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updates

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This commit is contained in:
dank074 2025-04-17 23:31:42 -05:00
parent 54f6685082
commit c845d2070c
4 changed files with 87 additions and 19 deletions
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2
package-lock.json generated
View File

@ -9377,7 +9377,7 @@
},
"node_modules/spacebar-webrtc-types": {
"version": "1.0.1",
"resolved": "git+ssh://git@github.com/dank074/spacebar-webrtc-types.git#20e9d4d838e053101487813e666b759f3c486fce",
"resolved": "git+ssh://git@github.com/dank074/spacebar-webrtc-types.git#04044902e71f2c80fc39c256ec5d27fdb8ca425a",
"dev": true,
"license": "ISC"
},

6
src/webrtc/opcodes/Identify.ts
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@ -30,6 +30,7 @@ import {
WebRtcWebSocket,
Send,
} from "@spacebar/webrtc";
import { subscribeToProducers } from "./Video";
export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
clearTimeout(this.readyTimeout);
@ -91,6 +92,11 @@ export async function onIdentify(this: WebRtcWebSocket, data: VoicePayload) {
mediaServer.onClientClose(this.webRtcClient!);
});
// once connected subscribe to tracks from other users
this.webRtcClient.emitter.once("connected", async () => {
await subscribeToProducers.call(this);
});
await Send(this, {
op: VoiceOPCodes.READY,
d: {

6
src/webrtc/opcodes/SelectProtocol.ts
View File

@ -35,7 +35,7 @@ export async function onSelectProtocol(
payload.d,
) as SelectProtocolSchema;
const answer = await mediaServer.onOffer(
const response = await mediaServer.onOffer(
this.webRtcClient,
data.sdp!,
data.codecs ?? [],
@ -44,8 +44,8 @@ export async function onSelectProtocol(
await Send(this, {
op: VoiceOPCodes.SESSION_DESCRIPTION,
d: {
video_codec: "H264",
sdp: answer,
video_codec: response.selectedVideoCodec,
sdp: response.sdp,
media_session_id: this.session_id,
audio_codec: "opus",
},

92
src/webrtc/opcodes/Video.ts
View File

@ -45,18 +45,16 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
}
}
const stream = d.streams?.find((element) => element !== undefined);
const stream = d.streams?.find((element) => element.active);
await Send(this, { op: VoiceOPCodes.MEDIA_SINK_WANTS, d: { any: 100 } });
const ssrcs = this.webRtcClient.getIncomingStreamSSRCs();
const clientsThatNeedUpdate = new Set<WebRtcClient<WebRtcWebSocket>>();
// check if client has signaled that it will send audio
if (d.audio_ssrc !== 0) {
// check if we already have incoming media for this ssrcs, if not, publish a new audio track for it
if (ssrcs.audio_ssrc != d.audio_ssrc) {
// 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}`,
);
@ -65,14 +63,13 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
});
}
// now check that all clients have outgoing media for this ssrcs
// now check that all clients have subscribed to our audio
for (const client of mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
rtc_server_id,
)) {
if (client.user_id === this.user_id) continue;
const ssrcs = client.getOutgoingStreamSSRCsForUser(this.user_id);
if (ssrcs.audio_ssrc != d.audio_ssrc) {
if (!client.isSubscribedToTrack(this.user_id, "audio")) {
console.log(
`[${client.user_id}] subscribing to audio track ssrcs: ${d.audio_ssrc}`,
);
@ -84,8 +81,9 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
}
// check if client has signaled that it will send video
if (d.video_ssrc !== 0 && stream?.active) {
// check if we already have incoming media for this ssrcs, if not, publish a new video track for it
if (ssrcs.video_ssrc != d.video_ssrc) {
this.webRtcClient!.videoStream = stream;
// check if we are already publishing video, if not, publish a new video track for it
if (!this.webRtcClient!.isProducingVideo()) {
console.log(
`[${this.user_id}] publishing new video track ssrc:${d.video_ssrc}`,
);
@ -95,16 +93,13 @@ export async function onVideo(this: WebRtcWebSocket, payload: VoicePayload) {
});
}
// now check that all clients have outgoing media for this ssrcs
// now check that all clients have subscribed to our video track
for (const client of mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
rtc_server_id,
)) {
if (client.user_id === this.user_id) continue;
const ssrcs = client.getOutgoingStreamSSRCsForUser(
this.webRtcClient.user_id,
);
if (ssrcs.video_ssrc != d.video_ssrc) {
if (!client.isSubscribedToTrack(this.user_id, "video")) {
console.log(
`[${client.user_id}] subscribing to video track ssrc: ${d.video_ssrc}`,
);
@ -136,3 +131,70 @@ 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<void> {
const clients = mediaServer.getClientsForRtcServer<WebRtcWebSocket>(
this.webRtcClient!.rtc_server_id,
);
await Promise.all(
Array.from(clients).map((client) => {
let needsUpdate = false;
if (!client.isProducingAudio() && !client.isProducingVideo)
return Promise.resolve();
if (
client.isProducingAudio() &&
!this.webRtcClient!.isSubscribedToTrack(client.user_id, "audio")
) {
this.webRtcClient!.subscribeToTrack(client.user_id, "audio");
needsUpdate = true;
}
if (
client.isProducingVideo() &&
!this.webRtcClient!.isSubscribedToTrack(client.user_id, "video")
) {
this.webRtcClient!.subscribeToTrack(client.user_id, "video");
needsUpdate = true;
}
if (!needsUpdate) return Promise.resolve();
const ssrcs = this.webRtcClient!.getOutgoingStreamSSRCsForUser(
client.user_id,
);
return Send(this, {
op: VoiceOPCodes.VIDEO,
d: {
user_id: client.user_id,
audio_ssrc: ssrcs.audio_ssrc ?? 0,
video_ssrc: ssrcs.video_ssrc ?? 0,
rtx_ssrc: ssrcs.rtx_ssrc ?? 0,
streams: [
client.videoStream ?? {
type: this.type === "stream" ? "screen" : "video",
rid: "100",
ssrc: ssrcs.video_ssrc ?? 0,
active: client.isProducingVideo(),
quality: 100,
rtx_ssrc: ssrcs.rtx_ssrc,
max_bitrate: 2500000,
max_framerate: 20,
max_resolution: {
type: "fixed",
width: 1280,
height: 720,
},
},
],
} as VoiceVideoSchema,
});
}),
);
}