changes

package-lock.json

@@ -9377,7 +9377,7 @@
 		},
 		"node_modules/spacebar-webrtc-types": {
 			"version": "1.0.1",
-			"resolved": "git+ssh://git@github.com/dank074/spacebar-webrtc-types.git#04044902e71f2c80fc39c256ec5d27fdb8ca425a",
+			"resolved": "git+ssh://git@github.com/dank074/spacebar-webrtc-types.git#e8e13a32c7c0b446fb5c1b35111591ff1a55b919",
 			"dev": true,
 			"license": "ISC"
 		},

src/webrtc/opcodes/Identify.ts

@@ -40,7 +40,7 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 	// 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
 	// luckily we will only have to determine this once
-	let type: "guild-voice" | "dm-voice" | "stream";
+	let type: "guild-voice" | "dm-voice" | "stream" = "guild-voice";
 	let authenticated = false;
 
 	// first check if its a guild voice connection or DM voice call
@@ -54,6 +54,9 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 	if (voiceState) {
 		type = voiceState.guild_id === server_id ? "guild-voice" : "dm-voice";
 		authenticated = true;
+		this.guild_id =
+			type === "guild-voice" ? voiceState.guild_id : undefined;
+		this.channel_id = voiceState.channel_id;
 	} else {
 		// if its not a guild/dm voice connection, check if it is a go live stream
 		const streamSession = await StreamSession.findOne({
@@ -64,6 +67,7 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 				session_id,
 				used: false,
 			},
+			relations: ["stream"],
 		});
 
 		if (streamSession) {
@@ -72,6 +76,8 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 			streamSession.used = true;
 			await streamSession.save();
 
+			this.channel_id = streamSession.stream.channel_id;
+
 			this.once("close", async () => {
 				await streamSession.remove();
 			});
@@ -84,8 +90,15 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 	this.user_id = user_id;
 	this.session_id = session_id;
 
-	this.type = type!;
+	this.type = type;
-	this.webRtcClient = mediaServer.join(server_id, this.user_id, this, type!);
+
+	const voiceRoomId = type === "stream" ? server_id : voiceState!.channel_id;
+	this.webRtcClient = 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

src/webrtc/opcodes/Speaking.ts

@@ -32,7 +32,7 @@ export async function onSpeaking(this: WebRtcWebSocket, data: VoicePayload) {
 	await Promise.all(
 		Array.from(
 			mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
-				this.webRtcClient.rtc_server_id,
+				this.webRtcClient.voiceRoomId,
 			),
 		).map((client) => {
 			if (client.user_id === this.user_id) return Promise.resolve();

src/webrtc/opcodes/Video.ts

@@ -28,13 +28,13 @@ import type { WebRtcClient } from "spacebar-webrtc-types";
 export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 	if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
 
-	const { rtc_server_id } = this.webRtcClient;
+	const { voiceRoomId } = this.webRtcClient;
 
 	const d = validateSchema("VoiceVideoSchema", payload.d) as VoiceVideoSchema;
 
 	if (this.type === "stream") {
 		const stream = await Stream.findOne({
-			where: { id: rtc_server_id },
+			where: { id: voiceRoomId },
 		});
 
 		if (!stream) return;
@@ -50,9 +50,20 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 	await Send(this, { op: VoiceOPCodes.MEDIA_SINK_WANTS, d: { any: 100 } });
 
 	const clientsThatNeedUpdate = new Set<WebRtcClient<WebRtcWebSocket>>();
+	const wantsToProduceAudio = d.audio_ssrc !== 0;
+	const wantsToProduceVideo = d.video_ssrc !== 0 && stream?.active;
+
+	// first check if we need stop any tracks
+	if (!wantsToProduceAudio && this.webRtcClient.isProducingAudio()) {
+		this.webRtcClient.stopPublishingTrack("audio");
+	}
+
+	if (!wantsToProduceVideo && this.webRtcClient.isProducingVideo()) {
+		this.webRtcClient.stopPublishingTrack("video");
+	}
 
 	// check if client has signaled that it will send audio
-	if (d.audio_ssrc !== 0) {
+	if (wantsToProduceAudio) {
 		// check if we are already producing audio, if not, publish a new audio track for it
 		if (!this.webRtcClient!.isProducingAudio()) {
 			console.log(
@@ -65,7 +76,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 
 		// now check that all clients have subscribed to our audio
 		for (const client of mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
-			rtc_server_id,
+			voiceRoomId,
 		)) {
 			if (client.user_id === this.user_id) continue;
 
@@ -80,7 +91,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 		}
 	}
 	// check if client has signaled that it will send video
-	if (d.video_ssrc !== 0 && stream?.active) {
+	if (wantsToProduceVideo) {
 		this.webRtcClient!.videoStream = stream;
 		// check if we are already publishing video, if not, publish a new video track for it
 		if (!this.webRtcClient!.isProducingVideo()) {
@@ -95,7 +106,7 @@ 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<WebRtcWebSocket>(
-			rtc_server_id,
+			voiceRoomId,
 		)) {
 			if (client.user_id === this.user_id) continue;
 
@@ -136,8 +147,10 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 export async function subscribeToProducers(
 	this: WebRtcWebSocket,
 ): Promise<void> {
+	if (!this.webRtcClient || !this.webRtcClient.webrtcConnected) return;
+
 	const clients = mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
-		this.webRtcClient!.rtc_server_id,
+		this.webRtcClient.voiceRoomId,
 	);
 
 	await Promise.all(

src/webrtc/util/WebRtcWebSocket.ts

@@ -3,5 +3,7 @@ import type { WebRtcClient } from "spacebar-webrtc-types";
 
 export interface WebRtcWebSocket extends WebSocket {
 	type: "guild-voice" | "dm-voice" | "stream";
+	guild_id?: string;
+	channel_id: string;
 	webRtcClient?: WebRtcClient<WebRtcWebSocket>;
 }