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)
Second root cause (2026-08-12): even with yt-dlp http_headers forwarded,
YouTube still returns 403 when a signed DASH URL from --dump-single-json is
fetched raw by ffmpeg/curl on some videos (verified on fONoh7Pc6VU: curl
with the EXACT headers got 403; yt-dlp's own downloader succeeded). The
signature is tied to the extracting client context (po_token/visitor), not
just UA/IP.
Fix: getDirectScreenInput now spawns 'yt-dlp -o -' and returns its stdout
as a Readable — the same mechanism resolveMediaUrl already uses for music.
yt-dlp handles auth, cookies and transient retries internally. Merge
fragments go to /tmp/gmw-ytdlp-tmp (Nix store CWD is read-only → EACCES).
Removed resolveScreenInput + mergeScreenStreams (dead code).
Controller resolveInputWithRetry unchanged: tees the stream, waits for the
first byte (12s), retries with a fresh yt-dlp run up to 3x on error/EOF/
timeout, and destroys stuck inputs (EPIPE) so no process leaks.
Tests: rewritten for streaming (yt-dlp emits bytes; fail mode = exit 8
without stdout → stream must terminate with zero bytes).
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).
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).
prepareStream (from @dank074/discord-video-stream) unconditionally appends
audio filters 'volume@internal_lib' + 'azmq' that exist ONLY in its custom
node-av jellyfin-ffmpeg build. The Nix deployment runs plain ffmpeg-headless
on PATH, so fluent-ffmpeg died instantly with 'Filter not found' (exit 8),
the NUT output stream stayed empty, and playStream's node-av demux failed
with 'Failed to open input from Readable stream: Invalid data found when
processing input' — every screen share failed ~100ms after start.
Fix: pass customFfmpegFlags ['-filter:a','anull'] — ffmpeg applies the LAST
-filter:a for a stream, so the trailing no-op filter overrides the custom
chain (verified: command ends with '-filter:a anull', transcode runs, node-av
demux finds video+audio). Realtime volume control was already removed from
GMW (a690e5b), so dropping the filters is lossless.
Verified end-to-end with the failing URL (youtu.be/fONoh7Pc6VU, AV1+Opus
DASH): getDirectScreenInput → NUT merge → patched prepareStream → node-av
demux finds H264 video + Opus audio streams.
- Loop: toggle via POST /api/media/loop → COMMAND_MEDIA_LOOP; gateway
replays finished music track on natural end (queue untouched); status
payload exposes loop flag; FE tombol Loop di music-player + mini-player.
- Kualitas suara: music playback sekarang di-transcode sekali via ffmpeg ke
OggOpus 48kHz stereo 192kbps dengan volume di-bake ke encode — menghindari
double lossy encode (inlineVolume) yang bikin suara buram. Screen share
tetap pakai jalur lama.
- Backend: MediaState.loop, setLoop service, route + schema validation.
Volume play music terlalu besar buat user — default sekarang 0.3 (30%)
di semua layer: player gateway (musicVolume=0.3), backend state/schema
(default 0.3), dan UI slider (fallback 0.3). User tetap bisa naikin
manual via slider volume di dashboard.
Music playback produced no audio: with `-o -` yt-dlp streams media on
stdout and emits its `--print` title/duration headers on stderr, but
resolveMediaUrl read them from stdout — stripping two binary 'lines' off
the WebM container and corrupting the stream (player 'playing' but silent).
Now headers are read from stderr and the stdout media stream is returned
untouched.
Screenshare was failing with EACCES: getDirectScreenInput used --no-simulate,
making yt-dlp write .f*.part files into the read-only Nix store CWD. Dropped
it — simulate mode still returns requested_formats[].url in the JSON.
Adds tests/mediaResolve.test.ts (stderr-header + untouched-stream regression).
getDirectVideoUrl used yt-dlp --get-url with bestvideo+bestaudio, which
prints the video-only and audio-only URLs on SEPARATE lines. Only the
first (video-only) line was used, so ffmpeg had no audio track and the
GoLive stream had no sound.
Replace with getDirectScreenInput which:
- uses --dump-single-json to fetch both fresh URLs in ONE yt-dlp run
(signature URLs expire quickly)
- returns the merged progressive URL directly when one exists
- otherwise merges the video-only + audio-only DASH URLs locally via a
child ffmpeg into a single NUT stream consumed as a Readable
- tracks the merge ffmpeg process in cleanup() so shutdown kills it too
Verified end-to-end with real YouTube URLs: yt-dlp pair → live ffmpeg
merge (NUT) → H264+opus transcode yields both streams. Added
tests/screenShareInput.test.ts covering URL / DASH-pair / error paths.
GoLive (dank074 Streamer) needs the single voice session; after the stream
ends, an automatic @discordjs/voice reconnect often races Discord's session
teardown and times out (AbortError). Restore is now best-effort with a 5s
delay; if it fails the FE shows disconnected and the user clicks Connect —
an accepted tradeoff for one-voice-session-per-user.
Immediate reconnect after Streamer.stop() races Discord's voice session
teardown → AbortError. Wait 4s so the old session is fully released before
re-joining with @discordjs/voice.
The restore callback previously read getVoiceStatus() AFTER disconnectGuild
had already cleared it, so it never knew which guild/channel to reconnect.
Now release/restore receive the status captured BEFORE the audio connection
is released, so reconnect actually happens after the GoLive stream ends.
The dank074 Streamer creates its own WebRTC voice connection, but Discord
allows only ONE voice session per user. When VoiceController (audio) was
already connected, the Streamer join hung forever (never got
VOICE_SERVER_UPDATE). Now:
1. ScreenShareController takes releaseVoice/restoreVoice callbacks.
2. Before joining, it disconnects the @discordjs audio connection via
VoiceController.disconnectGuild.
3. Streamer joins + streams GoLive.
4. After the stream ends, restoreVoice reconnects the audio connection so
mic/listen keep working.
5. media.handler wires these via a new setVoiceController accessor from
commandHandler; VoiceController is the single source of truth.
Also adds caller-bound timeouts & safe .catch() everywhere so a stream
failure can never become an unhandledRejection again.
Root cause: ScreenShareController created @dank074 Streamer but never called
streamer.joinVoiceChannel() — playStream threw 'Bot is not connected to a
voice channel', and since the code only used .finally() (no .catch), the
rejection became an unhandledRejection that took down the whole gateway
(graceful shutdown triggered, systemd restarted).
Fixes:
1. Resolve active channel + streamer.joinVoiceChannel(channel) before
prepareStream/playStream (dank074 needs its OWN WebRTC voice connection).
2. .catch() on the playStream done promise — log + kill ffmpeg instead of
crashing the process.
3. .catch() on playback.done in media.handler too.
4. stop() now kills ffmpeg AND stops the streamer's voice connection.
Build & Deploy / build-and-push (backend) (push) Failing after 35s
Build & Deploy / build-and-push (discord-gateway) (push) Failing after 25s
Build & Deploy / build-and-push (proxy) (push) Failing after 25s
- Remove pnpm workspace, moon repo, and all monorepo tooling
- Delete packages/shared/, embed shared code directly into each service
- Copy packages/shared/src/* -> services/backend/src/shared/ and services/discord-gateway/src/shared/
- Replace all @bete/shared imports with @/shared/ path alias
- Remove @bete/shared workspace dependency from both services
- Update root package.json scripts from --filter to --prefix
- Rewrite Dockerfiles to build each service standalone
- Clean up biome.json, .gitignore, remove root drizzle.config.ts
Build & Deploy / build-and-push (discord-gateway) (push) Failing after 2m22s
Build & Deploy / build-and-push (backend) (push) Failing after 3m22s
Build & Deploy / build-and-push (proxy) (push) Successful in 1m36s
Build & Deploy / deploy (push) Skipped
- Consolidate all DB schema definitions into packages/shared as single source of truth
- Migrate backend from raw SQL to Drizzle ORM across all modules
- Extract frontend inline UI into separate component files
- Refactor discord-gateway circuitBreaker into conversationState + moderationState
- Convert messageStore to Proxy singleton pattern
- Add validateBody/validateQuery middleware + Zod schemas for API endpoints
- Modernize Docker builds with multi-stage + pnpm deploy
- Migrate CI/CD from deployment to image-based pipeline
- Remove 60+ unused/dead files (~15K lines)
- Update color scheme from sky-blue to teal-cyan
- Move DB connection management to @bete/shared/database
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
- Implemented AppHeader component with theme toggle and connection status.
- Created AppSidebar component for navigation with connection status indicator.
- Added MobileNav component for mobile navigation with responsive design.
- Introduced shared components: DetailStat, EmptyState, ErrorState, LoadingSkeleton, and StatCard for consistent UI.
- Developed hooks for async data fetching: useAsync, useConfig, useDashboard, useGuilds, useMedia, useMessages, useRecordings, and useVoice.
- Added chatbot API functions for sending messages and managing chat history.
- Added new dependencies for Next.js and lucide-react in pnpm-workspace.yaml.
- Refactored DashboardPage component to improve readability and error handling.
- Enhanced Header component to display error status with an alert icon.
- Updated MobileTabBar and Sidebar components to use a centralized tabs definition.
- Improved ChannelsView in dashboard-panel to handle channel fetching more cleanly.
- Fixed ActiveSpeaker type to use camelCase for userId.
- Updated MessagesPanel to handle guildId checks more gracefully.
- Adjusted API calls in dashboard and messages to align with backend expectations.
- Refined type definitions across various interfaces for consistency and clarity.
- Fix noImplicitAnyLet: add type to let match variable
- Fix noAssignInExpressions: use matchAll() + for-of instead of while
- Suppress useExhaustiveDependencies in mascot scroll effect
- Suppress useSemanticElements for message card click handler
Redis root cause: VoiceHandler passed redisPub to transmitter.start(),
which called .subscribe() on it — permanently converting the publish
connection to subscriber mode. Every subsequent command reply and status
update failed.
Fixes:
1. Transmitter now creates its own Redis client via new IORedis()
2. Mutual exclusion gate serialises start/stop to prevent null-deref races
3. PassThrough drain listeners cleaned up to stop MaxListenersExceeded
4. FFmpeg SIGTERM flagged as expected exit (no more false level-50 errors)
5. Voice recording repo captures PG error code/detail for diagnostics
Co-Authored-By: Claude <noreply@anthropic.com>
This commit introduces several significant improvements across the backend and gateway services:
- **Data Retention**: Added an automated cleanup scheduler in `discord-gateway` to prune expired messages, attachments, and voice recordings based on configurable retention policies.
- **Observability**: Integrated `prom-client` in the `backend` service to expose Prometheus metrics via `/api/metrics` and added default Node.js runtime metrics.
- **Media Handling**: Enhanced `MediaHandler` in `discord-gateway` to support media URL resolution and improved playback status tracking.
- **API & Config**: Expanded the configuration endpoint to expose more system settings and reorganized `.env.example` for better readability.
- **Refactoring & Cleanup**:
- Removed unused `better-sqlite3` dependency.
- Refactored voice channel routing.
- Improved error handling and testing coverage with comprehensive unit tests for shared utilities and error classes.
- **Documentation**: Added `MEMORY.md` for project context.
Refactor the voice recording and playback systems to improve efficiency, reduce latency, and enhance Docker build performance.
- **Infrastructure**: Optimize Dockerfiles using build mounts for pnpm cache and reorder layers for better caching of dependencies and build tools.
- **Backend/Gateway**:
- Refactor `broadcast` module to use a generic event-based system instead of hardcoded functions.
- Simplify voice recording logic by merging metadata and segment management into a unified `segment.ts`.
- Optimize audio downsampling in `streamSetup.ts` using `Int16Array` views for better performance.
- Implement `withFallback` utility for more robust Redis/Database command execution.
- **Frontend**:
- Optimize audio playback visualization using pre-computed level shapes and efficient RMS calculation.
- Reduce latency in voice commands by prioritizing WebSocket communication over HTTP.
- Improve base64 encoding efficiency in audio transmission.
- **General**:
- Add default value for `ADMIN_PASSWORD` in shared config.
- Fix Docker healthcheck to use `127.0.0.1` instead of `localhost`.
asepharyana