can now watch golive streams

src/gateway/opcodes/StreamCreate.ts

@@ -8,11 +8,14 @@ import {
 	Channel,
 	Config,
 	emitEvent,
+	Member,
 	Region,
 	Snowflake,
 	Stream,
 	StreamCreateSchema,
 	StreamSession,
+	VoiceState,
+	VoiceStateUpdateEvent,
 } from "@spacebar/util";
 import { check } from "./instanceOf";
 
@@ -20,9 +23,22 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
 	check.call(this, StreamCreateSchema, data.d);
 	const body = data.d as StreamCreateSchema;
 
-	// TODO: first check if we are in a voice channel already. cannot create a stream if there's no existing voice connection
 	if (body.channel_id.trim().length === 0) return;
 
+	// first check if we are in a voice channel already. cannot create a stream if there's no existing voice connection
+	const voiceState = await VoiceState.findOne({
+		where: { user_id: this.user_id },
+	});
+
+	if (!voiceState || !voiceState.channel_id) return;
+
+	if (body.guild_id) {
+		voiceState.member = await Member.findOneOrFail({
+			where: { id: voiceState.user_id, guild_id: voiceState.guild_id },
+			relations: ["user", "roles"],
+		});
+	}
+
 	// TODO: permissions check - if it's a guild, check if user is allowed to create stream in this guild
 
 	const channel = await Channel.findOne({
@@ -41,6 +57,11 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
 		(r) => r.id === regions.default,
 	)[0];
 
+	// first make sure theres no other streams for this user that somehow didnt get cleared
+	await Stream.delete({
+		owner_id: this.user_id,
+	});
+
 	// create a new entry in db containing the token for authenticating user in stream gateway IDENTIFY
 	const stream = Stream.create({
 		id: Snowflake.generate(),
@@ -78,8 +99,7 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
 			region: guildRegion.name,
 			paused: false,
 		},
-		guild_id: body.guild_id,
-		channel_id: body.channel_id,
+		user_id: this.user_id,
 	});
 
 	await emitEvent({
@@ -92,6 +112,16 @@ export async function onStreamCreate(this: WebSocket, data: Payload) {
 		},
 		user_id: this.user_id,
 	});
+
+	voiceState.self_stream = true;
+	await voiceState.save();
+
+	await emitEvent({
+		event: "VOICE_STATE_UPDATE",
+		data: { ...voiceState },
+		guild_id: voiceState.guild_id,
+		channel_id: voiceState.channel_id,
+	} as VoiceStateUpdateEvent);
 }
 
 //stream key:

src/gateway/opcodes/StreamDelete.ts

@@ -1,5 +1,11 @@
 import { parseStreamKey, Payload, WebSocket } from "@spacebar/gateway";
-import { emitEvent, Stream, StreamDeleteSchema } from "@spacebar/util";
+import {
+	emitEvent,
+	Stream,
+	StreamDeleteSchema,
+	VoiceState,
+	VoiceStateUpdateEvent,
+} from "@spacebar/util";
 import { check } from "./instanceOf";
 
 export async function onStreamDelete(this: WebSocket, data: Payload) {
@@ -33,6 +39,22 @@ export async function onStreamDelete(this: WebSocket, data: Payload) {
 
 	await stream.remove();
 
+	const voiceState = await VoiceState.findOne({
+		where: { user_id: this.user_id },
+	});
+
+	if (voiceState) {
+		voiceState.self_stream = false;
+		await voiceState.save();
+
+		await emitEvent({
+			event: "VOICE_STATE_UPDATE",
+			data: { ...voiceState },
+			guild_id: guildId,
+			channel_id: channelId,
+		} as VoiceStateUpdateEvent);
+	}
+
 	await emitEvent({
 		event: "STREAM_DELETE",
 		data: {

src/gateway/opcodes/StreamWatch.ts

@@ -79,8 +79,8 @@ export async function onStreamWatch(this: WebSocket, data: Payload) {
 			region: guildRegion.name,
 			paused: false,
 		},
-		guild_id: guildId,
 		channel_id: channelId,
+		user_id: this.user_id,
 	});
 
 	await emitEvent({

src/webrtc/opcodes/Identify.ts

@@ -29,8 +29,10 @@ import {
 	VoicePayload,
 	WebRtcWebSocket,
 	Send,
+	generateSsrc,
 } from "@spacebar/webrtc";
 import { subscribeToProducers } from "./Video";
+import { SSRCs } from "spacebar-webrtc-types";
 
 export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 	clearTimeout(this.readyTimeout);
@@ -110,15 +112,19 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 		await subscribeToProducers.call(this);
 	});
 
+	// the server generates a unique ssrc for the audio and video stream. Must be unique among users connected to same server
+	// UDP clients will respect this ssrc, but websocket clients will generate and replace it with their own
+	const generatedSsrc: SSRCs = {
+		audio_ssrc: generateSsrc(),
+		video_ssrc: generateSsrc(),
+		rtx_ssrc: generateSsrc(),
+	};
+	this.webRtcClient.initIncomingSSRCs(generatedSsrc);
+
 	await Send(this, {
 		op: VoiceOPCodes.READY,
 		d: {
-			streams: streams?.map((x) => ({
-				...x,
-				ssrc: 2,
-				rtx_ssrc: 3,
-			})),
-			ssrc: 1,
+			ssrc: generatedSsrc.audio_ssrc,
 			port: mediaServer.port,
 			modes: [
 				"aead_aes256_gcm_rtpsize",
@@ -131,6 +137,12 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
 			],
 			ip: mediaServer.ip,
 			experiments: [],
+			streams: streams?.map((x) => ({
+				...x,
+				ssrc: generatedSsrc.video_ssrc,
+				rtx_ssrc: generatedSsrc.rtx_ssrc,
+				type: "video", // client expects this to be overriden for some reason???
+			})),
 		},
 	});
 }

src/webrtc/opcodes/Video.ts

@@ -92,7 +92,7 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 	}
 	// check if client has signaled that it will send video
 	if (wantsToProduceVideo) {
-		this.webRtcClient!.videoStream = stream;
+		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(
@@ -129,13 +129,16 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
 				op: VoiceOPCodes.VIDEO,
 				d: {
 					user_id: this.user_id,
-					audio_ssrc: ssrcs.audio_ssrc ?? 0,
+					audio_ssrc:
+						ssrcs.audio_ssrc ??
+						this.webRtcClient!.getIncomingStreamSSRCs().audio_ssrc, // can never send audio ssrc as 0, it will mess up client state for some reason
 					video_ssrc: ssrcs.video_ssrc ?? 0,
 					rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
 					streams: d.streams?.map((x) => ({
 						...x,
 						ssrc: ssrcs.video_ssrc ?? 0,
 						rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
+						type: "video",
 					})),
 				} as VoiceVideoSchema,
 			});
@@ -157,6 +160,8 @@ export async function subscribeToProducers(
 		Array.from(clients).map((client) => {
 			let needsUpdate = false;
 
+			if (client.user_id === this.user_id) return Promise.resolve(); // cannot subscribe to self
+
 			if (!client.isProducingAudio() && !client.isProducingVideo)
 				return Promise.resolve();
 
@@ -186,17 +191,19 @@ export async function subscribeToProducers(
 				op: VoiceOPCodes.VIDEO,
 				d: {
 					user_id: client.user_id,
-					audio_ssrc: ssrcs.audio_ssrc ?? 0,
+					audio_ssrc:
+						ssrcs.audio_ssrc ??
+						client.getIncomingStreamSSRCs().audio_ssrc, // can never send audio ssrc as 0, it will mess up client state for some reason
 					video_ssrc: ssrcs.video_ssrc ?? 0,
 					rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
 					streams: [
 						client.videoStream ?? {
-							type: this.type === "stream" ? "screen" : "video",
+							type: "video",
 							rid: "100",
 							ssrc: ssrcs.video_ssrc ?? 0,
 							active: client.isProducingVideo(),
 							quality: 100,
-							rtx_ssrc: ssrcs.rtx_ssrc,
+							rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
 							max_bitrate: 2500000,
 							max_framerate: 20,
 							max_resolution: {

src/webrtc/util/MediaServer.ts

@@ -65,3 +65,13 @@ export const loadWebRtcLibrary = async () => {
 		return Promise.reject();
 	}
 };
+
+const MAX_INT32BIT = 2 ** 32;
+
+let count = 1;
+export const generateSsrc = () => {
+	count++;
+	if (count >= MAX_INT32BIT) count = 1;
+
+	return count;
+};