Commit Graph
9 Commits
Author SHA1 Message Date
asepharyana 60faaa9304 fix(goLive): backpressure-throttle screen-share pipeline to 1x (video freezes while audio plays)
Root cause: prepareStream's ffmpeg consumed a YouTube VOD at download/CPU
speed (~10x real-time), so the demuxer buffered a huge frame backlog.
The sender paces at 30fps but always emitted the OLDEST buffered frames, so
the viewer saw frozen/laggy video while audio (tiny, jitter-buffer
recoverable) stayed smooth. That is exactly the 'video stuck, voice normal'
symptom reported live.

Fix: propagate vPipe backpressure UP to the demuxer's ffmpeg stdout — when
the sender can't keep up, pause the source, which stalls the demuxer and
back-pressures the encoder, pinning the whole pipeline to 1x. Also add a
realtime (-re) option for file/URL inputs (no-op for the streaming path,
which is what screen share uses).

Verified: 10s test clip encodes in 1.8s without -re vs 9.5s with it; tsc --noEmit clean.
2026-08-13 11:59:33 +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 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 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 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