extract stuff to external library
This commit is contained in:
dank074
parent
77b8d45543
commit
b13198ce66
@ -67,6 +67,7 @@
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"husky": "^9.1.7",
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"husky": "^9.1.7",
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"prettier": "^3.5.3",
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"prettier": "^3.5.3",
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"pretty-quick": "^4.1.1",
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"pretty-quick": "^4.1.1",
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"spacebar-webrtc-types": "github:dank074/spacebar-webrtc-types",
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"typescript": "^5.8.3"
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"typescript": "^5.8.3"
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},
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},
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"dependencies": {
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"dependencies": {
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@ -124,8 +125,6 @@
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"optionalDependencies": {
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"optionalDependencies": {
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"@yukikaze-bot/erlpack": "^1.0.1",
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"@yukikaze-bot/erlpack": "^1.0.1",
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"jimp": "^1.6.0",
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"jimp": "^1.6.0",
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"@dank074/medooze-media-server": "1.156.4",
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"semantic-sdp": "^3.31.1",
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"mysql": "^2.18.1",
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"mysql": "^2.18.1",
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"nodemailer-mailgun-transport": "^2.1.5",
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"nodemailer-mailgun-transport": "^2.1.5",
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"nodemailer-mailjet-transport": "github:n0script22/nodemailer-mailjet-transport",
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"nodemailer-mailjet-transport": "github:n0script22/nodemailer-mailjet-transport",
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@ -21,7 +21,12 @@ import dotenv from "dotenv";
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import http from "http";
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import http from "http";
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import ws from "ws";
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import ws from "ws";
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import { Connection } from "./events/Connection";
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import { Connection } from "./events/Connection";
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import { mediaServer } from "./util/MediaServer";
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import {
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mediaServer,
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WRTC_PORT_MAX,
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WRTC_PORT_MIN,
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WRTC_PUBLIC_IP,
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} from "./util/MediaServer";
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import { green, yellow } from "picocolors";
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import { green, yellow } from "picocolors";
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dotenv.config();
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dotenv.config();
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@ -77,7 +82,7 @@ export class Server {
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console.log(`[WebRTC] ${yellow("WEBRTC disabled")}`);
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console.log(`[WebRTC] ${yellow("WEBRTC disabled")}`);
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return Promise.resolve();
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return Promise.resolve();
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}
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}
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await mediaServer.start();
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await mediaServer.start(WRTC_PUBLIC_IP, WRTC_PORT_MIN, WRTC_PORT_MAX);
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if (!this.server.listening) {
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if (!this.server.listening) {
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this.server.listen(this.port);
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this.server.listen(this.port);
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console.log(`[WebRTC] ${green(`online on 0.0.0.0:${this.port}`)}`);
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console.log(`[WebRTC] ${green(`online on 0.0.0.0:${this.port}`)}`);
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@ -1,274 +0,0 @@
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import { CodecInfo, MediaInfo, SDPInfo } from "semantic-sdp";
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import { SignalingDelegate } from "../util/SignalingDelegate";
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import { Codec, WebRtcClient } from "../util/WebRtcClient";
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import { MediaServer, Endpoint } from "@dank074/medooze-media-server";
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import { VoiceRoom } from "./VoiceRoom";
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import { MedoozeWebRtcClient } from "./MedoozeWebRtcClient";
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export class MedoozeSignalingDelegate implements SignalingDelegate {
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private _rooms: Map<string, VoiceRoom> = new Map();
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private _ip: string;
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private _port: number;
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private _endpoint: Endpoint;
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public start(): Promise<void> {
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MediaServer.enableLog(true);
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this._ip = process.env.PUBLIC_IP || "127.0.0.1";
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try {
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const range = process.env.WEBRTC_PORT_RANGE || "3690-3960";
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var ports = range.split("-");
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const min = Number(ports[0]);
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const max = Number(ports[1]);
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MediaServer.setPortRange(min, max);
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} catch (error) {
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console.error(
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"Invalid env var: WEBRTC_PORT_RANGE",
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process.env.WEBRTC_PORT_RANGE,
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error,
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);
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process.exit(1);
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}
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//MediaServer.setAffinity(2)
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this._endpoint = MediaServer.createEndpoint(this._ip);
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this._port = this._endpoint.getLocalPort();
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return Promise.resolve();
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}
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public join(
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rtcServerId: string,
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userId: string,
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ws: any,
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type: "guild-voice" | "dm-voice" | "stream",
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): WebRtcClient<any> {
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// make sure user isn't already in a room of the same type
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// user can be in two simultanous rooms of different type though (can be in a voice channel and watching a stream for example)
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const rooms = this.rooms
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.values()
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.filter((room) =>
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type === "stream"
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? room.type === "stream"
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: room.type === "dm-voice" || room.type === "guild-voice",
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);
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let existingClient;
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for (const room of rooms) {
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let result = room.getClientById(userId);
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if (result) {
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existingClient = result;
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break;
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}
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}
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if (existingClient) {
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console.log("client already connected, disconnect..");
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this.onClientClose(existingClient);
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}
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if (!this._rooms.has(rtcServerId)) {
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console.debug("no channel created, creating one...");
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this.createRoom(rtcServerId, type);
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}
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const room = this._rooms.get(rtcServerId)!;
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const client = new MedoozeWebRtcClient(userId, rtcServerId, ws, room);
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room?.onClientJoin(client);
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return client;
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}
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public async onOffer(
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client: WebRtcClient<any>,
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sdpOffer: string,
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codecs: Codec[],
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): Promise<string> {
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const room = this._rooms.get(client.rtc_server_id);
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if (!room) {
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console.error(
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"error, client sent an offer but has not authenticated",
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);
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Promise.reject();
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}
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const offer = SDPInfo.parse("m=audio\n" + sdpOffer);
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const rtpHeaders = new Map(offer.medias[0].extensions);
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const getIdForHeader = (
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rtpHeaders: Map<number, string>,
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headerUri: string,
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) => {
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for (const [key, value] of rtpHeaders) {
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if (value == headerUri) return key;
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}
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return -1;
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};
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const audioMedia = new MediaInfo("0", "audio");
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const audioCodec = new CodecInfo(
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"opus",
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codecs.find((val) => val.name == "opus")?.payload_type ?? 111,
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);
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audioCodec.addParam("minptime", "10");
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audioCodec.addParam("usedtx", "1");
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audioCodec.addParam("useinbandfec", "1");
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audioCodec.setChannels(2);
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audioMedia.addCodec(audioCodec);
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audioMedia.addExtension(
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getIdForHeader(
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rtpHeaders,
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"urn:ietf:params:rtp-hdrext:ssrc-audio-level",
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),
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"urn:ietf:params:rtp-hdrext:ssrc-audio-level",
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);
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if (audioCodec.type === 111)
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// if this is chromium, apply this header
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audioMedia.addExtension(
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getIdForHeader(
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rtpHeaders,
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
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),
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
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);
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const videoMedia = new MediaInfo("1", "video");
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const videoCodec = new CodecInfo(
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"H264",
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codecs.find((val) => val.name == "H264")?.payload_type ?? 102,
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);
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videoCodec.setRTX(
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codecs.find((val) => val.name == "H264")?.rtx_payload_type ?? 103,
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);
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videoCodec.addParam("level-asymmetry-allowed", "1");
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videoCodec.addParam("packetization-mode", "1");
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videoCodec.addParam("profile-level-id", "42e01f");
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videoCodec.addParam("x-google-max-bitrate", "2500");
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videoMedia.addCodec(videoCodec);
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videoMedia.addExtension(
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getIdForHeader(
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rtpHeaders,
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
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),
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"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
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);
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videoMedia.addExtension(
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getIdForHeader(rtpHeaders, "urn:ietf:params:rtp-hdrext:toffset"),
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"urn:ietf:params:rtp-hdrext:toffset",
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);
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videoMedia.addExtension(
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getIdForHeader(
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rtpHeaders,
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"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
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),
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"http://www.webrtc.org/experiments/rtp-hdrext/playout-delay",
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);
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videoMedia.addExtension(
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getIdForHeader(
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rtpHeaders,
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
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),
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"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
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);
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if (audioCodec.type === 111)
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// if this is chromium, apply this header
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videoMedia.addExtension(
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getIdForHeader(rtpHeaders, "urn:3gpp:video-orientation"),
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"urn:3gpp:video-orientation",
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);
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offer.medias = [audioMedia, videoMedia];
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const transport = this._endpoint.createTransport(offer);
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transport.setRemoteProperties(offer);
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room?.onClientOffer(client, transport);
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const dtls = transport.getLocalDTLSInfo();
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const ice = transport.getLocalICEInfo();
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const fingerprint = dtls.getHash() + " " + dtls.getFingerprint();
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const candidates = transport.getLocalCandidates();
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const candidate = candidates[0];
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const answer =
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`m=audio ${this.port} ICE/SDP\n` +
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`a=fingerprint:${fingerprint}\n` +
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`c=IN IP4 ${this.ip}\n` +
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`a=rtcp:${this.port}\n` +
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`a=ice-ufrag:${ice.getUfrag()}\n` +
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`a=ice-pwd:${ice.getPwd()}\n` +
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`a=fingerprint:${fingerprint}\n` +
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`a=candidate:1 1 ${candidate.getTransport()} ${candidate.getFoundation()} ${candidate.getAddress()} ${candidate.getPort()} typ host\n`;
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return Promise.resolve(answer);
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}
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public onClientClose = (client: WebRtcClient<any>) => {
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this._rooms.get(client.rtc_server_id)?.onClientLeave(client);
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};
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public updateSDP(offer: string): void {
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throw new Error("Method not implemented.");
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}
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public createRoom(
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rtcServerId: string,
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type: "guild-voice" | "dm-voice" | "stream",
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): void {
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this._rooms.set(rtcServerId, new VoiceRoom(rtcServerId, type, this));
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}
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public disposeRoom(rtcServerId: string): void {
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const room = this._rooms.get(rtcServerId);
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room?.dispose();
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this._rooms.delete(rtcServerId);
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}
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get rooms(): Map<string, VoiceRoom> {
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return this._rooms;
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}
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public getClientsForRtcServer(rtcServerId: string): Set<WebRtcClient<any>> {
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if (!this._rooms.has(rtcServerId)) {
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return new Set();
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}
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return new Set(this._rooms.get(rtcServerId)?.clients.values())!;
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}
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private getClientForUserId = (
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userId: string,
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): MedoozeWebRtcClient | undefined => {
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for (const channel of this.rooms.values()) {
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let result = channel.getClientById(userId);
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if (result) {
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return result;
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}
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}
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return undefined;
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};
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get ip(): string {
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return this._ip;
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}
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get port(): number {
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return this._port;
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}
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get endpoint(): Endpoint {
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return this._endpoint;
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}
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public stop(): Promise<void> {
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return Promise.resolve();
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}
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}
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@ -1,152 +0,0 @@
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import {
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IncomingStream,
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OutgoingStream,
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Transport,
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} from "@dank074/medooze-media-server";
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import { SSRCs, WebRtcClient } from "webrtc/util";
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import { VoiceRoom } from "./VoiceRoom";
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export class MedoozeWebRtcClient implements WebRtcClient<any> {
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websocket: any;
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user_id: string;
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rtc_server_id: string;
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webrtcConnected: boolean;
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public transport?: Transport;
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public incomingStream?: IncomingStream;
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public outgoingStream?: OutgoingStream;
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public room?: VoiceRoom;
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public isStopped?: boolean;
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constructor(
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userId: string,
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rtcServerId: string,
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websocket: any,
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room: VoiceRoom,
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) {
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this.user_id = userId;
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this.rtc_server_id = rtcServerId;
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this.websocket = websocket;
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this.room = room;
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this.webrtcConnected = false;
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this.isStopped = false;
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}
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public isProducingAudio(): boolean {
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if (!this.webrtcConnected) return false;
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const audioTrack = this.incomingStream?.getTrack(
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`audio-${this.user_id}`,
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);
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if (audioTrack) return true;
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return false;
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}
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|
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public isProducingVideo(): boolean {
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if (!this.webrtcConnected) return false;
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const videoTrack = this.incomingStream?.getTrack(
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`video-${this.user_id}`,
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);
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if (videoTrack) return true;
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return false;
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}
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|
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public getIncomingStreamSSRCs(): SSRCs {
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if (!this.webrtcConnected)
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return { audio_ssrc: 0, video_ssrc: 0, rtx_ssrc: 0 };
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|
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const audioTrack = this.incomingStream?.getTrack(
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`audio-${this.user_id}`,
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);
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|
||||||
const audio_ssrc =
|
|
||||||
audioTrack?.getSSRCs()[audioTrack.getDefaultEncoding().id];
|
|
||||||
const videoTrack = this.incomingStream?.getTrack(
|
|
||||||
`video-${this.user_id}`,
|
|
||||||
);
|
|
||||||
const video_ssrc =
|
|
||||||
videoTrack?.getSSRCs()[videoTrack.getDefaultEncoding().id];
|
|
||||||
|
|
||||||
return {
|
|
||||||
audio_ssrc: audio_ssrc?.media ?? 0,
|
|
||||||
video_ssrc: video_ssrc?.media ?? 0,
|
|
||||||
rtx_ssrc: video_ssrc?.rtx ?? 0,
|
|
||||||
};
|
|
||||||
}
|
|
||||||
|
|
||||||
public getOutgoingStreamSSRCsForUser(user_id: string): SSRCs {
|
|
||||||
const outgoingStream = this.outgoingStream;
|
|
||||||
|
|
||||||
const audioTrack = outgoingStream?.getTrack(`audio-${user_id}`);
|
|
||||||
const audio_ssrc = audioTrack?.getSSRCs();
|
|
||||||
const videoTrack = outgoingStream?.getTrack(`video-${user_id}`);
|
|
||||||
const video_ssrc = videoTrack?.getSSRCs();
|
|
||||||
|
|
||||||
return {
|
|
||||||
audio_ssrc: audio_ssrc?.media ?? 0,
|
|
||||||
video_ssrc: video_ssrc?.media ?? 0,
|
|
||||||
rtx_ssrc: video_ssrc?.rtx ?? 0,
|
|
||||||
};
|
|
||||||
}
|
|
||||||
|
|
||||||
public publishTrack(type: "audio" | "video", ssrc: SSRCs) {
|
|
||||||
if (!this.transport) return;
|
|
||||||
|
|
||||||
const id = `${type}-${this.user_id}`;
|
|
||||||
const existingTrack = this.incomingStream?.getTrack(id);
|
|
||||||
|
|
||||||
if (existingTrack) {
|
|
||||||
console.error(`error: attempted to create duplicate track ${id}`);
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
let ssrcs;
|
|
||||||
if (type === "audio") {
|
|
||||||
ssrcs = { media: ssrc.audio_ssrc! };
|
|
||||||
} else {
|
|
||||||
ssrcs = { media: ssrc.video_ssrc!, rtx: ssrc.rtx_ssrc };
|
|
||||||
}
|
|
||||||
const track = this.transport?.createIncomingStreamTrack(
|
|
||||||
type,
|
|
||||||
{ id, ssrcs: ssrcs, media: type },
|
|
||||||
this.incomingStream,
|
|
||||||
);
|
|
||||||
|
|
||||||
//this.channel?.onClientPublishTrack(this, track, ssrcs);
|
|
||||||
}
|
|
||||||
|
|
||||||
public subscribeToTrack(user_id: string, type: "audio" | "video") {
|
|
||||||
if (!this.transport) return;
|
|
||||||
|
|
||||||
const id = `${type}-${user_id}`;
|
|
||||||
|
|
||||||
const otherClient = this.room?.getClientById(user_id);
|
|
||||||
const incomingStream = otherClient?.incomingStream;
|
|
||||||
const incomingTrack = incomingStream?.getTrack(id);
|
|
||||||
|
|
||||||
if (!incomingTrack) {
|
|
||||||
console.error(`error subscribing, not track found ${id}`);
|
|
||||||
return;
|
|
||||||
}
|
|
||||||
|
|
||||||
let ssrcs;
|
|
||||||
if (type === "audio") {
|
|
||||||
ssrcs = {
|
|
||||||
media: otherClient?.getIncomingStreamSSRCs().audio_ssrc!,
|
|
||||||
};
|
|
||||||
} else {
|
|
||||||
ssrcs = {
|
|
||||||
media: otherClient?.getIncomingStreamSSRCs().video_ssrc!,
|
|
||||||
rtx: otherClient?.getIncomingStreamSSRCs().rtx_ssrc,
|
|
||||||
};
|
|
||||||
}
|
|
||||||
|
|
||||||
const outgoingTrack = this.transport?.createOutgoingStreamTrack(
|
|
||||||
incomingTrack.media,
|
|
||||||
{ id, ssrcs, media: incomingTrack.media },
|
|
||||||
this.outgoingStream,
|
|
||||||
);
|
|
||||||
|
|
||||||
outgoingTrack?.attachTo(incomingTrack);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
@ -1,119 +0,0 @@
|
|||||||
import { MedoozeSignalingDelegate } from "./MedoozeSignalingDelegate";
|
|
||||||
import {
|
|
||||||
IncomingStreamTrack,
|
|
||||||
SSRCs,
|
|
||||||
Transport,
|
|
||||||
} from "@dank074/medooze-media-server";
|
|
||||||
import { MedoozeWebRtcClient } from "./MedoozeWebRtcClient";
|
|
||||||
import { StreamInfo } from "semantic-sdp";
|
|
||||||
|
|
||||||
export class VoiceRoom {
|
|
||||||
private _clients: Map<string, MedoozeWebRtcClient>;
|
|
||||||
private _id: string;
|
|
||||||
private _sfu: MedoozeSignalingDelegate;
|
|
||||||
private _type: "guild-voice" | "dm-voice" | "stream";
|
|
||||||
|
|
||||||
constructor(
|
|
||||||
id: string,
|
|
||||||
type: "guild-voice" | "dm-voice" | "stream",
|
|
||||||
sfu: MedoozeSignalingDelegate,
|
|
||||||
) {
|
|
||||||
this._id = id;
|
|
||||||
this._type = type;
|
|
||||||
this._clients = new Map();
|
|
||||||
this._sfu = sfu;
|
|
||||||
}
|
|
||||||
|
|
||||||
onClientJoin = (client: MedoozeWebRtcClient) => {
|
|
||||||
// do shit here
|
|
||||||
this._clients.set(client.user_id, client);
|
|
||||||
};
|
|
||||||
|
|
||||||
onClientOffer = (client: MedoozeWebRtcClient, transport: Transport) => {
|
|
||||||
client.transport = transport;
|
|
||||||
|
|
||||||
client.transport.on("dtlsstate", (state, self) => {
|
|
||||||
if (state === "connected") {
|
|
||||||
client.webrtcConnected = true;
|
|
||||||
console.log("connected");
|
|
||||||
}
|
|
||||||
});
|
|
||||||
|
|
||||||
client.incomingStream = transport.createIncomingStream(
|
|
||||||
new StreamInfo(`in-${client.user_id}`),
|
|
||||||
);
|
|
||||||
|
|
||||||
client.outgoingStream = transport.createOutgoingStream(
|
|
||||||
new StreamInfo(`out-${client.user_id}`),
|
|
||||||
);
|
|
||||||
|
|
||||||
client.webrtcConnected = true;
|
|
||||||
|
|
||||||
// subscribe to all current streams from this channel
|
|
||||||
// for(const otherClient of this._clients.values()) {
|
|
||||||
// const incomingStream = otherClient.incomingStream
|
|
||||||
|
|
||||||
// if(!incomingStream) continue;
|
|
||||||
|
|
||||||
// for(const track of (incomingStream.getTracks())) {
|
|
||||||
// client.subscribeToTrack(otherClient.user_id, track.media)
|
|
||||||
// }
|
|
||||||
// }
|
|
||||||
};
|
|
||||||
|
|
||||||
onClientLeave = (client: MedoozeWebRtcClient) => {
|
|
||||||
console.log("stopping client");
|
|
||||||
this._clients.delete(client.user_id);
|
|
||||||
|
|
||||||
// stop the client
|
|
||||||
if (!client.isStopped) {
|
|
||||||
client.isStopped = true;
|
|
||||||
|
|
||||||
for (const otherClient of this.clients.values()) {
|
|
||||||
//remove outgoing track for this user
|
|
||||||
otherClient.outgoingStream
|
|
||||||
?.getTrack(`audio-${client.user_id}`)
|
|
||||||
?.stop();
|
|
||||||
otherClient.outgoingStream
|
|
||||||
?.getTrack(`video-${client.user_id}`)
|
|
||||||
?.stop();
|
|
||||||
}
|
|
||||||
|
|
||||||
client.incomingStream?.stop();
|
|
||||||
client.outgoingStream?.stop();
|
|
||||||
|
|
||||||
client.transport?.stop();
|
|
||||||
client.room = undefined;
|
|
||||||
client.incomingStream = undefined;
|
|
||||||
client.outgoingStream = undefined;
|
|
||||||
client.transport = undefined;
|
|
||||||
client.websocket = undefined;
|
|
||||||
}
|
|
||||||
};
|
|
||||||
|
|
||||||
get clients(): Map<string, MedoozeWebRtcClient> {
|
|
||||||
return this._clients;
|
|
||||||
}
|
|
||||||
|
|
||||||
getClientById = (id: string) => {
|
|
||||||
return this._clients.get(id);
|
|
||||||
};
|
|
||||||
|
|
||||||
get id(): string {
|
|
||||||
return this._id;
|
|
||||||
}
|
|
||||||
|
|
||||||
get type(): "guild-voice" | "dm-voice" | "stream" {
|
|
||||||
return this._type;
|
|
||||||
}
|
|
||||||
|
|
||||||
public dispose(): void {
|
|
||||||
const clients = this._clients.values();
|
|
||||||
for (const client of clients) {
|
|
||||||
this.onClientLeave(client);
|
|
||||||
}
|
|
||||||
this._clients.clear();
|
|
||||||
this._sfu = undefined!;
|
|
||||||
this._clients = undefined!;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
@ -20,10 +20,10 @@ import {
|
|||||||
mediaServer,
|
mediaServer,
|
||||||
VoiceOPCodes,
|
VoiceOPCodes,
|
||||||
VoicePayload,
|
VoicePayload,
|
||||||
WebRtcClient,
|
|
||||||
WebRtcWebSocket,
|
WebRtcWebSocket,
|
||||||
Send,
|
Send,
|
||||||
} from "@spacebar/webrtc";
|
} from "@spacebar/webrtc";
|
||||||
|
import type { WebRtcClient } from "spacebar-webrtc-types";
|
||||||
|
|
||||||
export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
|
export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
|
||||||
if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
|
if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
|
||||||
|
|||||||
@ -16,25 +16,49 @@
|
|||||||
along with this program. If not, see <https://www.gnu.org/licenses/>.
|
along with this program. If not, see <https://www.gnu.org/licenses/>.
|
||||||
*/
|
*/
|
||||||
|
|
||||||
//import { MedoozeSignalingDelegate } from "../medooze/MedoozeSignalingDelegate";
|
import type { SignalingDelegate } from "spacebar-webrtc-types";
|
||||||
import { SignalingDelegate } from "./SignalingDelegate";
|
|
||||||
import { green, red } from "picocolors";
|
import { green, red } from "picocolors";
|
||||||
|
|
||||||
export let mediaServer: SignalingDelegate;
|
export let mediaServer: SignalingDelegate;
|
||||||
|
|
||||||
|
export const WRTC_PUBLIC_IP = process.env.WRTC_PUBLIC_IP ?? "127.0.0.1";
|
||||||
|
export const WRTC_PORT_MIN = process.env.WRTC_PORT_MIN
|
||||||
|
? parseInt(process.env.WRTC_PORT_MIN)
|
||||||
|
: 2000;
|
||||||
|
export const WRTC_PORT_MAX = process.env.WRTC_PORT_MAX
|
||||||
|
? parseInt(process.env.WRTC_PORT_MAX)
|
||||||
|
: 65000;
|
||||||
|
|
||||||
|
const selectedWrtcLibrary = process.env.WRTC_LIBRARY;
|
||||||
|
|
||||||
|
// could not find a way to hide stack trace from base Error object
|
||||||
|
class NoConfiguredLibraryError implements Error {
|
||||||
|
name: string;
|
||||||
|
message: string;
|
||||||
|
stack?: string | undefined;
|
||||||
|
cause?: unknown;
|
||||||
|
|
||||||
|
constructor(message: string) {
|
||||||
|
this.name = "NoConfiguredLibraryError";
|
||||||
|
this.message = message;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
(async () => {
|
(async () => {
|
||||||
try {
|
try {
|
||||||
//mediaServer = require('../medooze/MedoozeSignalingDelegate');
|
//mediaServer = require('medooze-spacebar-wrtc');
|
||||||
mediaServer = new (
|
if (!selectedWrtcLibrary)
|
||||||
await import("../medooze/MedoozeSignalingDelegate")
|
throw new NoConfiguredLibraryError("No library configured in .env");
|
||||||
).MedoozeSignalingDelegate();
|
|
||||||
|
mediaServer = new // @ts-ignore
|
||||||
|
(await import(selectedWrtcLibrary)).default();
|
||||||
|
|
||||||
console.log(
|
console.log(
|
||||||
`[WebRTC] ${green("Succesfully loaded MedoozeSignalingDelegate")}`,
|
`[WebRTC] ${green(`Succesfully loaded ${selectedWrtcLibrary}`)}`,
|
||||||
);
|
);
|
||||||
} catch (e) {
|
} catch (error) {
|
||||||
console.log(
|
console.log(
|
||||||
`[WebRTC] ${red("Failed to import MedoozeSignalingDelegate")}`,
|
`[WebRTC] ${red(`Failed to import ${selectedWrtcLibrary}: ${error instanceof NoConfiguredLibraryError ? error.message : ""}`)}`,
|
||||||
);
|
);
|
||||||
}
|
}
|
||||||
})();
|
})();
|
||||||
|
|||||||
@ -1,22 +0,0 @@
|
|||||||
import { Codec, WebRtcClient } from "./WebRtcClient";
|
|
||||||
|
|
||||||
export interface SignalingDelegate {
|
|
||||||
start: () => Promise<void>;
|
|
||||||
stop: () => Promise<void>;
|
|
||||||
join<T>(
|
|
||||||
rtcServerId: string,
|
|
||||||
userId: string,
|
|
||||||
ws: T,
|
|
||||||
type: "guild-voice" | "dm-voice" | "stream",
|
|
||||||
): WebRtcClient<T>;
|
|
||||||
onOffer<T>(
|
|
||||||
client: WebRtcClient<T>,
|
|
||||||
offer: string,
|
|
||||||
codecs: Codec[],
|
|
||||||
): Promise<string>;
|
|
||||||
onClientClose<T>(client: WebRtcClient<T>): void;
|
|
||||||
updateSDP(offer: string): void;
|
|
||||||
getClientsForRtcServer<T>(rtcServerId: string): Set<WebRtcClient<T>>;
|
|
||||||
get ip(): string;
|
|
||||||
get port(): number;
|
|
||||||
}
|
|
||||||
@ -1,31 +0,0 @@
|
|||||||
export interface WebRtcClient<T> {
|
|
||||||
websocket: T;
|
|
||||||
user_id: string;
|
|
||||||
rtc_server_id: string;
|
|
||||||
webrtcConnected: boolean;
|
|
||||||
getIncomingStreamSSRCs: () => SSRCs;
|
|
||||||
getOutgoingStreamSSRCsForUser: (user_id: string) => SSRCs;
|
|
||||||
isProducingAudio: () => boolean;
|
|
||||||
isProducingVideo: () => boolean;
|
|
||||||
publishTrack: (type: "audio" | "video", ssrc: SSRCs) => void;
|
|
||||||
subscribeToTrack: (user_id: string, type: "audio" | "video") => void;
|
|
||||||
}
|
|
||||||
|
|
||||||
export interface SSRCs {
|
|
||||||
audio_ssrc?: number;
|
|
||||||
video_ssrc?: number;
|
|
||||||
rtx_ssrc?: number;
|
|
||||||
}
|
|
||||||
|
|
||||||
export interface RtpHeader {
|
|
||||||
uri: string;
|
|
||||||
id: number;
|
|
||||||
}
|
|
||||||
|
|
||||||
export interface Codec {
|
|
||||||
name: "opus" | "VP8" | "VP9" | "H264";
|
|
||||||
type: "audio" | "video";
|
|
||||||
priority: number;
|
|
||||||
payload_type: number;
|
|
||||||
rtx_payload_type?: number;
|
|
||||||
}
|
|
||||||
@ -1,5 +1,5 @@
|
|||||||
import { WebSocket } from "@spacebar/gateway";
|
import { WebSocket } from "@spacebar/gateway";
|
||||||
import { WebRtcClient } from "./WebRtcClient";
|
import type { WebRtcClient } from "spacebar-webrtc-types";
|
||||||
|
|
||||||
export interface WebRtcWebSocket extends WebSocket {
|
export interface WebRtcWebSocket extends WebSocket {
|
||||||
type: "guild-voice" | "dm-voice" | "stream";
|
type: "guild-voice" | "dm-voice" | "stream";
|
||||||
|
|||||||
@ -18,6 +18,5 @@
|
|||||||
|
|
||||||
export * from "./Constants";
|
export * from "./Constants";
|
||||||
export * from "./MediaServer";
|
export * from "./MediaServer";
|
||||||
export * from "./WebRtcClient";
|
|
||||||
export * from "./WebRtcWebSocket";
|
export * from "./WebRtcWebSocket";
|
||||||
export * from "./Send";
|
export * from "./Send";
|
||||||
|
|||||||
Loading…
x
Reference in New Issue
Block a user