diff --git a/src/gateway/opcodes/GuildSubscriptionsBulk.ts b/src/gateway/opcodes/GuildSubscriptionsBulk.ts
index 54fe78e24..3c1bdbd5b 100644
--- a/src/gateway/opcodes/GuildSubscriptionsBulk.ts
+++ b/src/gateway/opcodes/GuildSubscriptionsBulk.ts
@@ -3,10 +3,7 @@ import { onLazyRequest } from "./LazyRequest";
import { GuildSubscriptionsBulkSchema } from "@spacebar/schemas";
import { check } from "./instanceOf";
-export async function onGuildSubscriptionsBulk(
- this: WebSocket,
- payload: Payload,
-) {
+export async function onGuildSubscriptionsBulk(this: WebSocket, payload: Payload) {
const startTime = Date.now();
check.call(this, GuildSubscriptionsBulkSchema, payload.d);
const body = payload.d as GuildSubscriptionsBulkSchema;
@@ -22,7 +19,5 @@ export async function onGuildSubscriptionsBulk(
},
});
}
- console.log(
- `[Gateway] GuildSubscriptionsBulk processed ${Object.keys(body.subscriptions).length} subscriptions for user ${this.user_id} in ${Date.now() - startTime}ms`,
- );
+ console.log(`[Gateway] GuildSubscriptionsBulk processed ${Object.keys(body.subscriptions).length} subscriptions for user ${this.user_id} in ${Date.now() - startTime}ms`);
}
diff --git a/src/gateway/opcodes/RequestGuildMembers.ts b/src/gateway/opcodes/RequestGuildMembers.ts
index 034dce6b2..9807e3de4 100644
--- a/src/gateway/opcodes/RequestGuildMembers.ts
+++ b/src/gateway/opcodes/RequestGuildMembers.ts
@@ -16,18 +16,11 @@
along with this program. If not, see .
*/
-import {
- getDatabase,
- getPermission,
- GuildMembersChunkEvent,
- Member,
- Presence,
- Session,
-} from "@spacebar/util";
+import { getDatabase, getPermission, GuildMembersChunkEvent, Member, Presence, Session } from "@spacebar/util";
import { WebSocket, Payload, OPCODES, Send } from "@spacebar/gateway";
import { check } from "./instanceOf";
import { FindManyOptions, ILike, In } from "typeorm";
-import { RequestGuildMembersSchema } from "@spacebar/schemas"
+import { RequestGuildMembersSchema } from "@spacebar/schemas";
export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
const startTime = Date.now();
@@ -39,11 +32,7 @@ export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
check.call(this, RequestGuildMembersSchema, d);
- const {
- presences,
- nonce,
- query: requestQuery,
- } = d as RequestGuildMembersSchema;
+ const { presences, nonce, query: requestQuery } = d as RequestGuildMembersSchema;
let { limit, user_ids, guild_id } = d as RequestGuildMembersSchema;
// some discord libraries send empty string as query when they meant to send undefined, which was leading to errors being thrown in this handler
@@ -63,8 +52,7 @@ export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
}
// TODO: Configurable limit?
- if ((query || (user_ids && user_ids.length > 0)) && (!limit || limit > 100))
- limit = 100;
+ if ((query || (user_ids && user_ids.length > 0)) && (!limit || limit > 100)) limit = 100;
const permissions = await getPermission(this.user_id, guild_id);
permissions.hasThrow("VIEW_CHANNEL");
@@ -107,10 +95,7 @@ export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
.leftJoinAndSelect("member.roles", "role")
.leftJoinAndSelect("member.user", "user")
.leftJoinAndSelect("user.sessions", "session")
- .andWhere(
- "',' || member.roles || ',' NOT LIKE :everyoneRoleIdList",
- { everyoneRoleIdList: "%," + guild_id + ",%" },
- )
+ .andWhere("',' || member.roles || ',' NOT LIKE :everyoneRoleIdList", { everyoneRoleIdList: "%," + guild_id + ",%" })
.addOrderBy("user.username", "ASC")
.limit(memberFind.take);
@@ -145,10 +130,7 @@ export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
const chunkCount = Math.ceil(members.length / 1000);
let notFound: string[] = [];
- if (user_ids && user_ids.length > 0)
- notFound = user_ids.filter(
- (id) => !members.some((member) => member.id == id),
- );
+ if (user_ids && user_ids.length > 0) notFound = user_ids.filter((id) => !members.some((member) => member.id == id));
const chunks: GuildMembersChunkEvent["data"][] = [];
while (members.length > 0) {
@@ -202,7 +184,5 @@ export async function onRequestGuildMembers(this: WebSocket, { d }: Payload) {
});
});
- console.log(
- `[Gateway] REQUEST_GUILD_MEMBERS took ${Date.now() - startTime}ms for guild ${guild_id} with ${members.length} members`,
- );
+ console.log(`[Gateway] REQUEST_GUILD_MEMBERS took ${Date.now() - startTime}ms for guild ${guild_id} with ${members.length} members`);
}
diff --git a/src/gateway/opcodes/StreamCreate.ts b/src/gateway/opcodes/StreamCreate.ts
index 3defc1232..00c8c947c 100644
--- a/src/gateway/opcodes/StreamCreate.ts
+++ b/src/gateway/opcodes/StreamCreate.ts
@@ -1,9 +1,4 @@
-import {
- genVoiceToken,
- Payload,
- WebSocket,
- generateStreamKey,
-} from "@spacebar/gateway";
+import { genVoiceToken, Payload, WebSocket, generateStreamKey } from "@spacebar/gateway";
import {
Channel,
Config,
@@ -18,7 +13,7 @@ import {
VoiceStateUpdateEvent,
} from "@spacebar/util";
import { check } from "./instanceOf";
-import { StreamCreateSchema } from "@spacebar/schemas"
+import { StreamCreateSchema } from "@spacebar/schemas";
export async function onStreamCreate(this: WebSocket, data: Payload) {
const startTime = Date.now();
@@ -47,17 +42,11 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
where: { id: body.channel_id },
});
- if (
- !channel ||
- (body.type === "guild" && channel.guild_id != body.guild_id)
- )
- return this.close(4000, "invalid channel");
+ if (!channel || (body.type === "guild" && channel.guild_id != body.guild_id)) return this.close(4000, "invalid channel");
// TODO: actually apply preferred_region from the event payload
const regions = Config.get().regions;
- const guildRegion = regions.available.filter(
- (r) => r.id === regions.default,
- )[0];
+ const guildRegion = regions.available.filter((r) => r.id === regions.default)[0];
// first make sure theres no other streams for this user that somehow didnt get cleared
await Stream.delete({
@@ -85,12 +74,7 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
await streamSession.save();
- const streamKey = generateStreamKey(
- body.type,
- body.guild_id,
- body.channel_id,
- this.user_id,
- );
+ const streamKey = generateStreamKey(body.type, body.guild_id, body.channel_id, this.user_id);
await emitEvent({
event: "STREAM_CREATE",
@@ -125,9 +109,7 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
channel_id: voiceState.channel_id,
} as VoiceStateUpdateEvent);
- console.log(
- `[Gateway] STREAM_CREATE for user ${this.user_id} in channel ${body.channel_id} with stream key ${streamKey} in ${Date.now() - startTime}ms`,
- );
+ console.log(`[Gateway] STREAM_CREATE for user ${this.user_id} in channel ${body.channel_id} with stream key ${streamKey} in ${Date.now() - startTime}ms`);
}
//stream key:
diff --git a/src/gateway/opcodes/StreamDelete.ts b/src/gateway/opcodes/StreamDelete.ts
index 069d6b2a8..34a3b9df1 100644
--- a/src/gateway/opcodes/StreamDelete.ts
+++ b/src/gateway/opcodes/StreamDelete.ts
@@ -1,13 +1,7 @@
import { parseStreamKey, Payload, WebSocket } from "@spacebar/gateway";
-import {
- emitEvent,
- Stream,
- StreamDeleteEvent,
- VoiceState,
- VoiceStateUpdateEvent,
-} from "@spacebar/util";
+import { emitEvent, Stream, StreamDeleteEvent, VoiceState, VoiceStateUpdateEvent } from "@spacebar/util";
import { check } from "./instanceOf";
-import { StreamDeleteSchema } from "@spacebar/schemas"
+import { StreamDeleteSchema } from "@spacebar/schemas";
export async function onStreamDelete(this: WebSocket, data: Payload) {
const startTime = Date.now();
@@ -75,7 +69,5 @@ export async function onStreamDelete(this: WebSocket, data: Payload) {
channel_id: channelId,
} as StreamDeleteEvent);
- console.log(
- `[Gateway] STREAM_DELETE for user ${this.user_id} in channel ${channelId} with stream key ${body.stream_key} in ${Date.now() - startTime}ms`,
- );
+ console.log(`[Gateway] STREAM_DELETE for user ${this.user_id} in channel ${channelId} with stream key ${body.stream_key} in ${Date.now() - startTime}ms`);
}
diff --git a/src/gateway/opcodes/StreamWatch.ts b/src/gateway/opcodes/StreamWatch.ts
index 88bc0463b..3bb1ce779 100644
--- a/src/gateway/opcodes/StreamWatch.ts
+++ b/src/gateway/opcodes/StreamWatch.ts
@@ -1,20 +1,8 @@
-import {
- genVoiceToken,
- parseStreamKey,
- Payload,
- WebSocket,
-} from "@spacebar/gateway";
-import {
- Config,
- emitEvent,
- Stream,
- StreamCreateEvent,
- StreamServerUpdateEvent,
- StreamSession,
-} from "@spacebar/util";
+import { genVoiceToken, parseStreamKey, Payload, WebSocket } from "@spacebar/gateway";
+import { Config, emitEvent, Stream, StreamCreateEvent, StreamServerUpdateEvent, StreamSession } from "@spacebar/util";
import { check } from "./instanceOf";
import { Not } from "typeorm";
-import { StreamWatchSchema } from "@spacebar/schemas"
+import { StreamWatchSchema } from "@spacebar/schemas";
export async function onStreamWatch(this: WebSocket, data: Payload) {
const startTime = Date.now();
@@ -45,13 +33,10 @@ export async function onStreamWatch(this: WebSocket, data: Payload) {
if (!stream) return this.close(4000, "Invalid stream key");
- if (type === "guild" && stream.channel.guild_id != guildId)
- return this.close(4000, "Invalid stream key");
+ if (type === "guild" && stream.channel.guild_id != guildId) return this.close(4000, "Invalid stream key");
const regions = Config.get().regions;
- const guildRegion = regions.available.find(
- (r) => r.endpoint === stream.endpoint,
- );
+ const guildRegion = regions.available.find((r) => r.endpoint === stream.endpoint);
if (!guildRegion) return this.close(4000, "Unknown region");
@@ -97,7 +82,5 @@ export async function onStreamWatch(this: WebSocket, data: Payload) {
user_id: this.user_id,
} as StreamServerUpdateEvent);
- console.log(
- `[Gateway] STREAM_WATCH for user ${this.user_id} in channel ${channelId} with stream key ${body.stream_key} in ${Date.now() - startTime}ms`,
- );
+ console.log(`[Gateway] STREAM_WATCH for user ${this.user_id} in channel ${channelId} with stream key ${body.stream_key} in ${Date.now() - startTime}ms`);
}
diff --git a/src/gateway/opcodes/VoiceStateUpdate.ts b/src/gateway/opcodes/VoiceStateUpdate.ts
index 7d1a80e9e..a2922e0ce 100644
--- a/src/gateway/opcodes/VoiceStateUpdate.ts
+++ b/src/gateway/opcodes/VoiceStateUpdate.ts
@@ -17,19 +17,10 @@
*/
import { Payload, WebSocket } from "@spacebar/gateway";
-import {
- Config,
- emitEvent,
- Guild,
- Member,
- Region,
- VoiceServerUpdateEvent,
- VoiceState,
- VoiceStateUpdateEvent,
-} from "@spacebar/util";
+import { Config, emitEvent, Guild, Member, Region, VoiceServerUpdateEvent, VoiceState, VoiceStateUpdateEvent } from "@spacebar/util";
import { genVoiceToken } from "../util/SessionUtils";
import { check } from "./instanceOf";
-import { VoiceStateUpdateSchema } from "@spacebar/schemas"
+import { VoiceStateUpdateSchema } from "@spacebar/schemas";
// TODO: check if a voice server is setup
// Notice: Bot users respect the voice channel's user limit, if set.
@@ -50,10 +41,7 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
voiceState = await VoiceState.findOneOrFail({
where: { user_id: this.user_id },
});
- if (
- voiceState.session_id !== this.session_id &&
- body.channel_id === null
- ) {
+ if (voiceState.session_id !== this.session_id && body.channel_id === null) {
//Should we also check guild_id === null?
//changing deaf or mute on a client that's not the one with the same session of the voicestate in the database should be ignored
return;
@@ -62,11 +50,7 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
if (voiceState.channel_id !== body.channel_id) isChanged = true;
//If a user change voice channel between guild we should send a left event first
- if (
- voiceState.guild_id &&
- voiceState.guild_id !== body.guild_id &&
- voiceState.session_id === this.session_id
- ) {
+ if (voiceState.guild_id && voiceState.guild_id !== body.guild_id && voiceState.session_id === this.session_id) {
await emitEvent({
event: "VOICE_STATE_UPDATE",
data: { ...voiceState.toPublicVoiceState(), channel_id: null },
@@ -89,12 +73,7 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
}
// if user left voice channel, send an update to previous channel/guild to let other people know that the user left
- if (
- voiceState.session_id === this.session_id &&
- body.guild_id == null &&
- body.channel_id == null &&
- (prevState?.guild_id || prevState?.channel_id)
- ) {
+ if (voiceState.session_id === this.session_id && body.guild_id == null && body.channel_id == null && (prevState?.guild_id || prevState?.channel_id)) {
await emitEvent({
event: "VOICE_STATE_UPDATE",
data: {
@@ -118,8 +97,7 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
}
//If the session changed we generate a new token
- if (voiceState.session_id !== this.session_id)
- voiceState.token = genVoiceToken();
+ if (voiceState.session_id !== this.session_id) voiceState.token = genVoiceToken();
voiceState.session_id = this.session_id;
const { member } = voiceState;
@@ -146,13 +124,9 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
const regions = Config.get().regions;
let guildRegion: Region;
if (guild && guild.region) {
- guildRegion = regions.available.filter(
- (r) => r.id === guild.region,
- )[0];
+ guildRegion = regions.available.filter((r) => r.id === guild.region)[0];
} else {
- guildRegion = regions.available.filter(
- (r) => r.id === regions.default,
- )[0];
+ guildRegion = regions.available.filter((r) => r.id === regions.default)[0];
}
await emitEvent({
@@ -161,15 +135,11 @@ export async function onVoiceStateUpdate(this: WebSocket, data: Payload) {
token: voiceState.token,
guild_id: voiceState.guild_id,
endpoint: guildRegion.endpoint,
- channel_id: voiceState.guild_id
- ? undefined
- : voiceState.channel_id, // only DM voice calls have this set, and DM channel is one where guild_id is null
+ channel_id: voiceState.guild_id ? undefined : voiceState.channel_id, // only DM voice calls have this set, and DM channel is one where guild_id is null
},
user_id: voiceState.user_id,
} as VoiceServerUpdateEvent);
}
- console.log(
- `[Gateway] VOICE_STATE_UPDATE for user ${this.user_id} in channel ${voiceState.channel_id} in guild ${voiceState.guild_id} in ${Date.now() - startTime}ms`,
- );
+ console.log(`[Gateway] VOICE_STATE_UPDATE for user ${this.user_id} in channel ${voiceState.channel_id} in guild ${voiceState.guild_id} in ${Date.now() - startTime}ms`);
}
diff --git a/src/webrtc/Server.ts b/src/webrtc/Server.ts
index 3a270e68d..a20531d9e 100644
--- a/src/webrtc/Server.ts
+++ b/src/webrtc/Server.ts
@@ -21,13 +21,7 @@ import { closeDatabase, Config, initDatabase, initEvent } from "@spacebar/util";
import http from "http";
import ws from "ws";
import { Connection } from "./events/Connection";
-import {
- loadWebRtcLibrary,
- mediaServer,
- WRTC_PORT_MAX,
- WRTC_PORT_MIN,
- WRTC_PUBLIC_IP,
-} from "./util/MediaServer";
+import { loadWebRtcLibrary, mediaServer, WRTC_PORT_MAX, WRTC_PORT_MIN, WRTC_PUBLIC_IP } from "./util/MediaServer";
import { green, yellow } from "picocolors";
export class Server {
@@ -36,15 +30,7 @@ export class Server {
public server: http.Server;
public production: boolean;
- constructor({
- port,
- server,
- production,
- }: {
- port: number;
- server?: http.Server;
- production?: boolean;
- }) {
+ constructor({ port, server, production }: { port: number; server?: http.Server; production?: boolean }) {
this.port = port;
this.production = production || false;
diff --git a/src/webrtc/events/Connection.ts b/src/webrtc/events/Connection.ts
index a068a8fd8..6e028c647 100644
--- a/src/webrtc/events/Connection.ts
+++ b/src/webrtc/events/Connection.ts
@@ -28,11 +28,7 @@ import { onMessage } from "./Message";
// TODO: specify rate limit in config
// TODO: check msg max size
-export async function Connection(
- this: WS.Server,
- socket: WebRtcWebSocket,
- request: IncomingMessage,
-) {
+export async function Connection(this: WS.Server, socket: WebRtcWebSocket, request: IncomingMessage) {
try {
socket.on("close", onClose.bind(socket));
socket.on("message", onMessage.bind(socket));
@@ -57,8 +53,7 @@ export async function Connection(
socket.encoding = "json";
socket.version = Number(searchParams.get("v")) || 5;
- if (socket.version < 3)
- return socket.close(CLOSECODES.Unknown_error, "invalid version");
+ if (socket.version < 3) return socket.close(CLOSECODES.Unknown_error, "invalid version");
setHeartbeat(socket);
diff --git a/src/webrtc/events/Message.ts b/src/webrtc/events/Message.ts
index ccb88b72c..932510798 100644
--- a/src/webrtc/events/Message.ts
+++ b/src/webrtc/events/Message.ts
@@ -31,8 +31,7 @@ const PayloadSchema = {
export async function onMessage(this: WebRtcWebSocket, buffer: Buffer) {
try {
const data: VoicePayload = JSON.parse(buffer.toString());
- if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id)
- return this.close(CLOSECODES.Not_authenticated);
+ if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id) return this.close(CLOSECODES.Not_authenticated);
const OPCodeHandler = OPCodeHandlers[data.op];
if (!OPCodeHandler) {
@@ -42,11 +41,7 @@ export async function onMessage(this: WebRtcWebSocket, buffer: Buffer) {
return;
}
- if (
- ![VoiceOPCodes.HEARTBEAT, VoiceOPCodes.SPEAKING].includes(
- data.op as VoiceOPCodes,
- )
- ) {
+ if (![VoiceOPCodes.HEARTBEAT, VoiceOPCodes.SPEAKING].includes(data.op as VoiceOPCodes)) {
console.log("[WebRTC] Opcode " + VoiceOPCodes[data.op]);
}
diff --git a/src/webrtc/opcodes/Identify.ts b/src/webrtc/opcodes/Identify.ts
index 3abb26d62..8a2516c42 100644
--- a/src/webrtc/opcodes/Identify.ts
+++ b/src/webrtc/opcodes/Identify.ts
@@ -17,29 +17,15 @@
*/
import { CLOSECODES } from "@spacebar/gateway";
-import {
- StreamSession,
- VoiceState,
-} from "@spacebar/util";
-import {
- validateSchema,
- VoiceIdentifySchema,
-} from "@spacebar/schemas";
-import {
- generateSsrc,
- mediaServer,
- Send,
- VoiceOPCodes,
- VoicePayload,
- WebRtcWebSocket,
-} from "@spacebar/webrtc";
+import { StreamSession, VoiceState } from "@spacebar/util";
+import { validateSchema, VoiceIdentifySchema } from "@spacebar/schemas";
+import { generateSsrc, mediaServer, Send, VoiceOPCodes, VoicePayload, WebRtcWebSocket } from "@spacebar/webrtc";
import { SSRCs } from "@spacebarchat/spacebar-webrtc-types";
import { subscribeToProducers } from "./Video";
export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
clearTimeout(this.readyTimeout);
- const { server_id, user_id, session_id, token, streams, video } =
- validateSchema("VoiceIdentifySchema", data.d) as VoiceIdentifySchema;
+ const { server_id, user_id, session_id, token, streams, video } = validateSchema("VoiceIdentifySchema", data.d) as VoiceIdentifySchema;
// server_id can be one of the following: a unique id for a GO Live stream, a channel id for a DM voice call, or a guild id for a guild voice channel
// not sure if there's a way to determine whether a snowflake is a channel id or a guild id without checking if it exists in db
@@ -92,12 +78,7 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
this.type = type;
const voiceRoomId = type === "stream" ? server_id : voiceState!.channel_id;
- this.webRtcClient = await mediaServer.join(
- voiceRoomId,
- this.user_id,
- this,
- type!,
- );
+ this.webRtcClient = await mediaServer.join(voiceRoomId, this.user_id, this, type!);
this.on("close", () => {
// ice-lite media server relies on this to know when the peer went away
diff --git a/src/webrtc/opcodes/SelectProtocol.ts b/src/webrtc/opcodes/SelectProtocol.ts
index 2aa841f20..a29bcaccd 100644
--- a/src/webrtc/opcodes/SelectProtocol.ts
+++ b/src/webrtc/opcodes/SelectProtocol.ts
@@ -16,34 +16,17 @@
along with this program. If not, see .
*/
import { SelectProtocolSchema, validateSchema } from "@spacebar/schemas";
-import {
- VoiceOPCodes,
- VoicePayload,
- WebRtcWebSocket,
- mediaServer,
- Send,
-} from "@spacebar/webrtc";
+import { VoiceOPCodes, VoicePayload, WebRtcWebSocket, mediaServer, Send } from "@spacebar/webrtc";
-export async function onSelectProtocol(
- this: WebRtcWebSocket,
- payload: VoicePayload,
-) {
+export async function onSelectProtocol(this: WebRtcWebSocket, payload: VoicePayload) {
if (!this.webRtcClient) return;
- const data = validateSchema(
- "SelectProtocolSchema",
- payload.d,
- ) as SelectProtocolSchema;
+ const data = validateSchema("SelectProtocolSchema", payload.d) as SelectProtocolSchema;
// UDP protocol not currently supported. Maybe in the future?
- if (data.protocol !== "webrtc")
- return this.close(4000, "only webrtc protocol supported currently");
+ if (data.protocol !== "webrtc") return this.close(4000, "only webrtc protocol supported currently");
- const response = await mediaServer.onOffer(
- this.webRtcClient,
- data.sdp!,
- data.codecs ?? [],
- );
+ const response = await mediaServer.onOffer(this.webRtcClient, data.sdp!, data.codecs ?? []);
await Send(this, {
op: VoiceOPCodes.SESSION_DESCRIPTION,
diff --git a/src/webrtc/opcodes/Speaking.ts b/src/webrtc/opcodes/Speaking.ts
index bff0db979..fde63f342 100644
--- a/src/webrtc/opcodes/Speaking.ts
+++ b/src/webrtc/opcodes/Speaking.ts
@@ -16,13 +16,7 @@
along with this program. If not, see .
*/
-import {
- mediaServer,
- VoiceOPCodes,
- VoicePayload,
- WebRtcWebSocket,
- Send,
-} from "../util";
+import { mediaServer, VoiceOPCodes, VoicePayload, WebRtcWebSocket, Send } from "../util";
// {"speaking":1,"delay":5,"ssrc":2805246727}
@@ -30,11 +24,7 @@ export async function onSpeaking(this: WebRtcWebSocket, data: VoicePayload) {
if (!this.webRtcClient) return;
await Promise.all(
- Array.from(
- mediaServer.getClientsForRtcServer(
- this.webRtcClient.voiceRoomId,
- ),
- ).map((client) => {
+ Array.from(mediaServer.getClientsForRtcServer(this.webRtcClient.voiceRoomId)).map((client) => {
if (client.user_id === this.user_id) return Promise.resolve();
const ssrc = client.getOutgoingStreamSSRCsForUser(this.user_id);
diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index 1309fc4fe..ed8870e44 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -16,13 +16,7 @@
along with this program. If not, see .
*/
import { Stream } from "@spacebar/util";
-import {
- mediaServer,
- Send,
- VoiceOPCodes,
- VoicePayload,
- WebRtcWebSocket,
-} from "@spacebar/webrtc";
+import { mediaServer, Send, VoiceOPCodes, VoicePayload, WebRtcWebSocket } from "@spacebar/webrtc";
import type { WebRtcClient } from "@spacebarchat/spacebar-webrtc-types";
import { validateSchema, VoiceVideoSchema } from "@spacebar/schemas";
@@ -60,9 +54,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
try {
await Promise.race([
new Promise((resolve, reject) => {
- this.webRtcClient?.emitter.once("connected", () =>
- resolve(),
- );
+ this.webRtcClient?.emitter.once("connected", () => resolve());
}),
new Promise((resolve, reject) => {
// Reject after 3 seconds if still not connected
@@ -93,28 +85,19 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
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:${d.audio_ssrc}`,
- );
+ console.log(`[${this.user_id}] publishing new audio track ssrc:${d.audio_ssrc}`);
await this.webRtcClient.publishTrack("audio", {
audio_ssrc: d.audio_ssrc,
});
}
// now check that all clients have subscribed to our audio
- for (const client of mediaServer.getClientsForRtcServer(
- voiceRoomId,
- )) {
+ for (const client of mediaServer.getClientsForRtcServer(voiceRoomId)) {
if (client.user_id === this.user_id) continue;
if (!client.isSubscribedToTrack(this.user_id, "audio")) {
- console.log(
- `[${client.user_id}] subscribing to audio track ssrcs: ${d.audio_ssrc}`,
- );
- await client.subscribeToTrack(
- this.webRtcClient.user_id,
- "audio",
- );
+ console.log(`[${client.user_id}] subscribing to audio track ssrcs: ${d.audio_ssrc}`);
+ await client.subscribeToTrack(this.webRtcClient.user_id, "audio");
clientsThatNeedUpdate.add(client);
}
@@ -125,9 +108,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
this.webRtcClient!.videoStream = { ...stream, type: "video" }; // client sends "screen" on go live but expects "video" on response
// check if we are already publishing video, if not, publish a new video track for it
if (!this.webRtcClient!.isProducingVideo()) {
- console.log(
- `[${this.user_id}] publishing new video track ssrc:${d.video_ssrc}`,
- );
+ console.log(`[${this.user_id}] publishing new video track ssrc:${d.video_ssrc}`);
await this.webRtcClient.publishTrack("video", {
video_ssrc: d.video_ssrc,
rtx_ssrc: d.rtx_ssrc,
@@ -135,19 +116,12 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
}
// now check that all clients have subscribed to our video track
- for (const client of mediaServer.getClientsForRtcServer(
- voiceRoomId,
- )) {
+ for (const client of mediaServer.getClientsForRtcServer(voiceRoomId)) {
if (client.user_id === this.user_id) continue;
if (!client.isSubscribedToTrack(this.user_id, "video")) {
- console.log(
- `[${client.user_id}] subscribing to video track ssrc: ${d.video_ssrc}`,
- );
- await client.subscribeToTrack(
- this.webRtcClient.user_id,
- "video",
- );
+ console.log(`[${client.user_id}] subscribing to video track ssrc: ${d.video_ssrc}`);
+ await client.subscribeToTrack(this.webRtcClient.user_id, "video");
clientsThatNeedUpdate.add(client);
}
@@ -163,9 +137,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
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,
+ 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) => ({
@@ -181,14 +153,10 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
}
// check if we are not subscribed to producers in this server, if not, subscribe
-export async function subscribeToProducers(
- this: WebRtcWebSocket,
-): Promise {
+export async function subscribeToProducers(this: WebRtcWebSocket): Promise {
if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
- const clients = mediaServer.getClientsForRtcServer(
- this.webRtcClient.voiceRoomId,
- );
+ const clients = mediaServer.getClientsForRtcServer(this.webRtcClient.voiceRoomId);
await Promise.all(
Array.from(clients).map(async (client) => {
@@ -196,42 +164,26 @@ export async function subscribeToProducers(
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",
- );
+ 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",
- );
+ 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,
- );
+ 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,
+ audio_ssrc: ssrcs.audio_ssrc ?? client.getIncomingStreamSSRCs().audio_ssrc,
video_ssrc: ssrcs.video_ssrc ?? 0,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
streams: [