2 files changed, 22 insertions, 4 deletions
diff --git a/src/webrtc/util/Send.ts b/src/webrtc/util/Send.ts
index 1d8aa61e..c3f47bae 100644
--- a/src/webrtc/util/Send.ts
+++ b/src/webrtc/util/Send.ts
@@ -1,3 +1,21 @@
+/*
+ Spacebar: A FOSS re-implementation and extension of the Discord.com backend.
+ Copyright (C) 2025 Spacebar and Spacebar Contributors
+
+ This program is free software: you can redistribute it and/or modify
+ it under the terms of the GNU Affero General Public License as published
+ by the Free Software Foundation, either version 3 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU Affero General Public License for more details.
+
+ You should have received a copy of the GNU Affero General Public License
+ along with this program. If not, see <https://www.gnu.org/licenses/>.
+*/
+
import { JSONReplacer } from "@spacebar/util";
import { VoicePayload } from "./Constants";
import { WebRtcWebSocket } from "./WebRtcWebSocket";
@@ -12,13 +30,13 @@ export function Send(socket: WebRtcWebSocket, data: VoicePayload) {
else return;
return new Promise((res, rej) => {
- if (socket.readyState !== 1) {
+ if (socket.rawSocket.readyState !== 1) {
// return rej("socket not open");
- socket.close();
+ socket.rawSocket.close();
return;
}
- socket.send(buffer, (err) => {
+ socket.rawSocket.send(buffer, (err) => {
if (err) return rej(err);
return res(null);
});
diff --git a/src/webrtc/util/WebRtcWebSocket.ts b/src/webrtc/util/WebRtcWebSocket.ts
index e36b4f1d..8d07dfdf 100644
--- a/src/webrtc/util/WebRtcWebSocket.ts
+++ b/src/webrtc/util/WebRtcWebSocket.ts
@@ -1,7 +1,7 @@
import { WebSocket } from "@spacebar/gateway";
import type { WebRtcClient } from "@spacebarchat/spacebar-webrtc-types";
-export interface WebRtcWebSocket extends WebSocket {
+export class WebRtcWebSocket extends WebSocket {
type: "guild-voice" | "dm-voice" | "stream";
webRtcClient?: WebRtcClient<WebRtcWebSocket>;
}
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