perf(golive/spike): binding addTrack + TS port of @dank074 media stack
Phase 1 spike: replace @dank074/discord-video-stream + node-datachannel +
node-av (1.3GB) with minimal libdatachannel N-API binding + native RTP
packetizers (H264 FU-A, RTCP SR/NACK, pacer) + pure-TS GoLive stack.
Binding v0.4: addTrack (m=audio/video SDP), TrackWrap w/ setPacketizer +
sendFrame (raw RTP to transport) + addTimestamp — verified by two-peer
handshake emitting SDP with audio(opus 120)+video(H264 101) and 8-frame
RTP roundtrip.
TS layer (src/goLive/, 21 files): CodecPayloadType, VoiceOpCodes,
GatewayOpCodes, utils, BaseMediaConnection (voice WS + DAVE + heartbeat),
VoiceConnection, StreamConnection, Streamer, WebRtcWrapper (SDP mungling,
DAVE encrypt, packetizer chain), BaseMediaStream (pacing/sync), VideoStream,
AudioStream, Demuxer (ffmpeg-spawn NUT/AnnexB, no node-av 114M binary),
Encoders, prepareStream/playStream.
Integration: screenShareController.ts now imports from ../../goLive/index.js —
prepareStream(prepared, ...) + playStream(prepared, streamer, {...}).
Tests: tests/goLive-port.test.ts (8/8 pass). tsc --noEmit clean. biome clean.
This commit is contained in:
asepharyana
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/**
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* WebRTC connection wrapper for GoLive — ported from
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* @dank074/discord-video-stream WebRtcWrapper.js, with the media stack
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* (packetizers, RTCP SR/NACK, pacing) provided by the libdatachannel-min
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* binding instead of node-datachannel's JS-exposed media classes.
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*/
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import {
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H264Helpers,
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H264NalUnitTypes,
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splitNalu,
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startCode3,
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} from "./AnnexBHelper.js";
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import { CodecPayloadType } from "./CodecPayloadType.js";
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import type { NativePeerConnection, NativeTrack } from "./native.js";
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import { loadNative } from "./native.js";
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import { rewriteSPSVUI } from "./SPSVUIRewriter.js";
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import { normalizeVideoCodec } from "./utils.js";
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export type WebRtcVideoCodec = "H264" | "H265" | "VP8" | "VP9" | "AV1";
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export interface WebRtcParams {
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address: string;
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port: number;
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audioSsrc: number;
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videoSsrc: number;
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rtxSsrc: number;
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supportedEncryptionModes: string[];
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}
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/** Minimal surface of the media connection that WebRtcWrapper drives. */
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export interface VideoAttribute {
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fps: number;
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width: number;
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height: number;
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}
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export interface MediaConnectionLike {
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daveReady: boolean;
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daveSession: {
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encryptOpus(frame: Buffer): Buffer;
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encrypt(mediaType: number, codec: number, frame: Buffer): Buffer;
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} | null;
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webRtcParams: WebRtcParams | null;
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setSpeaking(speaking: boolean): void;
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setVideoAttributes(enabled: boolean, attr?: VideoAttribute): void;
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}
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/** Media types used by DAVE encryption (from @dank074). */
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export enum DaveMediaType {
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AUDIO = 0,
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VIDEO = 1,
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}
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/** DAVE codec ids (from @dank074). */
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export enum DaveCodec {
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UNKNOWN = 0,
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VP8 = 2,
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VP9 = 3,
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H264 = 4,
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H265 = 5,
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AV1 = 6,
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}
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export class WebRtcConnWrapper {
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private _mediaConn: MediaConnectionLike;
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private _webRtcConn: NativePeerConnection | null = null;
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private _audioTrack: NativeTrack | null = null;
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private _videoTrack: NativeTrack | null = null;
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private _videoCodec: WebRtcVideoCodec | null = null;
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/** Assigned by BaseMediaConnection to send the gathered SDP to Discord. */
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onLocalDescription: ((sdp: string) => void) | null = null;
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constructor(mediaConn: MediaConnectionLike) {
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this._mediaConn = mediaConn;
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}
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initWebRtc(): NativePeerConnection {
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const native = loadNative();
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this._webRtcConn = new native.PeerConnection({
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iceServers: ["stun:stun.l.google.com:19302"],
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});
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// Track mids must match @dank074: "0" audio, "1" video.
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this._audioTrack = this._webRtcConn.addTrack("0", "audio");
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this._videoTrack = this._webRtcConn.addTrack("1", "video");
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return this._webRtcConn;
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}
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close(): void {
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this._webRtcConn?.close();
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this._webRtcConn = null;
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}
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get webRtcConn(): NativePeerConnection | null {
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return this._webRtcConn;
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}
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get ready(): boolean {
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return this._webRtcConn?.state() === "connected";
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}
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get mediaConnection(): MediaConnectionLike {
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return this._mediaConn;
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}
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sendAudioFrame(frame: Buffer, frametime: number): void {
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if (!this.ready || !this._audioTrack) return;
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const clockRate = CodecPayloadType.opus.clockRate;
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if (this.mediaConnection.daveReady && this.mediaConnection.daveSession) {
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frame = this.mediaConnection.daveSession.encryptOpus(frame);
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}
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this._audioTrack.sendFrame(frame);
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this._audioTrack.addTimestamp(Math.round((frametime * clockRate) / 1000));
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}
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sendVideoFrame(frame: Buffer, frametime: number): void {
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if (!this.ready || !this._videoTrack) return;
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const clockRate = CodecPayloadType[this._videoCodec ?? "H264"].clockRate;
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if (this._videoCodec === "H264") {
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let spsRewritten = false;
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const nalus = splitNalu(frame).map((el) => {
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if (H264Helpers.getUnitType(el) === H264NalUnitTypes.SPS) {
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spsRewritten = true;
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return rewriteSPSVUI(el);
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}
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return el;
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});
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if (spsRewritten)
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frame = Buffer.concat(nalus.flatMap((el) => [startCode3, el]));
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}
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if (this.mediaConnection.daveReady && this.mediaConnection.daveSession) {
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let daveCodec = DaveCodec.UNKNOWN;
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switch (this._videoCodec) {
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case "H264":
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daveCodec = DaveCodec.H264;
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break;
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case "H265":
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daveCodec = DaveCodec.H265;
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break;
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case "VP8":
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daveCodec = DaveCodec.VP8;
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break;
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case "VP9":
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daveCodec = DaveCodec.VP9;
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break;
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case "AV1":
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daveCodec = DaveCodec.AV1;
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break;
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default:
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break;
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}
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frame = this.mediaConnection.daveSession.encrypt(
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DaveMediaType.VIDEO,
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daveCodec,
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frame,
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);
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}
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this._videoTrack.sendFrame(frame);
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this._videoTrack.addTimestamp(Math.round((frametime * clockRate) / 1000));
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}
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setPacketizer(videoCodec: string): void {
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if (!this.mediaConnection.webRtcParams) {
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throw new Error("WebRTC connection not ready");
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}
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const { audioSsrc, videoSsrc } = this.mediaConnection.webRtcParams;
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this._videoCodec = normalizeVideoCodec(videoCodec);
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// Audio packetizer: opus 120 @ 48kHz, playout delay ext id 5 (like @dank074)
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this._audioTrack?.setPacketizer(
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"audio",
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audioSsrc,
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CodecPayloadType.opus.payload_type,
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CodecPayloadType.opus.clockRate,
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5,
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0,
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1,
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);
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// Video packetizer: H264/H265/AV1 with their payload types
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const codecEntry = CodecPayloadType[this._videoCodec];
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if (!codecEntry) {
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throw new Error(`Packetizer not implemented for ${this._videoCodec}`);
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}
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const nativeKind =
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this._videoCodec === "H264"
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? "h264"
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: this._videoCodec === "H265"
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? "h265"
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: this._videoCodec === "AV1"
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? "av1"
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: (() => {
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throw new Error(
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`Packetizer not implemented for ${this._videoCodec}`,
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);
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})();
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this._videoTrack?.setPacketizer(
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nativeKind,
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videoSsrc,
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codecEntry.payload_type,
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codecEntry.clockRate,
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5,
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0,
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10,
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);
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}
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}
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