fixes
this makes chromium happy, but firefox isn't, can't test webkit, not sure on that
This commit is contained in:
MathMan05
parent
472031f9bc
commit
2ffc0ca94c
1 changed files with 144 additions and 93 deletions
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@ -8,7 +8,10 @@ class VoiceFactory {
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voices = new Map<string, Map<string, Voice>>();
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voiceChannels = new Map<string, Voice>();
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currentVoice?: Voice;
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guildUrlMap = new Map<string, {url?: string; geturl: Promise<void>; gotUrl: () => void}>();
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guildUrlMap = new Map<
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string,
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{url?: string; token?: string; geturl: Promise<void>; gotUrl: () => void}
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>();
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makeVoice(guildid: string, channelId: string, settings: Voice["settings"]) {
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let guild = this.voices.get(guildid);
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if (!guild) {
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@ -26,10 +29,11 @@ class VoiceFactory {
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onJoin = (_voice: Voice) => {};
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onLeave = (_voice: Voice) => {};
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joinVoice(channelId: string, guildId: string) {
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if (this.currentVoice) {
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const voice = this.voiceChannels.get(channelId);
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if (this.currentVoice && this.currentVoice.ws) {
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this.currentVoice.leave();
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}
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const voice = this.voiceChannels.get(channelId);
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if (!voice) throw new Error(`Voice ${channelId} does not exist`);
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voice.join();
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this.currentVoice = voice;
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@ -38,7 +42,7 @@ class VoiceFactory {
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d: {
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guild_id: guildId,
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channel_id: channelId,
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self_mute: true, //todo
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self_mute: false, //todo
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self_deaf: false, //todo
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self_video: false, //What is this? I have some guesses
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flags: 2, //?????
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@ -71,6 +75,7 @@ class VoiceFactory {
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const obj = this.guildUrlMap.get(update.d.guild_id);
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if (!obj) return;
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obj.url = update.d.endpoint;
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obj.token = update.d.token;
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obj.gotUrl();
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}
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}
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@ -88,7 +93,7 @@ class Voice {
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}
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readonly userid: string;
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settings: {bitrate: number};
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urlobj: {url?: string; geturl: Promise<void>; gotUrl: () => void};
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urlobj: {url?: string; token?: string; geturl: Promise<void>; gotUrl: () => void};
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constructor(userid: string, settings: Voice["settings"], urlobj: Voice["urlobj"]) {
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this.userid = userid;
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this.settings = settings;
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@ -163,7 +168,13 @@ class Voice {
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}
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}
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}
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offer?: string;
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hoffer?: string;
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get offer() {
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return this.hoffer;
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}
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set offer(e: string | undefined) {
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this.hoffer = e;
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}
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cleanServerSDP(sdp: string): string {
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const pc = this.pc;
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if (!pc) throw new Error("pc isn't defined");
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@ -188,7 +199,7 @@ class Voice {
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.value as string;
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const candidate = (parsed1.atr.get("candidate") as Set<string>).values().next().value as string;
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let build = `v=0\r
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o=- 1420070400000 0 IN IP4 127.0.0.1\r
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o=- 1420070400000 0 IN IP4 ${this.urlobj.url}\r
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s=-\r
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t=0 0\r
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a=msid-semantic: WMS *\r
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@ -210,7 +221,7 @@ a=fmtp:111 minptime=10;useinbandfec=1;usedtx=1\r
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a=rtcp:${rtcport}\r
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a=rtcp-fb:111 transport-cc\r
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a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r
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a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/r/n
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a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
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a=setup:passive\r
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a=mid:${bundles[i]}\r
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a=maxptime:60\r
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@ -222,23 +233,24 @@ a=candidate:${candidate}\r
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a=rtcp-mux\r`;
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} else {
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build += `
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m=video ${rtcport} UDP/TLS/RTP/SAVPF 102 103\r
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m=video ${rtcport} UDP/TLS/RTP/SAVPF 103 104\r
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${cline}\r
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a=rtpmap:102 H264/90000\r
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a=rtpmap:103 rtx/90000\r
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a=fmtp:102 x-google-max-bitrate=2500;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r
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a=fmtp:103 apt=102\r
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a=rtpmap:103 H264/90000\r
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a=rtpmap:104 rtx/90000\r
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a=fmtp:103 x-google-max-bitrate=2500;level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r
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a=fmtp:104 apt=103\r
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a=rtcp:${rtcport}\r
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a=rtcp-fb:102 ccm fir\r
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a=rtcp-fb:102 nack\r
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a=rtcp-fb:102 nack pli\r
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a=rtcp-fb:102 goog-remb\r
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a=rtcp-fb:102 transport-cc\r
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a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time/r/n
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a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01/r/n
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a=rtcp-fb:103 ccm fir\r
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a=rtcp-fb:103 nack\r
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a=rtcp-fb:103 nack pli\r
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a=rtcp-fb:103 goog-remb\r
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a=rtcp-fb:103 transport-cc\r
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a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
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a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
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a=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r
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a=extmap:13 urn:3gpp:video-orientation\r
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a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay/r/na=setup:passive/r/n
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a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
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a=setup:passive
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a=mid:${bundles[i]}\r
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a=${mode}\r
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a=ice-ufrag:${ICE_UFRAG}\r
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@ -254,25 +266,13 @@ a=rtcp-mux\r`;
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}
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counter?: string;
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negotationneeded() {
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if (this.pc && this.offer) {
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if (this.pc) {
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const pc = this.pc;
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pc.addEventListener("negotiationneeded", async () => {
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this.offer = (
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await pc.createOffer({
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offerToReceiveAudio: true,
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offerToReceiveVideo: true,
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})
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).sdp;
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await pc.setLocalDescription({sdp: this.offer});
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const sendOffer = async () => {
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console.trace("neg need");
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await pc.setLocalDescription();
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console.warn(pc.localDescription?.sdp, this.offer);
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if (!this.counter) throw new Error("counter isn't defined");
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const counter = this.counter;
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const remote: {sdp: string; type: RTCSdpType} = {
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sdp: this.cleanServerSDP(counter),
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type: "answer",
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};
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console.log(remote);
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await pc.setRemoteDescription(remote);
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const senders = this.senders.difference(this.ssrcMap);
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for (const sender of senders) {
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for (const thing of (await sender.getStats()) as Map<string, any>) {
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@ -282,8 +282,44 @@ a=rtcp-mux\r`;
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}
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}
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}
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};
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pc.addEventListener("negotiationneeded", async () => {
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await sendOffer();
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console.log(this.ssrcMap);
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});
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pc.addEventListener("signalingstatechange", async () => {
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while (!this.counter) await new Promise((res) => setTimeout(res, 100));
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if (this.pc && this.counter) {
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if (pc.signalingState === "have-local-offer") {
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const counter = this.counter;
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const remote: {sdp: string; type: RTCSdpType} = {
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sdp: this.cleanServerSDP(counter),
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type: "answer",
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};
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console.log(remote);
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await pc.setRemoteDescription(remote);
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}
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}
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});
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pc.addEventListener("connectionstatechange", async () => {
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if (pc.connectionState === "connecting") {
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await pc.setLocalDescription();
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}
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});
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pc.addEventListener("icegatheringstatechange", async () => {
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console.log("icegatheringstatechange", pc.iceGatheringState, this.pc, this.counter);
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if (this.pc && this.counter) {
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if (pc.iceGatheringState === "complete") {
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console.log("icegatheringstatechange");
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const counter = this.counter;
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const remote: {sdp: string; type: RTCSdpType} = {
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sdp: this.cleanServerSDP(counter),
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type: "answer",
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};
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await pc.setRemoteDescription(remote);
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}
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}
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});
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}
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}
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async makeOp12(
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@ -294,6 +330,7 @@ a=rtcp-mux\r`;
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sender = sender[0];
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}
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if (this.ws) {
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console.log(this.ssrcMap);
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this.ws.send(
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JSON.stringify({
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op: 12,
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@ -374,58 +411,26 @@ a=rtcp-mux\r`;
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}
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async continueWebRTC(data: sdpback) {
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if (this.pc && this.offer) {
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const pc = this.pc;
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this.negotationneeded();
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this.status = "Starting Audio streams";
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const audioStream = await navigator.mediaDevices.getUserMedia({video: false, audio: true});
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for (const track of audioStream.getAudioTracks()) {
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//Add track
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this.setupMic(audioStream);
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const sender = pc.addTrack(track);
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this.senders.add(sender);
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console.log(sender);
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}
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for (let i = 0; i < 10; i++) {
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pc.addTransceiver("audio", {
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direction: "recvonly",
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streams: [],
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sendEncodings: [{active: true, maxBitrate: this.settings.bitrate}],
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});
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}
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for (let i = 0; i < 10; i++) {
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pc.addTransceiver("video", {
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direction: "recvonly",
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streams: [],
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sendEncodings: [{active: true, maxBitrate: this.settings.bitrate}],
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});
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}
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this.counter = data.d.sdp;
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pc.ontrack = async (e) => {
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this.status = "Done";
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if (e.track.kind === "video") {
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return;
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}
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const media = e.streams[0];
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console.log("got audio:", e);
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for (const track of media.getTracks()) {
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console.log(track);
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}
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const context = new AudioContext();
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await context.resume();
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const ss = context.createMediaStreamSource(media);
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console.log(media);
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ss.connect(context.destination);
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new Audio().srcObject = media; //weird I know, but it's for chromium/webkit bug
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this.recivers.add(e.receiver);
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};
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} else {
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this.status = "Connection failed";
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}
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}
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reciverMap = new Map<number, RTCRtpReceiver>();
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off?: Promise<RTCSessionDescriptionInit>;
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async makeOffer() {
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if (this.pc?.localDescription?.sdp) return {sdp: this.pc?.localDescription?.sdp};
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if (this.off) return this.off;
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return (this.off = new Promise<RTCSessionDescriptionInit>(async (res) => {
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if (!this.pc) throw new Error("stupid");
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console.error("stupid!");
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const offer = await this.pc.createOffer({
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offerToReceiveAudio: true,
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offerToReceiveVideo: true,
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});
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res(offer);
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}));
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}
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async figureRecivers() {
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await new Promise((res) => setTimeout(res, 500));
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for (const reciver of this.recivers) {
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@ -441,15 +446,61 @@ a=rtcp-mux\r`;
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async startWebRTC() {
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this.status = "Making offer";
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const pc = new RTCPeerConnection();
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this.pc = pc;
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const offer = await pc.createOffer({
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offerToReceiveAudio: true,
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offerToReceiveVideo: true,
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});
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this.status = "Starting RTC connection";
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const sdp = offer.sdp;
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this.offer = sdp;
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pc.ontrack = async (e) => {
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this.status = "Done";
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if (e.track.kind === "video") {
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return;
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}
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const media = e.streams[0];
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console.log("got audio:", e);
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for (const track of media.getTracks()) {
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console.log(track);
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}
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const context = new AudioContext();
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await context.resume();
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const ss = context.createMediaStreamSource(media);
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console.log(media);
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ss.connect(context.destination);
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new Audio().srcObject = media; //weird I know, but it's for chromium/webkit bug
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this.recivers.add(e.receiver);
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};
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const audioStream = await navigator.mediaDevices.getUserMedia({video: false, audio: true});
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for (const track of audioStream.getAudioTracks()) {
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//Add track
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this.setupMic(audioStream);
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const sender = pc.addTrack(track);
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this.senders.add(sender);
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console.log(sender);
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}
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for (let i = 0; i < 10; i++) {
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pc.addTransceiver("audio", {
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direction: "recvonly",
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streams: [],
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sendEncodings: [{active: true, maxBitrate: this.settings.bitrate}],
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});
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}
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for (let i = 0; i < 10; i++) {
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pc.addTransceiver("video", {
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direction: "recvonly",
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streams: [],
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sendEncodings: [{active: true, maxBitrate: this.settings.bitrate}],
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});
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}
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this.pc = pc;
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this.negotationneeded();
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await new Promise((res) => setTimeout(res, 100));
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let sdp = this.offer;
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if (!sdp) {
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const offer = await this.makeOffer();
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this.status = "Starting RTC connection";
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sdp = offer.sdp;
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this.offer = sdp;
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}
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await pc.setLocalDescription();
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if (!sdp) {
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this.status = "No SDP";
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this.ws?.close();
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@ -658,7 +709,7 @@ a=rtcp-mux\r`;
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server_id: update.d.guild_id,
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user_id: update.d.user_id,
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session_id: update.d.session_id,
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token: update.d.token,
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token: this.urlobj.token,
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video: false,
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streams: [
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{
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