diff --git a/src/webrtc/events/Connection.ts b/src/webrtc/events/Connection.ts
index 9300b6b2..210b8b07 100644
--- a/src/webrtc/events/Connection.ts
+++ b/src/webrtc/events/Connection.ts
@@ -5,7 +5,7 @@ import WS from "ws";
import { VoiceOPCodes } from "../util";
import { onClose } from "./Close";
import { onMessage } from "./Message";
-var erlpack: any;
+let erlpack: any;
try {
erlpack = require("@yukikaze-bot/erlpack");
} catch (error) {}
diff --git a/src/webrtc/events/Message.ts b/src/webrtc/events/Message.ts
index 38676f6c..862710bf 100644
--- a/src/webrtc/events/Message.ts
+++ b/src/webrtc/events/Message.ts
@@ -12,7 +12,7 @@ const PayloadSchema = {
export async function onMessage(this: WebSocket, buffer: Buffer) {
try {
- var data: Payload = JSON.parse(buffer.toString());
+ let data: Payload = JSON.parse(buffer.toString());
if (data.op !== VoiceOPCodes.IDENTIFY && !this.user_id)
return this.close(CLOSECODES.Not_authenticated);
diff --git a/src/webrtc/opcodes/Video.ts b/src/webrtc/opcodes/Video.ts
index dcbc9aa0..87d1b57c 100644
--- a/src/webrtc/opcodes/Video.ts
+++ b/src/webrtc/opcodes/Video.ts
@@ -14,7 +14,7 @@ export async function onVideo(this: WebSocket, payload: Payload) {
const id = "stream" + this.user_id;
- var stream = this.client.in.stream!;
+ let stream = this.client.in.stream!;
if (!stream) {
stream = this.client.transport!.createIncomingStream(
// @ts-ignore
@@ -88,7 +88,7 @@ function attachTrack(
);
outTrack.attachTo(track);
this.client.out.stream!.addTrack(outTrack);
- var ssrcs = this.client.out.tracks.get(user_id)!;
+ let ssrcs = this.client.out.tracks.get(user_id)!;
if (!ssrcs)
ssrcs = this.client.out.tracks
.set(user_id, { audio_ssrc: 0, rtx_ssrc: 0, video_ssrc: 0 })
@@ -116,7 +116,7 @@ function handleSSRC(this: WebSocket, type: "audio" | "video", ssrcs: SSRCs) {
const transport = this.client.transport!;
const id = type + ssrcs.media;
- var track = stream.getTrack(id);
+ let track = stream.getTrack(id);
if (!track) {
console.log("createIncomingStreamTrack", id);
track = transport.createIncomingStreamTrack(type, { id, ssrcs });
diff --git a/src/webrtc/util/MediaServer.ts b/src/webrtc/util/MediaServer.ts
index 520b8682..3c6283a7 100644
--- a/src/webrtc/util/MediaServer.ts
+++ b/src/webrtc/util/MediaServer.ts
@@ -11,7 +11,7 @@ export const PublicIP = process.env.PUBLIC_IP || "127.0.0.1";
try {
const range = process.env.WEBRTC_PORT_RANGE || "4000";
- var ports = range.split("-");
+ let ports = range.split("-");
const min = Number(ports[0]);
const max = Number(ports[1]);
|
