fix(gateway): deliver audio + per-IDR SPS/PPS in GoLive screen share
Screen share showed a single frozen frame: the GoLive pipeline sent video only (-"-an", h264 muxer cannot carry audio) so the audio SSRC never transmitted and Discord kept the stream in thumbnail state. - prepareStream: mux NUT when includeAudio (h264 muxer drops audio) and return the actual container format - Demuxer: support NUT input with a second output pipe (fd3) carrying Ogg Opus; parse OGG pages into opus frames (20ms, 48kHz) emitted as GoLiveFrames; fix metadata parsing that dropped the audio stream line when it arrived in a later stderr chunk (early parsedMeta return) - playStream: pipe audio.stream into AudioStream → RTP on the audio SSRC - Encoders: -x264-params repeat-headers=1 → SPS/PPS inline before EVERY IDR (NUT remux drops container extradata; also enables PLI recovery) - screenShareController: includeAudio true - tests: demuxerNut.test.ts — OGG parser unit test + real ffmpeg NUT integration (video access units + parsed opus frames)
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@@ -168,6 +168,9 @@ export class BaseMediaConnection extends EventEmitter {
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}): void {
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// we hardcoded STREAMS_SIMULCAST, which will always be array of 1
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const stream = d.streams[0];
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console.log(
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`[goLive:${this.constructor.name}] READY ssrc=${d.ssrc} ip=${d.ip} port=${d.port} streams=${JSON.stringify(d.streams)}`,
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);
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this._webRtcParams = {
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address: d.ip,
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port: d.port,
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@@ -183,6 +186,10 @@ export class BaseMediaConnection extends EventEmitter {
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dave_protocol_version?: number;
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}): Promise<void> {
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if (!("sdp" in d)) throw new Error("Only WebRTC connections are allowed");
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// DEBUG: dump Discord's real answer SDP — which payload types did it select?
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console.log(
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`[goLive:${this.constructor.name}] DISCORD_ANSWER_SDP ${JSON.stringify(d.sdp ?? "").slice(0, 900)}`,
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);
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this._daveProtocolVersion = d.dave_protocol_version ?? 0;
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this.initDave();
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// Discord's SDP is garbage — generate our own from its pieces
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@@ -258,12 +265,10 @@ a=ice-lite
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`a=rtcp-fb:${el.payload_type} transport-cc`,
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])
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.join("\n");
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this._webRtcWrapper.webRtcConn?.setRemoteDescription(
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[audioSection, videoSection, videoRtpMap].join("\n"),
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"answer",
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);
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const builtAnswer = [audioSection, videoSection, videoRtpMap].join("\n");
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this._webRtcWrapper.webRtcConn?.setRemoteDescription(builtAnswer, "answer");
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console.log(
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`[goLive:${this.constructor.name}] SELECT_PROTOCOL_ACK processed — remote answer set (${[audioSection, videoSection].join("\n").length}B)`,
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`[goLive:${this.constructor.name}] SELECT_PROTOCOL_ACK processed — remote answer set (${builtAnswer.length}B) video_mline=${videoPayloadTypes.join(" ")}`,
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);
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this.emit("select_protocol_ack");
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}
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