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feat: Custom gst webrtc signaller, runtime GPU driver package install and more (#140)

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Yes lots of commits because rebasing and all.. thankfully I know Git
just enough to have backups 😅

---------

Co-authored-by: Wanjohi <elviswanjohi47@gmail.com>
Co-authored-by: Kristian Ollikainen <DatCaptainHorse@users.noreply.github.com>
Co-authored-by: Wanjohi <71614375+wanjohiryan@users.noreply.github.com>
Co-authored-by: AquaWolf <3daquawolf@gmail.com>
This commit is contained in:
Kristian Ollikainen
2024-12-08 15:37:36 +02:00
committed by GitHub
parent 20d5ff511e
commit b6196b1c69
27 changed files with 3402 additions and 1349 deletions
Download Patch File Download Diff File Expand all files Collapse all files

106
packages/input/src/webrtc-stream.ts
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@@ -25,51 +25,16 @@ export class WebRTCStream {
}
this._onConnected = connectedCallback;
this._setup(serverURL, roomName);
}
private _setup(serverURL: string, roomName: string) {
console.log("Setting up WebSocket");
// Replace http/https with ws/wss
const wsURL = serverURL.replace(/^http/, "ws");
this._ws = new WebSocket(`${wsURL}/api/ws/${roomName}`);
this._ws.onopen = async () => {
console.log("WebSocket opened");
console.log("Setting up PeerConnection");
this._pc = new RTCPeerConnection({
iceServers: [
{
urls: "stun:stun.l.google.com:19302"
}
],
});
this._pc.ontrack = (e) => {
console.log("Track received: ", e.track);
this._mediaStream = e.streams[e.streams.length - 1];
};
this._pc.onconnectionstatechange = () => {
console.log("Connection state: ", this._pc!.connectionState);
if (this._pc!.connectionState === "connected") {
if (this._onConnected && this._mediaStream)
this._onConnected(this._mediaStream);
}
};
this._pc.onicecandidate = (e) => {
if (e.candidate) {
const message: MessageICE = {
payload_type: "ice",
candidate: e.candidate
};
this._ws!.send(encodeMessage(message));
}
}
this._pc.ondatachannel = (e) => {
this._dataChannel = e.channel;
this._setupDataChannelEvents();
}
// Send join message
const joinMessage: MessageJoin = {
payload_type: "join",
@@ -87,6 +52,11 @@ export class WebRTCStream {
const message = await decodeMessage<MessageBase>(e.data);
switch (message.payload_type) {
case "sdp":
if (!this._pc) {
// Setup peer connection now
this._setupPeerConnection();
}
console.log("Received SDP: ", (message as MessageSDP).sdp);
await this._pc!.setRemoteDescription((message as MessageSDP).sdp);
// Create our answer
const answer = await this._pc!.createAnswer();
@@ -99,12 +69,13 @@ export class WebRTCStream {
}));
break;
case "ice":
if (!this._pc) break;
// If remote description is not set yet, hold the ICE candidates
if (this._pc!.remoteDescription) {
await this._pc!.addIceCandidate((message as MessageICE).candidate);
if (this._pc.remoteDescription) {
await this._pc.addIceCandidate((message as MessageICE).candidate);
// Add held ICE candidates
for (const ice of iceHolder) {
await this._pc!.addIceCandidate(ice);
await this._pc.addIceCandidate(ice);
}
iceHolder = [];
} else {
@@ -134,7 +105,19 @@ export class WebRTCStream {
}
this._ws.onclose = () => {
console.log("WebSocket closed");
console.log("WebSocket closed, reconnecting in 3 seconds");
if (this._onConnected)
this._onConnected(null);
// Clear PeerConnection
if (this._pc) {
this._pc.close();
this._pc = undefined;
}
setTimeout(() => {
this._setup(serverURL, roomName);
}, 3000);
}
this._ws.onerror = (e) => {
@@ -142,6 +125,45 @@ export class WebRTCStream {
}
}
private _setupPeerConnection() {
console.log("Setting up PeerConnection");
this._pc = new RTCPeerConnection({
iceServers: [
{
urls: "stun:stun.l.google.com:19302"
}
],
});
this._pc.ontrack = (e) => {
console.log("Track received: ", e.track);
this._mediaStream = e.streams[e.streams.length - 1];
};
this._pc.onconnectionstatechange = () => {
console.log("Connection state: ", this._pc!.connectionState);
if (this._pc!.connectionState === "connected") {
if (this._onConnected && this._mediaStream)
this._onConnected(this._mediaStream);
}
};
this._pc.onicecandidate = (e) => {
if (e.candidate) {
const message: MessageICE = {
payload_type: "ice",
candidate: e.candidate
};
this._ws!.send(encodeMessage(message));
}
}
this._pc.ondatachannel = (e) => {
this._dataChannel = e.channel;
this._setupDataChannelEvents();
}
}
// Forces opus to stereo in Chromium browsers, because of course
private forceOpusStereo(SDP: string): string {
// Look for "minptime=10;useinbandfec=1" and replace with "minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;"