Commit Graph
14 Commits
Author SHA1 Message Date
asepharyana 392bc35a0d fix(gateway): output raw H264+Opus (not NUT) so Demuxer parses NAL keyframes correctly
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.
2026-08-12 15:39:47 +07:00
asepharyana 196cb1d3af fix(gateway): await audio stream line before demux resolve — audio RTP was dropped by metadata race
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
2026-08-12 14:45:06 +07:00
asepharyana f1aa08cdf6 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)
2026-08-12 14:20:29 +07:00
asepharyana 67ab289caa fix(gateway): fail-fast + retry on screen share merge failure (black tile zombie)
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).
2026-08-12 12:17:43 +07:00
asepharyana ef9e243609 fix(goLive): encode H264 baseline to match SDP profile-level-id (black tile)
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.
2026-08-12 11:37:04 +07:00
asepharyana 42a503c206 fix(goLive): demux access-unit grouping + correct RTP timestamps (black tile)
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.
2026-08-12 11:13:00 +07:00
asepharyana 652974e23a fix(goLive): gateway crash on screenshare stop — unhandledRejection during teardown
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).
2026-08-12 00:36:53 +07:00
asepharyana 407e003399 fix(goLive): black screen root cause — h264 muxer can't carry audio; disable self_video camera
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.
2026-08-12 00:25:01 +07:00
asepharyana 968a43b0f4 debug(goLive): instrument frame pipeline — demux spawn/stderr/frames, playStream resolve, sendVideoFrame drop/send
Tile kosong meski STREAM_CREATE handshake penuh (22:18-22:19 retest):
- Demuxer logs spawn args, ffmpeg stderr errors, frame count every 30
- playStream logs createStream resolved + demux done + setPacketizer
- sendVideoFrame logs DROPPED (ready/track) + sent frame count
2026-08-11 22:31:31 +07:00
asepharyana 91c7a67d2f fix(goLive): stream demux directly instead of spool-to-file (empty screen share)
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).
2026-08-11 21:43:16 +07:00
asepharyana 8615383829 fix(goLive): STREAM_CREATE handshake — self_video voice state + retry + send instrumentation
- 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)
2026-08-11 20:45:00 +07:00
asepharyana ff554fcff2 fix(goLive): instrument voice/stream handshake + createStream timeout (12s) 2026-08-11 20:00:33 +07:00
asepharyana 9139e225f4 perf(golive): ffmpeg-spawn demuxer (no node-av) + E2E pipeline tests
Phase 2 — replace the 114MB node-av binary with a plain ffmpeg spawn:

Demuxer.ts: spool stream input to temp file → probe via ffmpeg stderr
(ffmpeg-headless ships NO ffprobe — parse 'Stream #0:0: Video: h264...
640x360, 30 fps' from -loglevel info) → ffmpeg -c copy -f h264 pipe:1
→ NAL-split frames. Falls back to h264 defaults when probe fails.

prepareStream.ts: resolve ffmpeg from FFMPEG_PATH env → Nix store
ffmpeg-headless (hash-prefixed entry!) → PATH; split encoder option
strings ('-forced-idr 1' → two argv) — fluent-ffmpeg used to split
automatically, spawn does not.

E2E tests (tsx, need LD_LIBRARY_PATH=/tmp/ldc-build):
- golive-demux-e2e.ts: real H264 file → 33 NAL frames + dims from probe
- golive-pipeline-e2e.ts: prepareStream → demux → 82 frames
- golive-videostream-e2e.ts: local peer pair → demux → VideoStream →
  native setPacketizer/sendFrame/addTimestamp → 33 frames sent connected

Pitfalls captured: setPacketizer before negotiation breaks createOffer
('No DataChannel or Track to negotiate'); track methods are read-only
(no monkeypatching); both peers must declare audio+video tracks or
answer hangs; state() returns 'closed' after close() — snapshot first.
2026-08-11 17:38:06 +07:00
asepharyana 9ae230d047 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.
2026-08-11 16:16:52 +07:00