Root cause: prepareStream muxed video+audio into a NUT container on pipe:1.
The Demuxer scans pipe:1 for AnnexB start codes (00 00 01) to split NAL
units into access units and classify keyframes (nal_type 5). NUT container
framing bytes sat in the stream and were scanned as NALs — NAL type 0
(NUT header) instead of 5 (IDR) → every frame classified key=false →
Discord decoder never got a decodable frame → static/black GoLive tile.
Fix: output raw H264 AnnexB on pipe:1 (demuxer target) and Ogg Opus on
fd3/pipe:3 for audio. NUT is only needed for *input* parsing (single
pipe carries both streams); output is demuxed into separate raw streams.
The demuxer resolved as soon as the VIDEO init line arrived on ffmpeg stderr.
With live NUT input the audio init line ('Stream #0:1: Audio: opus') lands in a
LATER stderr chunk (NUT info-stream packets are read incrementally from the
pipe), so `return { audio: aInfo }` captured undefined → playStream skipped
AudioStream → zero audio RTP on the audio SSRC → Discord showed a static
GoLive tile even though the NUT carried opus audio.
Fix:
- wait for BOTH video and audio init lines (when audio is expected) before
resolving demux metadata, with a 3s timeout fallback
- default aInfo to opus/48kHz when withAudio instead of undefined, so the
audio stream is always exposed even if the metadata line races the return
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)
Root cause (2026-08-12 11:50 test): merge ffmpeg hit a transient YouTube
403 and exited code 8 BEFORE prepareStream attached its input listeners
(voice release+join takes ~10s). The input's end/error events fired into
the void, the encoder stdin never received EOF, demux resolved with
fallback 0x0 metadata, setSpeaking fired anyway → stream 'started' with
zero frames for 8+ minutes (black tile, both ffmpeg processes hung).
Fixes:
- mediaSource: pass yt-dlp http_headers (UA/referer) to the merge ffmpeg
via -headers to suppress transient 403s; destroy the returned stream
with an error when the merge exits non-zero before producing bytes.
- screenShareController: resolveInputWithRetry — tee the merge stream and
wait for the first readable byte (12s timeout) before proceeding; on
error/EOF/timeout retry the whole resolution with a FRESH yt-dlp run
(signed DASH URLs expire fast) up to 3 attempts. Stuck merges get
EPIPE via input.destroy() so no process leaks per attempt.
- prepareStream: race guard — if the input already ended/destroyed before
listeners attach, EOF the encoder stdin immediately; first-frame
watchdog in playStream rejects 'started but nothing flowing' after 10s
instead of resolving with a silent black stream.
Tests: +2 (merge-fail zero-byte terminal state, -headers forwarding).
SDP offer advertises profile-level-id=42e01f (constrained baseline) but
x264 encoded the default High profile — Discord's receiver configures its
decoder from the negotiated profile, so the High-profile bitstream failed
to decode → black GoLive tile despite valid access units + correct RTP
timestamps (fixed in 42a503c).
- Add -profile:v baseline to H264 encoder options (matches @dank074's
proven config; SPS now 6742c01e → profile_idc=66 baseline, aligns with
the 42e01f fmtp).
- Default x264 tune film → zerolatency (no lookahead — correct for live
GoLive; @dank074 uses it).
- Update goLive-port test to assert baseline + zerolatency.
Demuxer emitted each AnnexB NAL as its own WebRTC frame (SPS/PPS/SEI
separate from slices) with a near-zero timestamp delta (duration=1 in a
1/90000 timebase → RTP +1/frame instead of +3000 @30fps). Discord's H264
receiver never receives a complete decodable access unit → black GoLive
tile despite frames flowing.
- Group NALs into access units: buffer param-set/SEI NALs, flush one
frame per slice with preceding parameter sets (AnnexB start codes kept
so the H264RtpPacketizer finds NAL boundaries).
- Timestamp each frame at the video frame rate: duration=1, timeBase
1/fps → BaseMediaStream frametime=1000/fps ms → RTP +clockRate/fps
(3000 @ 30fps/90kHz) and correct pacing.
- Thread explicit frameRate from playStream options (raw H264 has no
timing info; ffmpeg guesses 25fps on stderr).
- Strengthen golive-demux-live-e2e: validates every frame has a slice,
no bare param-set frames, keyframes carry SPS/PPS, timeBase 1/30.
Test 00:32 confirmed the video pipeline WORKS (1410 frames @ 1280x720 sent,
ready=true, camera off) but the gateway crashed at stream stop:
unhandledRejection → graceful shutdown → systemd restart (bot offline).
Root cause candidates (both were fire-and-forget promises without .catch):
- BaseMediaConnection.setProtocols().then(...) — rejects when the PC is
closed while setProtocols is in flight (stream teardown)
- void webRtcConn.createOffer().then(...) — rejects when the PC closes
while the offer is still gathering
Fixes:
- .catch on both promise chains (log + continue; teardown is expected)
- unhandledRejection handler now treats transient stream errors (EPIPE,
ERR_STREAM_DESTROYED, ERR_STREAM_WRITE_AFTER_END, ECONNRESET) like
uncaughtException already does — warn + continue instead of shutting
down the whole gateway. Non-transient rejections still log + shutdown
(with String(reason) so the detail actually shows).
ROOT CAUSE of empty GoLive tile (finally): prepareStream ran with
includeAudio: true + output -f h264. The h264 muxer cannot mux audio
('h264 muxer does not support any stream of type audio') → header write
fails -22 → stdout empty → Demuxer ffmpeg 'Invalid data found when
processing input' → 0 frames → black tile. Reproduced locally end-to-end
(13s backpressure delay + prepareStream + demux).
Fixes:
- screenShareController: includeAudio: false (video-only GoLive; demux
path never delivers audio anyway)
- Demuxer: pin input format -f h264 for stream inputs (raw AnnexB H264
has no magic header → auto-detect unreliable on delayed pipes)
- Streamer.signalStream: self_video: false — stop flipping on the bot's
camera in Discord (user request; screen share ≠ camera)
Verified: local repro now emits 644 frames 1280x720 (was 0); tsc/biome/
vitest all green.
Root cause of 'tile appears but content empty': demux() spooled the live
NUT/H264 input to a temp file and awaited stream 'finish' — but the merge
ffmpeg output never ends during playback, so demux deadlocked, no probe,
no transcode, 0 frames sent.
- Demuxer: pipe input straight into ffmpeg stdin (-i pipe:0), parse NAL
frames live from stdout; parse video metadata from ffmpeg stderr with a
1.5s race (fall back to H264 defaults). No spool, no await-end.
- screenShareController: pass width/height/frameRate (1280x720@30) to
playStream — matches the prepareStream encode settings, so setVideoAttributes
gets real dimensions even when ffmpeg can't report metadata on an open pipe.
- Add tests/golive-demux-live-e2e.ts: proves frames flow while input is
still open (regression test for the deadlock).
- signalStream: flip voice state to self_video:true/self_deaf:false before
STREAM_CREATE (Discord silently ignores the request while video disabled)
- createStream: attach dispatch listeners before first signal (race), clean
up listeners on timeout, retry STREAM_CREATE every 3s up to 4 attempts
(upstream issue #217/#219 — Discord randomly drops the request)
- sendOpcode: direct [goLive:Streamer] log bypassing bootstrap debug filter
(proves op 18 is actually broadcast)