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/*
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/util";
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, reject) => {
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: ${d.audio_ssrc}`);
await client.subscribeToTrack(this.webRtcClient.user_id, "audio");
clientsThatNeedUpdate.add(client);
}
}
}
// check if client has signaled that it will send video
if (wantsToProduceVideo) {
this.webRtcClient!.videoStream = { ...stream, type: "video" }; // client sends "screen" on go live but expects "video" on response
// check if we are already publishing video, if not, publish a new video track for it
if (!this.webRtcClient!.isProducingVideo()) {
console.log(`[${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,
});
}),
);
}
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