first commit

2025-09-26 19:21:22 +08:00 · 2021-08-29 15:39:15 +03:00
commit b6bd37f38b
9 changed files with 640 additions and 0 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -0,0 +1 @@
+.idea/
--- a/README.md
+++ b/README.md
@@ -0,0 +1,75 @@
+# RTSPtoRMP
+
+RTSP Stream to YouTube or Twitch
+
+full native! not use ffmpeg or gstreamer
+
+ 
+### Download Source
+
+1. Download source
+   ```bash 
+   $ git clone https://github.com/deepch/RTSPtoRTMP 
+   ```
+3. CD to Directory
+   ```bash
+    $ cd RTSPtoRTMP/
+   ```
+4. Test Run
+   ```bash
+    $ GO111MODULE=on go run *.go
+   ```
+5. Edit config.json
+
+
+## Configuration
+
+### Edit file config.json
+
+format:
+
+```bash
+{
+  "streams": {
+    "H264_AAC": {
+      "on_demand": false,
+      "url": "rtsp://171.25.232.11/05dad5221e774c85abbde206385a38b4",
+      "broadcast": {
+        "enable": true,
+        "url": "rtmp://a.rtmp.youtube.com/live2/{YOU_KEY}"
+      }
+    }
+  }
+}
+
+```
+
+## Limitations
+
+Video Codecs Supported: H264 all profiles
+
+Audio Codecs Supported: AAC or NONE
+
+## Test
+
+CPU usage 0.2% one core cpu intel core i7 / stream
+
+## Team
+
+Deepch - https://github.com/deepch streaming developer
+
+Dmitry - https://github.com/vdalex25 web developer
+
+## Other Example
+
+Examples of working with video on golang
+
+- [RTSPtoWeb](https://github.com/deepch/RTSPtoWeb)
+- [RTSPtoWebRTC](https://github.com/deepch/RTSPtoWebRTC)
+- [RTSPtoWSMP4f](https://github.com/deepch/RTSPtoWSMP4f)
+- [RTSPtoImage](https://github.com/deepch/RTSPtoImage)
+- [RTSPtoHLS](https://github.com/deepch/RTSPtoHLS)
+- [RTSPtoHLSLL](https://github.com/deepch/RTSPtoHLSLL)
+- [RTSPtoRTMP](https://github.com/deepch/RTSPtoRTMP)
+- 
+[![paypal.me/AndreySemochkin](https://ionicabizau.github.io/badges/paypal.svg)](https://www.paypal.me/AndreySemochkin) - You can make one-time donations via PayPal. I'll probably buy a ~~coffee~~ tea. :tea:
--- a/config.go
+++ b/config.go
@@ -0,0 +1,206 @@
+package main
+
+import (
+	"crypto/rand"
+	"encoding/json"
+	"errors"
+	"fmt"
+	"io/ioutil"
+	"log"
+	"sync"
+	"time"
+
+	"github.com/deepch/vdk/av"
+)
+
+var (
+	Success                         = "success"
+	ErrorStreamNotFound             = errors.New("stream not found")
+	ErrorStreamAlreadyExists        = errors.New("stream already exists")
+	ErrorStreamChannelAlreadyExists = errors.New("stream channel already exists")
+	ErrorStreamNotHLSSegments       = errors.New("stream hls not ts seq found")
+	ErrorStreamNoVideo              = errors.New("stream no video")
+	ErrorStreamNoClients            = errors.New("stream no clients")
+	ErrorStreamRestart              = errors.New("stream restart")
+	ErrorStreamStopCoreSignal       = errors.New("stream stop core signal")
+	ErrorStreamStopRTSPSignal       = errors.New("stream stop rtsp signal")
+	ErrorStreamChannelNotFound      = errors.New("stream channel not found")
+	ErrorStreamChannelCodecNotFound = errors.New("stream channel codec not ready, possible stream offline")
+	ErrorStreamsLen0                = errors.New("streams len zero")
+)
+
+//Config global
+var Config = loadConfig()
+
+//ConfigST struct
+type ConfigST struct {
+	mutex   sync.RWMutex
+	Server  ServerST            `json:"server"`
+	Streams map[string]StreamST `json:"streams"`
+}
+
+//ServerST struct
+type ServerST struct {
+	HTTPPort string `json:"http_port"`
+}
+
+//StreamST struct
+type StreamST struct {
+	URL       string      `json:"url"`
+	Broadcast BroadcastST `json:"broadcast"`
+	Status    bool        `json:"status"`
+	OnDemand  bool        `json:"on_demand"`
+	RunLock   bool        `json:"-"`
+	Codecs    []av.CodecData
+	Cl        map[string]viewer
+}
+type BroadcastST struct {
+	Enable bool   `json:"enable"`
+	URL    string `json:"url"`
+}
+
+//Segment HLS cache section
+type Segment struct {
+	dur  time.Duration
+	data []*av.Packet
+}
+
+type viewer struct {
+	c chan av.Packet
+}
+
+func (element *ConfigST) RunIFNotRun(uuid string) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	if tmp, ok := element.Streams[uuid]; ok {
+		if tmp.OnDemand && !tmp.RunLock {
+			tmp.RunLock = true
+			element.Streams[uuid] = tmp
+			go RTSPWorkerLoop(uuid, tmp.URL, tmp.OnDemand)
+		}
+	}
+}
+
+func (element *ConfigST) RunUnlock(uuid string) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	if tmp, ok := element.Streams[uuid]; ok {
+		if tmp.OnDemand && tmp.RunLock {
+			tmp.RunLock = false
+			element.Streams[uuid] = tmp
+		}
+	}
+}
+
+func (element *ConfigST) HasViewer(uuid string) bool {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	if tmp, ok := element.Streams[uuid]; ok && len(tmp.Cl) > 0 {
+		return true
+	}
+	return false
+}
+
+func loadConfig() *ConfigST {
+	var tmp ConfigST
+	data, err := ioutil.ReadFile("config.json")
+	if err != nil {
+		log.Fatalln(err)
+	}
+	err = json.Unmarshal(data, &tmp)
+	if err != nil {
+		log.Fatalln(err)
+	}
+	for i, v := range tmp.Streams {
+		v.Cl = make(map[string]viewer)
+		tmp.Streams[i] = v
+	}
+	return &tmp
+}
+
+func (element *ConfigST) cast(uuid string, pck av.Packet) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	for _, v := range element.Streams[uuid].Cl {
+		if len(v.c) < cap(v.c) {
+			v.c <- pck
+		}
+	}
+}
+
+func (element *ConfigST) ext(suuid string) bool {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	_, ok := element.Streams[suuid]
+	return ok
+}
+
+func (element *ConfigST) coAd(suuid string, codecs []av.CodecData) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	t := element.Streams[suuid]
+	t.Codecs = codecs
+	element.Streams[suuid] = t
+}
+
+func (element *ConfigST) coGe(suuid string) []av.CodecData {
+	for i := 0; i < 100; i++ {
+		element.mutex.RLock()
+		tmp, ok := element.Streams[suuid]
+		element.mutex.RUnlock()
+		if !ok {
+			return nil
+		}
+		if tmp.Codecs != nil {
+			return tmp.Codecs
+		}
+		time.Sleep(50 * time.Millisecond)
+	}
+	return nil
+}
+
+func (element *ConfigST) clAd(suuid string) (string, chan av.Packet) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	cuuid := pseudoUUID()
+	ch := make(chan av.Packet, 100)
+	element.Streams[suuid].Cl[cuuid] = viewer{c: ch}
+	return cuuid, ch
+}
+
+func (element *ConfigST) list() (string, []string) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	var res []string
+	var fist string
+	for k := range element.Streams {
+		if fist == "" {
+			fist = k
+		}
+		res = append(res, k)
+	}
+	return fist, res
+}
+
+func (element *ConfigST) clDe(suuid, cuuid string) {
+	element.mutex.Lock()
+	defer element.mutex.Unlock()
+	delete(element.Streams[suuid].Cl, cuuid)
+}
+
+func (element *ConfigST) bcConfig(suuid string) (bool, string) {
+	element.mutex.RLock()
+	defer element.mutex.RUnlock()
+	return element.Streams[suuid].Broadcast.Enable, element.Streams[suuid].Broadcast.URL
+}
+
+func pseudoUUID() (uuid string) {
+	b := make([]byte, 16)
+	_, err := rand.Read(b)
+	if err != nil {
+		fmt.Println("Error: ", err)
+		return
+	}
+	uuid = fmt.Sprintf("%X-%X-%X-%X-%X", b[0:4], b[4:6], b[6:8], b[8:10], b[10:])
+	return
+}
--- a/config.json
+++ b/config.json
@@ -0,0 +1,12 @@
+{
+  "streams": {
+    "H264_AAC": {
+      "on_demand": false,
+      "url": "YOU_RTSP_CAMERA_URL",
+      "broadcast": {
+        "enable": true,
+        "url": "rtmp://a.rtmp.youtube.com/live2/YOU_YOUTUBE_KEY"
+      }
+    }
+  }
+}
--- a/fakeAudio.go
+++ b/fakeAudio.go
@@ -0,0 +1,42 @@
+package main
+
+import (
+	"time"
+
+	"github.com/deepch/vdk/codec/aacparser"
+
+	"github.com/deepch/vdk/av"
+)
+
+type FakeAAC struct {
+	closer              chan bool
+	OutgoingPacketQueue string
+}
+
+func NewFakeAudio() *FakeAAC {
+	return &FakeAAC{closer: make(chan bool)}
+}
+
+func (obj *FakeAAC) Start(CodecData []av.CodecData, OutgoingPacketQueue chan *av.Packet) ([]av.CodecData, error) {
+	go func() {
+		ticker := time.NewTicker(128 * time.Millisecond)
+		for {
+			select {
+			case <-obj.closer:
+				return
+			case <-ticker.C:
+				OutgoingPacketQueue <- &av.Packet{Duration: 128 * time.Millisecond, Idx: 1, CompositionTime: 1 * time.Millisecond, Data: []byte{1, 2, 52, 24, 67, 243, 0, 69, 107, 122, 169, 68, 50, 47, 46, 147, 26, 82, 161, 42, 80, 119, 97, 77, 199, 109, 20, 8, 26, 99, 53, 19, 85, 4, 186, 150, 69, 121, 208, 72, 42, 74, 224, 144, 172, 132, 146, 70, 246, 74, 13, 98, 76, 226, 5, 214, 169, 73, 18, 154, 33, 43, 11, 196, 2, 132, 43, 2, 240, 10, 14, 162, 64, 149, 243, 217, 96, 225, 85, 233, 90, 11, 19, 41, 67, 32, 5, 45, 5, 74, 72, 138, 215, 107, 33, 96, 49, 66, 85, 152, 150, 222, 87, 151, 89, 46, 80, 1, 172, 66, 150, 150, 150, 150, 150, 150, 150, 150, 150, 151, 7}}
+
+			}
+		}
+	}()
+	audioCodec, err := aacparser.NewCodecDataFromMPEG4AudioConfig(aacparser.MPEG4AudioConfig{SampleRate: 8000, SampleRateIndex: 11, ObjectType: 2, ChannelLayout: 1, ChannelConfig: 1})
+	if err != nil {
+		return nil, err
+	}
+	return append(CodecData, audioCodec), nil
+}
+
+func (obj *FakeAAC) Close() {
+	obj.closer <- true
+}
--- a/go.mod
+++ b/go.mod
@@ -0,0 +1,5 @@
+module github.com/deepch/RTSPtoWSMP4f
+
+go 1.15
+
+require github.com/deepch/vdk v0.0.0-20210829103406-e99c71546c52
--- a/go.sum
+++ b/go.sum
@@ -0,0 +1,153 @@
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--- a/main.go
+++ b/main.go
@@ -0,0 +1,23 @@
+package main
+
+import (
+	"log"
+	"os"
+	"os/signal"
+	"syscall"
+)
+
+func main() {
+	go serveStreams()
+	sigs := make(chan os.Signal, 1)
+	done := make(chan bool, 1)
+	signal.Notify(sigs, syscall.SIGINT, syscall.SIGTERM)
+	go func() {
+		sig := <-sigs
+		log.Println(sig)
+		done <- true
+	}()
+	log.Println("Server Start Awaiting Signal")
+	<-done
+	log.Println("Exiting")
+}
--- a/stream.go
+++ b/stream.go
@@ -0,0 +1,123 @@
+package main
+
+import (
+	"errors"
+	"log"
+	"time"
+
+	"github.com/deepch/vdk/av"
+
+	"github.com/deepch/vdk/format/rtmp"
+
+	"github.com/deepch/vdk/format/rtspv2"
+)
+
+var (
+	ErrorStreamExitNoVideoOnStream = errors.New("Stream Exit No Video On Stream")
+	ErrorStreamExitRtspDisconnect  = errors.New("Stream Exit Rtsp Disconnect")
+	ErrorStreamExitNoViewer        = errors.New("Stream Exit On Demand No Viewer")
+)
+
+func serveStreams() {
+	for k, v := range Config.Streams {
+		if v.OnDemand {
+			log.Println("OnDemand not supported")
+			v.OnDemand = false
+		}
+		if !v.OnDemand {
+			go RTSPWorkerLoop(k, v.URL, v.OnDemand)
+		}
+	}
+}
+
+func RTSPWorkerLoop(name, url string, OnDemand bool) {
+	defer Config.RunUnlock(name)
+	for {
+		log.Println(name, "Stream Try Connect")
+		err := RTSPWorker(name, url, OnDemand)
+		if err != nil {
+			log.Println(err)
+		}
+		if OnDemand && !Config.HasViewer(name) {
+			log.Println(name, ErrorStreamExitNoViewer)
+			return
+		}
+		time.Sleep(1 * time.Second)
+	}
+}
+
+func RTSPWorker(name, url string, OnDemand bool) error {
+	keyTest := time.NewTimer(20 * time.Second)
+	clientTest := time.NewTimer(20 * time.Second)
+	RTSPClient, err := rtspv2.Dial(rtspv2.RTSPClientOptions{URL: url, DisableAudio: false, DialTimeout: 3 * time.Second, ReadWriteTimeout: 3 * time.Second, Debug: false})
+	if err != nil {
+		return err
+	}
+	defer RTSPClient.Close()
+	var connRTMP *rtmp.Conn
+	broadCastEnable, broadCastURL := Config.bcConfig(name)
+	if broadCastEnable {
+		connRTMP, err = rtmp.DialTimeout(broadCastURL, 5*time.Second)
+		//conn, err := rtmp.Dial("rtmp://live-ber.twitch.tv/app/live_55956025_liaYwOlnCZaZDkvVmcwGsTmznQqfhk")
+		if err != nil {
+			return err
+		}
+		defer connRTMP.Close()
+	}
+	if len(RTSPClient.CodecData) < 2 {
+		AudioStreamFake := NewFakeAudio()
+		RTSPClient.CodecData, err = AudioStreamFake.Start(RTSPClient.CodecData, RTSPClient.OutgoingPacketQueue)
+		defer AudioStreamFake.Close()
+	} else if RTSPClient.CodecData[1].Type() != av.AAC {
+		log.Fatalln("Only AAC or Disable Audio")
+	}
+	if broadCastEnable {
+		err = connRTMP.WriteHeader(RTSPClient.CodecData)
+		if err != nil {
+			return err
+		}
+	}
+	if RTSPClient.CodecData != nil {
+		Config.coAd(name, RTSPClient.CodecData)
+	}
+	var start bool
+	var lineA time.Duration
+	var lineV time.Duration
+	for {
+		select {
+		case <-clientTest.C:
+			if OnDemand && !Config.HasViewer(name) {
+				return ErrorStreamExitNoViewer
+			}
+		case <-keyTest.C:
+			return ErrorStreamExitNoVideoOnStream
+		case signals := <-RTSPClient.Signals:
+			switch signals {
+			case rtspv2.SignalCodecUpdate:
+				Config.coAd(name, RTSPClient.CodecData)
+			case rtspv2.SignalStreamRTPStop:
+				return ErrorStreamExitRtspDisconnect
+			}
+		case packetAV := <-RTSPClient.OutgoingPacketQueue:
+			if packetAV.IsKeyFrame {
+				keyTest.Reset(20 * time.Second)
+			}
+			if packetAV.IsKeyFrame && !start {
+				start = true
+			}
+			if packetAV.Idx == 0 {
+				lineV += packetAV.Duration
+				packetAV.Time = lineV
+			}
+			if packetAV.Idx == 1 {
+				lineA += packetAV.Duration
+				packetAV.Time = lineA
+			}
+			if broadCastEnable && start {
+				if err = connRTMP.WritePacket(*packetAV); err != nil {
+					return err
+				}
+			}
+		}
+	}
+}