voice start
This commit is contained in:
MathMan05
parent
68fcc7a135
commit
369f30e5fc
6 changed files with 622 additions and 98 deletions
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@ -1,6 +1,6 @@
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import{ getBulkInfo }from"./login.js";
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class Voice{
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class AVoice{
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audioCtx: AudioContext;
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info: { wave: string | Function; freq: number };
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playing: boolean;
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@ -95,7 +95,7 @@ class Voice{
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static noises(noise: string): void{
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switch(noise){
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case"three": {
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const voicy = new Voice("sin", 800);
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const voicy = new AVoice("sin", 800);
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voicy.play();
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setTimeout(_=>{
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voicy.freq = 1000;
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@ -109,7 +109,7 @@ class Voice{
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break;
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}
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case"zip": {
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const voicy = new Voice((t: number, freq: number)=>{
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const voicy = new AVoice((t: number, freq: number)=>{
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return Math.sin((t + 2) ** Math.cos(t * 4) * Math.PI * 2 * freq);
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}, 700);
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voicy.play();
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@ -119,7 +119,7 @@ class Voice{
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break;
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}
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case"square": {
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const voicy = new Voice("square", 600, 0.4);
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const voicy = new AVoice("square", 600, 0.4);
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voicy.play();
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setTimeout(_=>{
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voicy.freq = 800;
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@ -133,7 +133,7 @@ class Voice{
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break;
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}
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case"beep": {
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const voicy = new Voice("sin", 800);
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const voicy = new AVoice("sin", 800);
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voicy.play();
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setTimeout(_=>{
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voicy.stop();
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@ -161,4 +161,4 @@ class Voice{
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return userinfos.preferences.notisound;
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}
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}
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export{ Voice };
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export{ AVoice as AVoice };
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@ -1,6 +1,6 @@
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"use strict";
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import{ Message }from"./message.js";
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import{ Voice }from"./audio.js";
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import{ AVoice }from"./audio.js";
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import{ Contextmenu }from"./contextmenu.js";
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import{ Dialog }from"./dialog.js";
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import{ Guild }from"./guild.js";
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@ -10,16 +10,10 @@ import{ Settings }from"./settings.js";
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import{ Role, RoleList }from"./role.js";
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import{ InfiniteScroller }from"./infiniteScroller.js";
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import{ SnowFlake }from"./snowflake.js";
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import{
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channeljson,
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embedjson,
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messageCreateJson,
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messagejson,
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readyjson,
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startTypingjson,
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}from"./jsontypes.js";
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import{channeljson,embedjson,messageCreateJson,messagejson,readyjson,startTypingjson}from"./jsontypes.js";
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import{ MarkDown }from"./markdown.js";
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import{ Member }from"./member.js";
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import { Voice } from "./voice.js";
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declare global {
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interface NotificationOptions {
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@ -55,6 +49,7 @@ class Channel extends SnowFlake{
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idToPrev: Map<string, string> = new Map();
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idToNext: Map<string, string> = new Map();
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messages: Map<string, Message> = new Map();
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voice?:Voice;
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static setupcontextmenu(){
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this.contextmenu.addbutton("Copy channel id", function(this: Channel){
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navigator.clipboard.writeText(this.id);
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@ -336,6 +331,9 @@ class Channel extends SnowFlake{
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}
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this.setUpInfiniteScroller();
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this.perminfo ??= {};
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if(this.type===2){
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this.voice=new Voice(this);
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}
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}
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get perminfo(){
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return this.guild.perminfo.channels[this.id];
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@ -840,6 +838,9 @@ class Channel extends SnowFlake{
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loading.classList.add("loading");
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this.rendertyping();
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this.localuser.getSidePannel();
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if(this.voice){
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this.voice.join();
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}
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await this.putmessages();
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await prom;
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if(id !== Channel.genid){
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@ -1334,7 +1335,7 @@ class Channel extends SnowFlake{
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);
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}
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notify(message: Message, deep = 0){
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Voice.noises(Voice.getNotificationSound());
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AVoice.noises(AVoice.getNotificationSound());
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if(!("Notification" in window)){
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}else if(Notification.permission === "granted"){
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let noticontent: string | undefined | null = message.content.textContent;
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@ -408,79 +408,109 @@ type wsjson =
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| "MESSAGE_REACTION_REMOVE_EMOJI";
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}
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| {
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op: 0;
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t: "GUILD_MEMBERS_CHUNK";
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d: memberChunk;
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s: number;
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op: 0;
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t: "GUILD_MEMBERS_CHUNK";
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d: memberChunk;
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s: number;
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}
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| {
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op: 0;
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d: {
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id: string;
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guild_id?: string;
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channel_id: string;
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};
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s: number;
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t: "MESSAGE_DELETE";
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op: 0;
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d: {
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id: string;
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guild_id?: string;
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channel_id: string;
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};
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s: number;
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t: "MESSAGE_DELETE";
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}
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| {
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op: 0;
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d: {
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guild_id?: string;
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channel_id: string;
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} & messagejson;
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s: number;
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t: "MESSAGE_UPDATE";
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op: 0;
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d: {
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guild_id?: string;
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channel_id: string;
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} & messagejson;
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s: number;
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t: "MESSAGE_UPDATE";
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}
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| messageCreateJson
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| readyjson
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| {
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op: 11;
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s: undefined;
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d: {};
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op: 11;
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s: undefined;
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d: {};
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}
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| {
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op: 10;
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s: undefined;
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d: {
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heartbeat_interval: number;
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};
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op: 10;
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s: undefined;
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d: {
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heartbeat_interval: number;
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};
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}
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| {
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op: 0;
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t: "MESSAGE_REACTION_ADD";
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d: {
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user_id: string;
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channel_id: string;
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message_id: string;
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guild_id?: string;
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emoji: emojijson;
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member?: memberjson;
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};
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s: number;
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op: 0;
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t: "MESSAGE_REACTION_ADD";
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d: {
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user_id: string;
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channel_id: string;
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message_id: string;
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guild_id?: string;
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emoji: emojijson;
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member?: memberjson;
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};
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s: number;
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}
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| {
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op: 0;
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t: "MESSAGE_REACTION_REMOVE";
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d: {
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user_id: string;
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channel_id: string;
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message_id: string;
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guild_id: string;
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emoji: emojijson;
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};
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s: 3;
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}|memberlistupdatejson;
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type memberChunk = {
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guild_id: string;
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nonce: string;
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members: memberjson[];
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presences: presencejson[];
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chunk_index: number;
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chunk_count: number;
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not_found: string[];
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};
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op: 0;
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t: "MESSAGE_REACTION_REMOVE";
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d: {
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user_id: string;
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channel_id: string;
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message_id: string;
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guild_id: string;
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emoji: emojijson;
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};
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s: 3;
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}|memberlistupdatejson|voiceupdate|voiceserverupdate;
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type memberChunk = {
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guild_id: string;
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nonce: string;
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members: memberjson[];
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presences: presencejson[];
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chunk_index: number;
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chunk_count: number;
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not_found: string[];
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};
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type voiceupdate={
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op: 0,
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t: "VOICE_STATE_UPDATE",
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d: {
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guild_id: string,
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channel_id: string,
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user_id: string,
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member: memberjson,
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session_id: string,
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token: string,
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deaf: boolean,
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mute: boolean,
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self_deaf: boolean,
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self_mute: boolean,
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self_video: boolean,
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suppress: boolean
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},
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s: number
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};
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type voiceserverupdate={
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op: 0,
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t: "VOICE_SERVER_UPDATE",
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d: {
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token: string,
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guild_id: string,
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endpoint: string
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},
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s: 6
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};
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type memberlistupdatejson={
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op: 0,
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s: number,
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@ -513,6 +543,65 @@ type memberlistupdatejson={
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}[]
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}
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}
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type webRTCSocket= {
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op: 8,
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d: {
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heartbeat_interval: number
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}
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}|{
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op:6,
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d:{t: number}
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}|{
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op: 2,
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d: {
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ssrc: number,
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"streams": {
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type: "video",//probally more options, but idk
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rid: string,
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quality: number,
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ssrc: number,
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rtx_ssrc:number
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}[],
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ip: number,
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port: number,
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"modes": [],//no clue
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"experiments": []//no clue
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}
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}|sdpback|opRTC12;
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type sdpback={
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op: 4,
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d: {
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audioCodec: string,
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videoCodec: string,
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media_session_id: string,
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sdp: string
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}
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};
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type opRTC12={
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op: 12,
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d: {
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user_id: string,
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audio_ssrc: number,
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video_ssrc: number,
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streams: [
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{
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type: "video",
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rid: "100",
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ssrc: number,
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active: boolean,
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quality: 100,
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rtx_ssrc: number,
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max_bitrate: 2500000,
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max_framerate: number,
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max_resolution: {
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type: "fixed",
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width: number,
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height: number
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}
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}
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]
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}
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}
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export{
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readyjson,
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dirrectjson,
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@ -532,5 +621,10 @@ export{
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messageCreateJson,
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memberChunk,
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invitejson,
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memberlistupdatejson
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memberlistupdatejson,
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voiceupdate,
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voiceserverupdate,
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webRTCSocket,
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sdpback,
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opRTC12
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};
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@ -1,27 +1,17 @@
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import{ Guild }from"./guild.js";
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import{ Channel }from"./channel.js";
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import{ Direct }from"./direct.js";
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import{ Voice }from"./audio.js";
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import{ AVoice }from"./audio.js";
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import{ User }from"./user.js";
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import{ Dialog }from"./dialog.js";
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import{ getapiurls, getBulkInfo, setTheme, Specialuser }from"./login.js";
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import{
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channeljson,
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guildjson,
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mainuserjson,
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memberjson,
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memberlistupdatejson,
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messageCreateJson,
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presencejson,
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readyjson,
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startTypingjson,
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wsjson,
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}from"./jsontypes.js";
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import{channeljson,guildjson,mainuserjson,memberjson,memberlistupdatejson,messageCreateJson,presencejson,readyjson,startTypingjson,voiceupdate,wsjson,}from"./jsontypes.js";
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import{ Member }from"./member.js";
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import{ Form, FormError, Options, Settings }from"./settings.js";
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import{ MarkDown }from"./markdown.js";
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import { Bot } from "./bot.js";
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import { Role } from "./role.js";
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import { Voice } from "./voice.js";
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const wsCodesRetry = new Set([4000, 4003, 4005, 4007, 4008, 4009]);
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@ -486,8 +476,21 @@ class Localuser{
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this.memberListUpdate(temp)
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break;
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}
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case "VOICE_STATE_UPDATE":
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if(this.waitingForVoice){
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this.waitingForVoice(temp);
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}
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break;
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case "VOICE_SERVER_UPDATE":
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if(this.currentVoice){
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Voice.url=temp.d.endpoint;
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Voice.gotUrl();
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}
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break;
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}
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}else if(temp.op === 10){
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if(!this.ws)return;
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console.log("heartbeat down");
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@ -501,6 +504,29 @@ class Localuser{
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}, this.heartbeat_interval);
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}
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}
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waitingForVoice?:((arg:voiceupdate|undefined)=>void);
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currentVoice?:Voice;
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async joinVoice(voice:Voice){
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this.currentVoice=voice;
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if(this.ws){
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this.ws.send(JSON.stringify({
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d:{
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guild_id: voice.guild.id,
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channel_id: voice.channel.id,
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self_mute: true,//todo
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self_deaf: false,//todo
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self_video: false,//What is this? I have some guesses
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flags: 2//?????
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},
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op:4
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}));
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if(this.waitingForVoice){
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this.waitingForVoice(undefined);
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}
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return await new Promise<voiceupdate|undefined>((res)=>{this.waitingForVoice=res;})
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}
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return undefined;
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}
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heartbeat_interval: number = 0;
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updateChannel(json: channeljson): void{
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const guild = this.guildids.get(json.guild_id);
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@ -1167,18 +1193,18 @@ class Localuser{
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);
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}
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{
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const sounds = Voice.sounds;
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const sounds = AVoice.sounds;
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tas
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.addSelect(
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"Notification sound:",
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_=>{
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Voice.setNotificationSound(sounds[_]);
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AVoice.setNotificationSound(sounds[_]);
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},
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sounds,
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{ defaultIndex: sounds.indexOf(Voice.getNotificationSound()) }
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{ defaultIndex: sounds.indexOf(AVoice.getNotificationSound()) }
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)
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.watchForChange(_=>{
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Voice.noises(sounds[_]);
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AVoice.noises(sounds[_]);
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});
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}
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|
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@ -116,12 +116,12 @@ function setDefaults(){
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setDefaults();
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class Specialuser{
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serverurls: {
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api: string;
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cdn: string;
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gateway: string;
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wellknown: string;
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login: string;
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};
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api: string;
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cdn: string;
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gateway: string;
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wellknown: string;
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login: string;
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};
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email: string;
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token: string;
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loggedin;
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|
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403
src/webpage/voice.ts
Normal file
403
src/webpage/voice.ts
Normal file
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@ -0,0 +1,403 @@
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import { Channel } from "./channel.js";
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import { sdpback, webRTCSocket } from "./jsontypes.js";
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class Voice{
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owner:Channel;
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static url?:string;
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static gotUrl:()=>void;
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static geturl=new Promise<void>(res=>{this.gotUrl=res})
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get channel(){
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return this.owner;
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}
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get guild(){
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return this.owner.owner;
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}
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get localuser(){
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return this.owner.localuser;
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}
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get info(){
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return this.owner.info;
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}
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constructor(owner:Channel){
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this.owner=owner;
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}
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pc?:RTCPeerConnection;
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ws?:WebSocket;
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timeout:number=30000;
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interval:NodeJS.Timeout=0 as unknown as NodeJS.Timeout;
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time:number=0;
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seq:number=0;
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sendAlive(){
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if(this.ws){
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this.ws.send(JSON.stringify({ op: 3,d:10}));
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}
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}
|
||||
readonly users= new Map<number,string>();
|
||||
packet(message:MessageEvent){
|
||||
const data=message.data
|
||||
if(typeof data === "string"){
|
||||
const json:webRTCSocket = JSON.parse(data);
|
||||
switch(json.op){
|
||||
case 2:
|
||||
this.startWebRTC();
|
||||
break;
|
||||
case 4:
|
||||
this.continueWebRTC(json);
|
||||
break;
|
||||
case 6:
|
||||
this.time=json.d.t;
|
||||
setTimeout(this.sendAlive.bind(this), this.timeout);
|
||||
break;
|
||||
case 8:
|
||||
this.timeout=json.d.heartbeat_interval;
|
||||
setTimeout(this.sendAlive.bind(this), 1000);
|
||||
break;
|
||||
case 12:
|
||||
this.users.set(json.d.audio_ssrc,json.d.user_id);
|
||||
break
|
||||
|
||||
|
||||
}
|
||||
}
|
||||
}
|
||||
offer?:string;
|
||||
cleanServerSDP(sdp:string,bundle1:string,bundle2:string):string{
|
||||
if(!this.offer) throw new Error("Offer is missing :P");
|
||||
let cline:string|undefined;
|
||||
console.log(sdp);
|
||||
for(const line of sdp.split("\n")){
|
||||
if(line.startsWith("c=")){
|
||||
cline=line;
|
||||
break;
|
||||
}
|
||||
}
|
||||
if(!cline) throw new Error("c line wasn't found");
|
||||
const parsed1=Voice.parsesdp(sdp).medias[0];
|
||||
//const parsed2=Voice.parsesdp(this.offer);
|
||||
const rtcport=(parsed1.atr.get("rtcp") as Set<string>).values().next().value as string;
|
||||
const ICE_UFRAG=(parsed1.atr.get("ice-ufrag") as Set<string>).values().next().value as string;
|
||||
const ICE_PWD=(parsed1.atr.get("ice-pwd") as Set<string>).values().next().value as string;
|
||||
const FINGERPRINT=(parsed1.atr.get("fingerprint") as Set<string>).values().next().value as string;
|
||||
const candidate=(parsed1.atr.get("candidate") as Set<string>).values().next().value as string;
|
||||
let build=`v=0\r
|
||||
o=- 1420070400000 0 IN IP4 127.0.0.1\r
|
||||
s=-\r
|
||||
t=0 0\r
|
||||
a=msid-semantic: WMS *\r
|
||||
a=group:BUNDLE ${bundle1} ${bundle2}\r
|
||||
m=audio ${parsed1.port} UDP/TLS/RTP/SAVPF 111\r
|
||||
${cline}\r
|
||||
a=rtpmap:111 opus/48000/2\r
|
||||
a=fmtp:111 minptime=10;useinbandfec=1;usedtx=1\r
|
||||
a=rtcp:${rtcport}\r
|
||||
a=rtcp-fb:111 transport-cc\r
|
||||
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r
|
||||
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/r/n
|
||||
a=setup:passive\r
|
||||
a=mid:${bundle1}\r
|
||||
a=maxptime:60\r
|
||||
a=inactive\r
|
||||
a=ice-ufrag:${ICE_UFRAG}\r
|
||||
a=ice-pwd:${ICE_PWD}\r
|
||||
a=fingerprint:${FINGERPRINT}\r
|
||||
a=candidate:${candidate}\r
|
||||
a=rtcp-mux\r
|
||||
m=video ${rtcport} UDP/TLS/RTP/SAVPF 102 103\r
|
||||
${cline}\r
|
||||
a=rtpmap:102 H264/90000\r
|
||||
a=rtpmap:103 rtx/90000\r
|
||||
a=fmtp:102 x-google-max-bitrate=2500;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r
|
||||
a=fmtp:103 apt=102\r
|
||||
a=rtcp:${rtcport}\r
|
||||
a=rtcp-fb:102 ccm fir\r
|
||||
a=rtcp-fb:102 nack\r
|
||||
a=rtcp-fb:102 nack pli\r
|
||||
a=rtcp-fb:102 goog-remb\r
|
||||
a=rtcp-fb:102 transport-cc\r
|
||||
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time/r/n
|
||||
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/r/n
|
||||
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r
|
||||
a=extmap:13 urn:3gpp:video-orientation\r
|
||||
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay/r/na=setup:passive/r/n
|
||||
a=mid:${bundle2}\r
|
||||
a=inactive\r
|
||||
a=ice-ufrag:${ICE_UFRAG}\r
|
||||
a=ice-pwd:${ICE_PWD}\r
|
||||
a=fingerprint:${FINGERPRINT}\r
|
||||
a=candidate:${candidate}\r
|
||||
a=rtcp-mux\r
|
||||
`;
|
||||
|
||||
return build;
|
||||
}
|
||||
counter?:string;
|
||||
negotationneeded(){
|
||||
if(this.pc&&this.offer){
|
||||
const pc=this.pc;
|
||||
pc.addEventListener("negotiationneeded", async ()=>{
|
||||
this.offer=(await pc.createOffer({
|
||||
offerToReceiveAudio: true,
|
||||
offerToReceiveVideo: true
|
||||
})).sdp;
|
||||
await pc.setLocalDescription({sdp:this.offer});
|
||||
const ld=pc.localDescription;
|
||||
if(!ld) throw new Error("localDescription isn't defined");
|
||||
if(!this.counter) throw new Error("localDescription isn't defined");
|
||||
const counter=this.counter;
|
||||
const parsed = Voice.parsesdp(ld.sdp);
|
||||
const group=parsed.atr.get("group");
|
||||
if(!group) throw new Error("group isn't in sdp");
|
||||
const groupings=(group.entries().next().value as [string, string])[0].split(" ") as [string,string,string];
|
||||
groupings[2]=groupings[2].replace("\r","");
|
||||
console.log(groupings);
|
||||
const remote:{sdp:string,type:RTCSdpType}={sdp:this.cleanServerSDP(counter,groupings[1],groupings[2]),type:"answer"};
|
||||
console.log(remote);
|
||||
await pc.setRemoteDescription(remote);
|
||||
const senders=this.senders.difference(this.ssrcMap);
|
||||
for(const sender of senders){
|
||||
for(const thing of (await sender.getStats())){
|
||||
console.log(thing[1]);
|
||||
if(thing[1].ssrc){
|
||||
this.ssrcMap.set(sender,thing[1].ssrc);
|
||||
this.makeOp12(sender)
|
||||
}
|
||||
}
|
||||
}
|
||||
console.log(this.ssrcMap);
|
||||
});
|
||||
}
|
||||
}
|
||||
async makeOp12(sender:RTCRtpSender){
|
||||
if(this.ws){
|
||||
this.ws.send(JSON.stringify({
|
||||
op: 12,
|
||||
d: {
|
||||
audio_ssrc: this.ssrcMap.get(sender),
|
||||
video_ssrc: 0,
|
||||
rtx_ssrc: 0,
|
||||
streams: [
|
||||
{
|
||||
type: "video",
|
||||
rid: "100",
|
||||
ssrc: 0,//TODO
|
||||
active: false,
|
||||
quality: 100,
|
||||
rtx_ssrc: 0,//TODO
|
||||
max_bitrate: 2500000,//TODO
|
||||
max_framerate: 0,//TODO
|
||||
max_resolution: {
|
||||
type: "fixed",
|
||||
width: 0,//TODO
|
||||
height: 0//TODO
|
||||
}
|
||||
}
|
||||
]
|
||||
}
|
||||
}))
|
||||
}
|
||||
}
|
||||
senders:Set<RTCRtpSender>=new Set();
|
||||
ssrcMap:Map<RTCRtpSender,string>=new Map();
|
||||
async continueWebRTC(data:sdpback){
|
||||
if(this.pc&&this.offer){
|
||||
const pc=this.pc;
|
||||
this.negotationneeded();
|
||||
const audioStream = await navigator.mediaDevices.getUserMedia({video: false, audio: true} );
|
||||
for (const track of audioStream.getTracks())
|
||||
{
|
||||
//Add track
|
||||
const sender = pc.addTrack(track,audioStream);
|
||||
this.senders.add(sender);
|
||||
}
|
||||
this.counter=data.d.sdp;
|
||||
pc.ontrack = ({ streams: [stream] }) => console.log("got audio stream", stream);
|
||||
|
||||
}
|
||||
}
|
||||
async startWebRTC(){
|
||||
const pc = new RTCPeerConnection();
|
||||
this.pc=pc;
|
||||
const offer = await pc.createOffer({
|
||||
offerToReceiveAudio: true,
|
||||
offerToReceiveVideo: true
|
||||
});
|
||||
const sdp=offer.sdp;
|
||||
this.offer=sdp;
|
||||
|
||||
if(!sdp){this.ws?.close();return;}
|
||||
const parsed=Voice.parsesdp(sdp);
|
||||
const video=new Map<string,[number,number]>();
|
||||
const audio=new Map<string,number>();
|
||||
let cur:[number,number]|undefined;
|
||||
let i=0;
|
||||
for(const thing of parsed.medias){
|
||||
try{
|
||||
if(thing.media==="video"){
|
||||
const rtpmap=thing.atr.get("rtpmap");
|
||||
if(!rtpmap) continue;
|
||||
for(const codecpair of rtpmap){
|
||||
|
||||
const [port, codec]=codecpair.split(" ");
|
||||
if(cur&&codec.split("/")[0]==="rtx"){
|
||||
cur[1]=Number(port);
|
||||
cur=undefined;
|
||||
continue
|
||||
}
|
||||
if(video.has(codec.split("/")[0])) continue;
|
||||
cur=[Number(port),-1];
|
||||
video.set(codec.split("/")[0],cur);
|
||||
}
|
||||
}else if(thing.media==="audio"){
|
||||
const rtpmap=thing.atr.get("rtpmap");
|
||||
if(!rtpmap) continue;
|
||||
for(const codecpair of rtpmap){
|
||||
const [port, codec]=codecpair.split(" ");
|
||||
if(audio.has(codec.split("/")[0])) { continue};
|
||||
audio.set(codec.split("/")[0],Number(port));
|
||||
}
|
||||
}
|
||||
}finally{
|
||||
i++;
|
||||
}
|
||||
}
|
||||
|
||||
const codecs:{
|
||||
name: string,
|
||||
type: "video"|"audio",
|
||||
priority: number,
|
||||
payload_type: number,
|
||||
rtx_payload_type: number|null
|
||||
}[]=[];
|
||||
const include=new Set<string>();
|
||||
const audioAlloweds=new Map([["opus",{priority:1000,}]]);
|
||||
for(const thing of audio){
|
||||
if(audioAlloweds.has(thing[0])){
|
||||
include.add(thing[0]);
|
||||
codecs.push({
|
||||
name:thing[0],
|
||||
type:"audio",
|
||||
priority:audioAlloweds.get(thing[0])?.priority as number,
|
||||
payload_type:thing[1],
|
||||
rtx_payload_type:null
|
||||
});
|
||||
}
|
||||
}
|
||||
const videoAlloweds=new Map([["H264",{priority:1000}],["VP8",{priority:2000}],["VP9",{priority:3000}]]);
|
||||
for(const thing of video){
|
||||
if(videoAlloweds.has(thing[0])){
|
||||
include.add(thing[0]);
|
||||
codecs.push({
|
||||
name:thing[0],
|
||||
type:"video",
|
||||
priority:videoAlloweds.get(thing[0])?.priority as number,
|
||||
payload_type:thing[1][0],
|
||||
rtx_payload_type:thing[1][1]
|
||||
});
|
||||
}
|
||||
}
|
||||
let sendsdp="a=extmap-allow-mixed";
|
||||
let first=true;
|
||||
for(const media of parsed.medias){
|
||||
|
||||
for(const thing of first?["ice-ufrag","ice-pwd","ice-options","fingerprint","extmap","rtpmap"]:["extmap","rtpmap"]){
|
||||
const thing2=media.atr.get(thing);
|
||||
if(!thing2) continue;
|
||||
for(const thing3 of thing2){
|
||||
if(thing === "rtpmap"){
|
||||
const name=thing3.split(" ")[1].split("/")[0];
|
||||
if(include.has(name)){
|
||||
include.delete(name);
|
||||
}else{
|
||||
continue;
|
||||
}
|
||||
}
|
||||
sendsdp+=`\na=${thing}:${thing3}`;
|
||||
}
|
||||
}
|
||||
first=false;
|
||||
}
|
||||
if(this.ws){
|
||||
this.ws.send(JSON.stringify({
|
||||
d:{
|
||||
codecs,
|
||||
protocol:"webrtc",
|
||||
data:sendsdp,
|
||||
sdp:sendsdp
|
||||
},
|
||||
op:1
|
||||
}));
|
||||
}
|
||||
}
|
||||
static parsesdp(sdp:string){
|
||||
let currentA=new Map<string,Set<string>>();
|
||||
const out:{version?:number,medias:{media:string,port:number,proto:string,ports:number[],atr:Map<string,Set<string>>}[],atr:Map<string,Set<string>>}={medias:[],atr:currentA};
|
||||
for(const line of sdp.split("\n")){
|
||||
const [code,setinfo]=line.split("=");
|
||||
switch(code){
|
||||
case "v":
|
||||
out.version=Number(setinfo);
|
||||
break;
|
||||
case "o":
|
||||
case "s":
|
||||
case "t":
|
||||
break;
|
||||
case "m":
|
||||
currentA=new Map();
|
||||
const [media,port,proto,...ports]=setinfo.split(" ");
|
||||
const portnums=ports.map(Number);
|
||||
out.medias.push({media,port:Number(port),proto,ports:portnums,atr:currentA});
|
||||
break;
|
||||
case "a":
|
||||
const [key, ...value] = setinfo.split(":");
|
||||
if(!currentA.has(key)){
|
||||
currentA.set(key,new Set());
|
||||
}
|
||||
currentA.get(key)?.add(value.join(":"));
|
||||
break;
|
||||
}
|
||||
}
|
||||
return out;
|
||||
}
|
||||
async join(){
|
||||
console.warn("Joining");
|
||||
const json = await this.localuser.joinVoice(this);
|
||||
if(!json) return;
|
||||
if(!Voice.url){
|
||||
await Voice.geturl;
|
||||
}
|
||||
if(this.localuser.currentVoice!==this){return}
|
||||
const ws=new WebSocket("ws://"+Voice.url as string);
|
||||
this.ws=ws;
|
||||
ws.addEventListener("message",(m)=>{
|
||||
this.packet(m);
|
||||
})
|
||||
await new Promise<void>(res=>{
|
||||
ws.addEventListener("open",()=>{
|
||||
res()
|
||||
})
|
||||
});
|
||||
ws.send(JSON.stringify({
|
||||
"op": 0,
|
||||
"d": {
|
||||
server_id: this.guild.id,
|
||||
user_id: json.d.user_id,
|
||||
session_id: json.d.session_id,
|
||||
token: json.d.token,
|
||||
video: false,
|
||||
"streams": [
|
||||
{
|
||||
type: "video",
|
||||
rid: "100",
|
||||
quality: 100
|
||||
}
|
||||
]
|
||||
}
|
||||
}));
|
||||
/*
|
||||
const pc=new RTCPeerConnection();
|
||||
this.pc=pc;
|
||||
//pc.setRemoteDescription({sdp:json.d.token,type:""})
|
||||
*/
|
||||
}
|
||||
}
|
||||
export {Voice};
|
||||
Loading…
Add table
Add a link
Reference in a new issue