forked from Monibuca/plugin-webrtc
-
Notifications
You must be signed in to change notification settings - Fork 0
Expand file tree
/
Copy pathmain.go
More file actions
361 lines (345 loc) · 8.59 KB
/
main.go
File metadata and controls
361 lines (345 loc) · 8.59 KB
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
package webrtc
import (
"encoding/json"
"fmt"
"io/ioutil"
"net/http"
"sync"
"time"
. "github.com/Monibuca/engine/v2"
"github.com/Monibuca/engine/v2/avformat"
. "github.com/Monibuca/plugin-rtp"
"github.com/pion/rtcp"
. "github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
)
var config struct {
ICEServers []string
}
// }{[]string{
// "stun:stun.ekiga.net",
// "stun:stun.ideasip.com",
// "stun:stun.schlund.de",
// "stun:stun.stunprotocol.org:3478",
// "stun:stun.voiparound.com",
// "stun:stun.voipbuster.com",
// "stun:stun.voipstunt.com",
// "stun:stun.voxgratia.org",
// "stun:stun.services.mozilla.com",
// "stun:stun.xten.com",
// "stun:stun.softjoys.com",
// "stun:stunserver.org",
// "stun:stun.schlund.de",
// "stun:stun.rixtelecom.se",
// "stun:stun.iptel.org",
// "stun:stun.ideasip.com",
// "stun:stun.fwdnet.net",
// "stun:stun.ekiga.net",
// "stun:stun01.sipphone.com",
// }}
// type udpConn struct {
// conn *net.UDPConn
// port int
// }
var playWaitList WaitList
type WaitList struct {
m map[string]*WebRTC
l sync.Mutex
}
func (wl *WaitList) Set(k string, v *WebRTC) {
wl.l.Lock()
defer wl.l.Unlock()
if wl.m == nil {
wl.m = make(map[string]*WebRTC)
}
wl.m[k] = v
}
func (wl *WaitList) Get(k string) *WebRTC {
wl.l.Lock()
defer wl.l.Unlock()
defer delete(wl.m, k)
return wl.m[k]
}
func init() {
InstallPlugin(&PluginConfig{
Config: &config,
Name: "WebRTC",
Type: PLUGIN_PUBLISHER | PLUGIN_SUBSCRIBER,
Run: run,
})
}
type WebRTC struct {
RTP
*PeerConnection
RemoteAddr string
videoTrack *Track
m MediaEngine
api *API
payloader avformat.H264
// codecs.H264Packet
// *os.File
}
func (rtc *WebRTC) Play(streamPath string) bool {
var sub Subscriber
sub.ID = rtc.RemoteAddr
sub.Type = "WebRTC"
var lastTimeStamp uint32
sub.OnData = func(packet *avformat.SendPacket) error {
if packet.Type == avformat.FLV_TAG_TYPE_AUDIO {
return nil
}
if packet.IsSequence {
rtc.payloader.PPS = sub.PPS
rtc.payloader.SPS = sub.SPS
} else {
var s uint32
if lastTimeStamp > 0 {
s = packet.Timestamp - lastTimeStamp
}
lastTimeStamp = packet.Timestamp
rtc.videoTrack.WriteSample(media.Sample{
Data: packet.Payload,
Samples: s * 90,
})
// if packet.IsKeyFrame {
// rtc.videoTrack.WriteSample(media.Sample{
// Data: sub.SPS,
// Samples: 0,
// })
// rtc.videoTrack.WriteSample(media.Sample{
// Data: sub.PPS,
// Samples: 0,
// })
// }
// for payload := packet.Payload[5:]; len(payload) > 4; {
// var naulLen = int(util.BigEndian.Uint32(payload))
// payload = payload[4:]
// rtc.videoTrack.WriteSample(media.Sample{
// Data: payload[:naulLen],
// Samples: s * 90,
// })
// s = 0
// payload = payload[naulLen:]
// }
}
return nil
}
// go sub.Subscribe(streamPath)
rtc.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
Printf("%s Connection State has changed %s ", streamPath, connectionState.String())
switch connectionState {
case ICEConnectionStateDisconnected:
sub.Close()
rtc.Close()
case ICEConnectionStateConnected:
//rtc.videoTrack = rtc.GetSenders()[0].Track()
sub.Subscribe(streamPath)
}
})
return true
}
func (rtc *WebRTC) Publish(streamPath string) bool {
rtc.m.RegisterCodec(NewRTPCodec(RTPCodecTypeVideo,
H264,
90000,
0,
"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
DefaultPayloadTypeH264,
new(avformat.H264)))
//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
rtc.api = NewAPI(WithMediaEngine(rtc.m))
peerConnection, err := rtc.api.NewPeerConnection(Configuration{
ICEServers: []ICEServer{
{
URLs: config.ICEServers,
},
},
})
if _, err = peerConnection.AddTransceiverFromKind(RTPCodecTypeVideo); err != nil {
if err != nil {
Println(err)
return false
}
}
if err != nil {
return false
}
peerConnection.OnICEConnectionStateChange(func(connectionState ICEConnectionState) {
Printf("%s Connection State has changed %s ", streamPath, connectionState.String())
switch connectionState {
case ICEConnectionStateDisconnected, ICEConnectionStateFailed:
if rtc.Stream != nil {
rtc.Stream.Close()
}
}
})
rtc.PeerConnection = peerConnection
if rtc.RTP.Publish(streamPath) {
//f, _ := os.OpenFile("resource/live/rtc.h264", os.O_TRUNC|os.O_WRONLY, 0666)
rtc.Stream.Type = "WebRTC"
peerConnection.OnTrack(func(track *Track, receiver *RTPReceiver) {
defer rtc.Stream.Close()
go func() {
ticker := time.NewTicker(time.Second * 2)
select {
case <-ticker.C:
if rtcpErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}}); rtcpErr != nil {
fmt.Println(rtcpErr)
}
case <-rtc.Done():
return
}
}()
pack := &RTPPack{
Type: RTPType(track.Kind() - 1),
}
for b := make([]byte, 1460); ; rtc.PushPack(pack) {
i, err := track.Read(b)
if err != nil {
return
}
if err = pack.Unmarshal(b[:i]); err != nil {
return
}
// rtc.Unmarshal(pack.Payload)
// f.Write(bytes)
}
})
} else {
return false
}
return true
}
func (rtc *WebRTC) GetAnswer() ([]byte, error) {
// Sets the LocalDescription, and starts our UDP listeners
answer, err := rtc.CreateAnswer(nil)
if err != nil {
return nil, err
}
if err := rtc.SetLocalDescription(answer); err != nil {
Println(err)
return nil, err
}
if bytes, err := json.Marshal(answer); err != nil {
Println(err)
return bytes, err
} else {
return bytes, nil
}
}
func run() {
http.HandleFunc("/webrtc/play", func(w http.ResponseWriter, r *http.Request) {
w.Header().Set("Access-Control-Allow-Credentials", "true")
origin := r.Header["Origin"]
if len(origin) == 0 {
w.Header().Set("Access-Control-Allow-Origin", "*")
} else {
w.Header().Set("Access-Control-Allow-Origin", origin[0])
}
w.Header().Set("Content-Type", "application/json")
streamPath := r.URL.Query().Get("streamPath")
var offer SessionDescription
var rtc WebRTC
bytes, err := ioutil.ReadAll(r.Body)
defer func() {
if err != nil {
Println(err)
fmt.Fprintf(w, `{"errmsg":"%s"}`, err)
return
}
rtc.Play(streamPath)
}()
if err != nil {
return
}
if err = json.Unmarshal(bytes, &offer); err != nil {
return
}
pli := "42001f"
if stream := FindStream(streamPath); stream != nil {
pli = fmt.Sprintf("%x", stream.SPS[1:4])
}
rtc.m.RegisterCodec(NewRTPCodec(RTPCodecTypeVideo,
H264,
90000,
0,
"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="+pli[:2]+"001f",
DefaultPayloadTypeH264,
&rtc.payloader))
//m.RegisterCodec(NewRTPPCMUCodec(DefaultPayloadTypePCMU, 8000))
rtc.api = NewAPI(WithMediaEngine(rtc.m))
peerConnection, err := rtc.api.NewPeerConnection(Configuration{
// ICEServers: []ICEServer{
// {
// URLs: config.ICEServers,
// },
// },
})
rtc.PeerConnection = peerConnection
rtc.OnICECandidate(func(ice *ICECandidate) {
if ice != nil {
Println(ice.ToJSON().Candidate)
}
})
// if r, err := peerConnection.AddTransceiverFromKind(RTPCodecTypeVideo); err == nil {
// rtc.videoTrack = r.Sender().Track()
// } else {
// Println(err)
// }
if err != nil {
return
}
rtc.RemoteAddr = r.RemoteAddr
if err = rtc.SetRemoteDescription(offer); err != nil {
return
}
// rtc.m.PopulateFromSDP(offer)
// var vpayloadType uint8 = 0
// for _, videoCodec := range rtc.m.GetCodecsByKind(RTPCodecTypeVideo) {
// if videoCodec.Name == H264 {
// vpayloadType = videoCodec.PayloadType
// videoCodec.Payloader = &rtc.payloader
// Printf("H264 fmtp %v", videoCodec.SDPFmtpLine)
// }
// }
// println(vpayloadType)
if rtc.videoTrack, err = rtc.NewTrack(DefaultPayloadTypeH264, 8, "video", "monibuca"); err != nil {
return
}
if _, err = rtc.AddTrack(rtc.videoTrack); err != nil {
return
}
if bytes, err := rtc.GetAnswer(); err == nil {
w.Write(bytes)
} else {
return
}
})
http.HandleFunc("/webrtc/publish", func(w http.ResponseWriter, r *http.Request) {
streamPath := r.URL.Query().Get("streamPath")
offer := SessionDescription{}
bytes, err := ioutil.ReadAll(r.Body)
err = json.Unmarshal(bytes, &offer)
if err != nil {
Println(err)
return
}
rtc := new(WebRTC)
rtc.RemoteAddr = r.RemoteAddr
if rtc.Publish(streamPath) {
if err := rtc.SetRemoteDescription(offer); err != nil {
Println(err)
return
}
if bytes, err = rtc.GetAnswer(); err == nil {
w.Write(bytes)
} else {
Println(err)
w.Write([]byte(err.Error()))
return
}
} else {
w.Write([]byte("bad name"))
}
})
}