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
|
/*
Spacebar: A FOSS re-implementation and extension of the Discord.com backend.
Copyright (C) 2023 Spacebar and Spacebar Contributors
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU Affero General Public License as published
by the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU Affero General Public License for more details.
You should have received a copy of the GNU Affero General Public License
along with this program. If not, see <https://www.gnu.org/licenses/>.
*/
import { Stream } from "@spacebar/database";
import { mediaServer, Send, VoiceOPCodes, VoicePayload, WebRtcWebSocket } from "@spacebar/webrtc";
import type { WebRtcClient } from "@spacebarchat/spacebar-webrtc-types";
import { validateSchema, VoiceVideoSchema } from "@spacebar/schemas";
export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
if (!this.webRtcClient) return;
const { voiceRoomId } = this.webRtcClient;
const d = validateSchema("VoiceVideoSchema", payload.d) as VoiceVideoSchema;
if (this.type === "stream") {
const stream = await Stream.findOne({
where: { id: voiceRoomId },
});
if (!stream) return;
// only the stream owner can publish to a go live stream
if (stream?.owner_id != this.user_id) {
return;
}
}
const stream = d.streams?.find((element) => element.active);
const clientsThatNeedUpdate = new Set<WebRtcClient<WebRtcWebSocket>>();
const wantsToProduceAudio = d.audio_ssrc !== 0;
const wantsToProduceVideo = d.video_ssrc !== 0 && stream?.active;
// this is to handle a really weird case where the client sends audio info before the
// dtls ice connection is completely connected. Wait for connection for 3 seconds
// and if no connection, just ignore this message
if (!this.webRtcClient.webrtcConnected) {
if (wantsToProduceAudio) {
try {
await Promise.race([
new Promise<void>((resolve, _) => {
this.webRtcClient?.emitter.once("connected", () => resolve());
}),
new Promise<void>((resolve, reject) => {
// Reject after 3 seconds if still not connected
setTimeout(() => {
if (this.webRtcClient?.webrtcConnected) resolve();
else reject();
}, 3000);
}),
]);
} catch (e) {
return; // just ignore this message if client didn't connect within 3 seconds
}
} else return;
}
await Send(this, { op: VoiceOPCodes.MEDIA_SINK_WANTS, d: { any: 100 } });
// first check if we need stop any tracks
if (!wantsToProduceAudio && this.webRtcClient.isProducingAudio()) {
this.webRtcClient.stopPublishingTrack("audio");
}
if (!wantsToProduceVideo && this.webRtcClient.isProducingVideo()) {
this.webRtcClient.stopPublishingTrack("video");
}
// check if client has signaled that it will send audio
if (wantsToProduceAudio) {
// check if we are already producing audio, if not, publish a new audio track for it
if (!this.webRtcClient!.isProducingAudio()) {
console.log(`[${this.user_id}] publishing new audio track ssrc:${d.audio_ssrc}`);
await this.webRtcClient.publishTrack("audio", {
audio_ssrc: d.audio_ssrc,
});
}
// now check that all clients have subscribed to our audio
for (const client of mediaServer.getClientsForRtcServer<WebRtcWebSocket>(voiceRoomId)) {
if (client.user_id === this.user_id) continue;
if (!client.isSubscribedToTrack(this.user_id, "audio")) {
console.log(`[${client.user_id}] subscribing to audio track ssrcs: ${d.audio_ssrc}`);
await client.subscribeToTrack(this.webRtcClient.user_id, "audio");
clientsThatNeedUpdate.add(client);
}
}
}
// check if client has signaled that it will send video
if (wantsToProduceVideo) {
// TODO: fix spacebar-webrtc-types...
// eslint-disable-next-line @typescript-eslint/ban-ts-comment
// @ts-expect-error
this.webRtcClient!.videoStream = { ...stream, type: "video" }; // client sends "screen" on go live but expects "video" on response
// check if we are already publishing video, if not, publish a new video track for it
if (!this.webRtcClient!.isProducingVideo()) {
console.log(`[${this.user_id}] publishing new video track ssrc:${d.video_ssrc}`);
await this.webRtcClient.publishTrack("video", {
video_ssrc: d.video_ssrc,
rtx_ssrc: d.rtx_ssrc,
});
}
// now check that all clients have subscribed to our video track
for (const client of mediaServer.getClientsForRtcServer<WebRtcWebSocket>(voiceRoomId)) {
if (client.user_id === this.user_id) continue;
if (!client.isSubscribedToTrack(this.user_id, "video")) {
console.log(`[${client.user_id}] subscribing to video track ssrc: ${d.video_ssrc}`);
await client.subscribeToTrack(this.webRtcClient.user_id, "video");
clientsThatNeedUpdate.add(client);
}
}
}
await Promise.all(
Array.from(clientsThatNeedUpdate).map((client) => {
const ssrcs = client.getOutgoingStreamSSRCsForUser(this.user_id);
return Send(client.websocket, {
op: VoiceOPCodes.VIDEO,
d: {
user_id: this.user_id,
// can never send audio ssrc as 0, it will mess up client state for some reason. send server generated ssrc as backup
audio_ssrc: ssrcs.audio_ssrc ?? this.webRtcClient!.getIncomingStreamSSRCs().audio_ssrc,
video_ssrc: ssrcs.video_ssrc ?? 0,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
streams: d.streams?.map((x) => ({
...x,
ssrc: ssrcs.video_ssrc ?? 0,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
type: "video",
})),
} as VoiceVideoSchema,
});
}),
);
}
// check if we are not subscribed to producers in this server, if not, subscribe
export async function subscribeToProducers(this: WebRtcWebSocket): Promise<void> {
if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
const clients = mediaServer.getClientsForRtcServer<WebRtcWebSocket>(this.webRtcClient.voiceRoomId);
await Promise.all(
Array.from(clients).map(async (client) => {
let needsUpdate = false;
if (client.user_id === this.user_id) return; // cannot subscribe to self
if (client.isProducingAudio() && !this.webRtcClient!.isSubscribedToTrack(client.user_id, "audio")) {
await this.webRtcClient!.subscribeToTrack(client.user_id, "audio");
needsUpdate = true;
}
if (client.isProducingVideo() && !this.webRtcClient!.isSubscribedToTrack(client.user_id, "video")) {
await this.webRtcClient!.subscribeToTrack(client.user_id, "video");
needsUpdate = true;
}
if (!needsUpdate) return;
const ssrcs = this.webRtcClient!.getOutgoingStreamSSRCsForUser(client.user_id);
await Send(this, {
op: VoiceOPCodes.VIDEO,
d: {
user_id: client.user_id,
// can never send audio ssrc as 0, it will mess up client state for some reason. send server generated ssrc as backup
audio_ssrc: ssrcs.audio_ssrc ?? client.getIncomingStreamSSRCs().audio_ssrc,
video_ssrc: ssrcs.video_ssrc ?? 0,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
streams: [
client.videoStream ?? {
type: "video",
rid: "100",
ssrc: ssrcs.video_ssrc ?? 0,
active: client.isProducingVideo(),
quality: 100,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
max_bitrate: 2500000,
max_framerate: 20,
max_resolution: {
type: "fixed",
width: 1280,
height: 720,
},
},
],
} as VoiceVideoSchema,
});
}),
);
}
|
