, rtsp)
}
}
err = sse.WriteJSON(info)
err = sse.WriteJSON(getRtspList())
}
})
http.HandleFunc("/api/rtsp/pull", func(w http.ResponseWriter, r *http.Request) {
CORS(w, r)
targetURL := r.URL.Query().Get("target")
streamPath := r.URL.Query().Get("streamPath")
if err := (&RTSP{RTSPClientInfo: RTSPClientInfo{Agent: "Monibuca"}}).PullStream(streamPath, targetURL); err == nil {
if err := (&RTSPClient{}).PullStream(streamPath, targetURL); err == nil {
w.Write([]byte(`{"code":0}`))
} else {
w.Write([]byte(fmt.Sprintf(`{"code":1,"msg":"%s"}`, err.Error())))
@@ -54,7 +65,7 @@ func runPlugin() {
})
if len(config.AutoPullList) > 0 {
for streamPath, url := range config.AutoPullList {
if err := (&RTSP{RTSPClientInfo: RTSPClientInfo{Agent: "Monibuca"}}).PullStream(streamPath, url); err != nil {
if err := (&RTSPClient{}).PullStream(streamPath, url); err != nil {
Println(err)
}
}
@@ -66,151 +77,13 @@ func runPlugin() {
func ListenRtsp(addr string) error {
defer log.Println("rtsp server start!")
listener, err := net.Listen("tcp", addr)
if err != nil {
return err
s := &gortsplib.Server{
Handler: &RTSPServer{},
UDPRTPAddress: config.UDPAddr,
UDPRTCPAddress: config.RTCPAddr,
MulticastIPRange: "224.1.0.0/16",
MulticastRTPPort: 8002,
MulticastRTCPPort: 8003,
}
var tempDelay time.Duration
networkBuffer := 204800
timeoutMillis := config.Timeout
for {
conn, err := listener.Accept()
conn.(*net.TCPConn).SetNoDelay(false)
if err != nil {
if ne, ok := err.(net.Error); ok && ne.Temporary() {
if tempDelay == 0 {
tempDelay = 5 * time.Millisecond
} else {
tempDelay *= 2
}
if max := 1 * time.Second; tempDelay > max {
tempDelay = max
}
fmt.Printf("rtsp: Accept error: %v; retrying in %v", err, tempDelay)
time.Sleep(tempDelay)
continue
}
return err
}
tempDelay = 0
timeoutTCPConn := &RichConn{conn, time.Duration(timeoutMillis) * time.Millisecond}
go (&RTSP{
ID: shortid.MustGenerate(),
Conn: timeoutTCPConn,
connRW: bufio.NewReadWriter(bufio.NewReaderSize(timeoutTCPConn, networkBuffer), bufio.NewWriterSize(timeoutTCPConn, networkBuffer)),
Timeout: config.Timeout,
vRTPChannel: -1,
vRTPControlChannel: -1,
aRTPChannel: -1,
aRTPControlChannel: -1,
}).AcceptPush()
}
return nil
}
type RTSP struct {
*Stream `json:"-"`
URL string
SDPRaw string
InBytes int
OutBytes int
RTSPClientInfo
ID string
Conn *RichConn `json:"-"`
connRW *bufio.ReadWriter
connWLock sync.RWMutex
Type SessionType
TransType TransType
SDPMap map[string]*SDPInfo
nonce string
ASdp *SDPInfo
VSdp *SDPInfo
Timeout int
//tcp channels
aRTPChannel int
aRTPControlChannel int
vRTPChannel int
vRTPControlChannel int
UDPServer *UDPServer `json:"-"`
UDPClient *UDPClient `json:"-"`
Auth func(string) string `json:"-"`
HasVideo bool
HasAudio bool
RtpAudio *RTPAudio
RtpVideo *RTPVideo
}
func (rtsp *RTSP) setVideoTrack() {
if rtsp.VSdp.Codec == "H264" {
rtsp.RtpVideo = rtsp.NewRTPVideo(7)
if len(rtsp.VSdp.SpropParameterSets) > 1 {
rtsp.RtpVideo.PushNalu(0, 0, rtsp.VSdp.SpropParameterSets...)
}
} else if rtsp.VSdp.Codec == "H265" {
rtsp.RtpVideo = rtsp.NewRTPVideo(12)
if len(rtsp.VSdp.VPS) > 0 {
rtsp.RtpVideo.PushNalu(0, 0, rtsp.VSdp.VPS, rtsp.VSdp.SPS, rtsp.VSdp.PPS)
}
}
}
func (rtsp *RTSP) setAudioTrack() {
var at *RTPAudio
if len(rtsp.ASdp.Config) > 0 {
at = rtsp.NewRTPAudio(0)
at.SetASC(rtsp.ASdp.Config)
} else {
switch rtsp.ASdp.Codec {
case "AAC":
at = rtsp.NewRTPAudio(10)
case "PCMA":
at = rtsp.NewRTPAudio(7)
at.SoundRate = rtsp.ASdp.TimeScale
at.SoundSize = 16
at.Channels = 1
at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
case "PCMU":
at = rtsp.NewRTPAudio(8)
at.SoundRate = rtsp.ASdp.TimeScale
at.SoundSize = 16
at.Channels = 1
at.ExtraData = []byte{(at.CodecID << 4) | (1 << 1)}
default:
Printf("rtsp audio codec not support:%s", rtsp.ASdp.Codec)
return
}
}
rtsp.RtpAudio = at
}
type RTSPClientInfo struct {
Agent string
Session string
authLine string
Seq int
}
type RichConn struct {
net.Conn
timeout time.Duration
}
func (conn *RichConn) Read(b []byte) (n int, err error) {
if conn.timeout > 0 {
conn.Conn.SetReadDeadline(time.Now().Add(conn.timeout))
} else {
var t time.Time
conn.Conn.SetReadDeadline(t)
}
return conn.Conn.Read(b)
}
func (conn *RichConn) Write(b []byte) (n int, err error) {
if conn.timeout > 0 {
conn.Conn.SetWriteDeadline(time.Now().Add(conn.timeout))
} else {
var t time.Time
conn.Conn.SetWriteDeadline(t)
}
return conn.Conn.Write(b)
return s.StartAndWait(addr)
}