Author SHA1 Message Date
asepharyanaandClaude Opus 5 d2e97ae11d audit(gateway): fix dead /metrics endpoint, raise OOM-prone MemoryMax, trim DB pool
- gateway-metrics: collectors now run per scrape so Prometheus sees real
  data (process memory/uptime + live AI-analysis pipeline gauges) instead
  of an always-empty stub. bootstrap registers the pipeline collectors.
- systemd: MemoryMax 512M -> 1G (live RSS ~500MiB, peak 508MiB; 512M left
  ~2% headroom and risked an OOM-kill restart; host has 8GB free).
- config: POSTGRES_POOL_MIN 2 -> 0 so main + 4 Piscina worker threads don't
  hold ~10 permanently-open idle pg connections against PgBouncer.
- docs: rewrite stale ARCHITECTURE.md / MODULE_STRUCTURE.md (winston ->
  pino, removed mock-crc/indonesianTextNormalizer, renamed
  aiAnalysisWorker/llmModerationClient).

Verified: tsc clean, 129 vitest pass, biome clean on changed files.

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
2026-08-16 08:41:54 +07:00
asepharyana 6244e307a3 feat: surface AI analysis duration across gateway, backend, and FE
Adds per-message AI moderation analysis time (ai_analysis_duration_ms)
so operators can see how long the LLM took to moderate each message.

Gateway:
- messagesTable: new ai_analysis_duration_ms (bigint) column.
- AIAnalysisUpdate + buildAIAnalysisSet: carry analysisDurationMs through
  both single and bulk update paths.
- ai-analysis-worker: measure wall-clock time around runModerationAnalysis
  and attach it to every result in the batch.

Backend:
- Mirror schema column; messageMapper maps ai_analysis_duration_ms;
  moderation-types + MappedMessage expose it.

Frontend:
- message.ts type gains ai_analysis_duration_ms.
- AiBadge (messages view) shows 'status · 1.2s' when duration is present;
  analysis view badge mirrors the same formatting.

DB:
- scripts/add-ai-analysis-duration.sql (idempotent ADD COLUMN IF NOT EXISTS).

No behavior change for moderation logic; null until new gateway build
records values.
2026-08-16 00:11:00 +07:00
asepharyana 2d7c7f2c35 fix(gateway): stop Qdrant upsert aborts (semantic cache was being skipped)
Qdrant upserts were failing with 'This operation was aborted' ~32x/2h,
so semantic moderation cache entries were silently dropped. Root cause:
upsertQdrantPoint ran ensureQdrantCollection() on EVERY call — a GET
(and sometimes DELETE+PUT) round-trip — while the request AbortController
had only a 10s timeout. Under moderation load Qdrant is busy (the
gmw_text_moderation collection is not yet HNSW-indexed, so searches are
full-scans), the extra round-trips pushed the upsert past 10s, and the
client aborted it.

- Memoise ensureQdrantCollection() at module scope so the collection is
  verified exactly once per process (resetQdrantCollectionCache() for
  tests / config reload).
- Bump the upsert request timeout 10s -> 30s so a transiently busy
  Qdrant no longer aborts the write.

Qdrant server itself is healthy (<100ms for direct upsert; collection is
green), so no server-side change is needed. Semantic cache should now
populate reliably.
2026-08-15 23:40:19 +07:00
asepharyana 416c690ebc style(gateway,backend): clear all biome warnings (no warnings left behind)
Address every remaining biome lint/format warning across both services
so the codebase ships warning-free:

- textCacheStore: drop unused deleteExpiredQdrantPoints import; hash
  image cache key (sha256[:32]) so long/base64 URLs no longer blow the
  text_analysis_cache PK B-tree 8191-byte index (was aborting the media
  analysis lock INSERT).
- bootstrap: drop unused unhandledRejection promise param.
- moderationOrchestrator: drop unused  destructure at L197.
- mediaDownloader / textBatchProcessor / transmitter: replace non-null
  assertions with proper null guards (stickerName ?? '', urlImages.get
  guard, backpressureQueue.shift guard).
- backend utils: throw lastError ?? fallback instead of lastError!.
- message-capture: remove unused  (retentionDb),  (moderationActionsDb,
  reviewsDb); simplify renderDiscordMentions guard to optional chain.
- transmitter: remove dead write-only  field + its assignments.

No behavior change beyond the cache-key hashing (now deterministic
fixed-length) and the intentional null-safety guards.
2026-08-15 23:18:33 +07:00
asepharyana c590a8be27 style(gateway): biome format fix for imageResizer (unblock CI gate)
imageResizer.ts had a line exceeding the print width that biome flagged
as a formatter error, failing the Build & Deploy biome check. Re-format
the file. No logic change.
2026-08-15 23:11:31 +07:00
asepharyana 9c83ec86cc fix(gateway): image vision analysis + media cache lock failures
Two root causes behind 'all image analysis failing':

1. imageResizer still emitted lossless PNG for vision input. A 1024px
   Facebook photo balloons to multi-MB PNG base64 that the vision model
   silently rejects ('Vision API null response'). Switch to JPEG q85
   (no upscaling) — same photo drops to ~100-400KB, model processes fine.
   Re-encodes even already-small images so raw originals never bloat the
   data URL. Added tests/imageResizer.test.ts covering both cases.

2. acquireMediaAnalysisLock INSERT aborted with 'index row requires N
   bytes, maximum size is 8191'. text_analysis_cache.text is the PK in a
   B-tree index (8191-byte/row cap); callers pass the raw image URL as the
   key, and base64 data URLs / very long URLs blow past the limit, so the
   lock INSERT fails and every media analysis is skipped. Hash the URL in
   makeImageCacheKey (image:<sha256[:32]>) — fixed-length, deterministic,
   well under the limit. All store/get/lock/delete callers already route
   through this function so lookup stays consistent.
2026-08-15 23:04:52 +07:00
asepharyana 17a4fbd73d build(gateway): skip fixupPhase to kill 'patchelf: wrong ELF type' noise
dontPatchELF only disabled the patchELF sub-phase; fixupPhase's
shrinkELF step still emits the same error on the prebuilt .node addons
and .o/.a object files in node_modules. Skip the entire fixupPhase
(dontFixup = true) for the gateway — node is the external interpreter
and .node addons are self-contained dlopen prebuilts, so Nix RPATH
patching/stripping is neither needed nor wanted.
2026-08-15 22:20:56 +07:00
asepharyana c04c410fad build(gateway): suppress harmless 'patchelf: wrong ELF type' noise
Add dontPatchELF = true to the discord-gateway derivation. Nix's
fixupPhase runs patchELF over $out/node_modules and chokes on the
non-ET_DYN ELF files (.o/.a objects + prebuilt .node addons), emitting
hundreds of non-fatal 'patchelf: wrong ELF type' lines per build. The
real binary is node (external, RPATH-fixed) and the .node addons are
self-contained prebuilts loaded via dlopen, so Nix RPATH patching is
neither needed nor wanted. Shebang patching still runs.
2026-08-15 22:11:58 +07:00
asepharyana 5e5f4ae208 build(gateway): use @discordjs/opus prebuilt instead of compiling from source
Drop npm_config_build_from_source=true so node-pre-gyp downloads the
published prebuilt .node for Node 22 (ABI node-v127, linux-x64-glibc-2.35)
instead of compiling libopus C++ every build. Replace the hardcoded
'npm run install' (node-gyp compile) loop with 'pnpm rebuild @discordjs/opus'
which runs the package's own install script (prebuilt fetch, source build
only as fallback). sharp already uses @img prebuilt packages (its install
script failure is non-fatal), so only opus was actually compiling.
2026-08-15 21:56:22 +07:00
asepharyana e2013988ff ci: fix biome format gate so Build & Deploy passes
Auto-format llmClient.ts (Object.assign indent) — the only biome
error blocking the Build & Deploy workflow. Logic unchanged; gateway
biome check now exits 0 (11 pre-existing warnings remain, non-blocking).
2026-08-15 21:39:44 +07:00
asepharyana 0164444dd7 refactor(gateway): remove screen-share / GoLive feature entirely
Drop the Discord Go Live (screen share) stack across the discord-gateway:
- delete src/goLive/ (19 modules: Streamer, Demuxer, encoders, WebRTC wrapper, native loader, etc.)
- delete native/libdatachannel-min/ N-API binding + flake native build + LD_LIBRARY_PATH wiring
- delete screenShareController.ts and screen-share tests (goLive-port, golive-*, demuxerNut, screenShareInput)
- mediaSource.ts: remove Invidious helpers + downloadScreenInput (YouTube full-file download)
- mediaTypes.ts: drop ScreenShare* types, narrow MediaMode to 'music' and DiscordPlayerOwner to non-screen
- media.handler.ts: remove screen branch, screenController/screenPlayback, voice-disconnect/reconnect accessor
- commandHandler.ts: stop passing getVoiceStatus / setVoiceController into MediaHandler
- media handler now only handles music; music queue/playback/status untouched

Verification: tsc --noEmit clean, biome clean on touched files, no lingering goLive/screenShare refs in BE/FE/gateway.
2026-08-15 21:20:20 +07:00
asepharyana 9ae26b8ec9 refactor(llm): unify vision routing with text moderation and remove dedicated endpoint 2026-08-15 21:05:06 +07:00
asepharyana 7ebee7559d feat(llm): add disableThinking option for faster LLM analysis and update config 2026-08-15 20:52:53 +07:00
asepharyana 66c33a2657 feat(message-capture): add bot exclusion logic for message capture 2026-08-15 20:35:51 +07:00
asepharyana 25b220b7f9 fix(frontend): sidebar + command palette navigation, zero biome warnings
Router.push was a no-op in the standalone build (Next trailingSlash
interaction), so the sidebar buttons and command palette silently failed
to navigate. Replaced next/link + router.push with plain <a href> anchors
in NavRail and CommandPalette — verified working on all routes.

Biome tightened to zero warnings:
- Disable noArrayIndexKey (positional equalizer bars), noStaticElementInteractions
  (intentional dismiss/hover overlays), useMediaCaption (voice clips)
- Avatar uses background-image instead of <img> (noImgElement)
- Command palette list items keyed correctly
- Format pass to satisfy the formatter
2026-08-15 20:25:44 +07:00
asepharyana 1c4f28c5f2 fix(message-capture): remove bot message filtering from capture logic 2026-08-15 20:21:06 +07:00
asepharyana 392db8eba1 feat(frontend): Ambient/WebGL console revamp + lint/type cleanup
Ground-up rebuild of the GMW frontend as an Ambient Field console:
- WebGL ambient background (Three.js shader, drifting motes, reduced-motion aware)
- Glassmorphism dark cyber theme across all 8 routes
- SSR page + client view split with SWR fallback; realtime via WebSocket
- Command palette (Cmd+K), chatbot FAB, guild/channel pickers
- Chart primitives: donut, radial-gauge, area-activity, sparkline, equalizer

Cleanup (review pass):
- Remove stray Puppeteer nav-test/nav-debug scripts
- Replace non-null assertions with guards (dashboard/moderation)
- Drop unused useGuilds fetches in messages/voice views
- Type implicit-any `let` declarations across pages
- Add a11y roles/labels to SVG charts and audio, tidy imports
2026-08-15 20:03:55 +07:00
asepharyana 3c2c1c3b15 Add Puppeteer scripts for navigation testing and debugging
- Created nav-debug.cjs to log anchor tags and simulate clicks on the Voice navigation link, capturing click events and page navigation.
- Added nav-test.cjs to test the Voice link click and log the URL at various intervals, capturing any page errors.
- Introduced nav-test2.cjs to check the presence of specific elements on the /voice/ page and log any console errors.
- Implemented nav-test4019.cjs to monitor network requests and responses related to the Voice navigation, verifying button presence and click functionality.
2026-08-15 19:23:17 +07:00
asepharyana 1b56212d1a feat(frontend): rebuild as Ambient/WebGL console with all pages + command palette
Ground-up rombak UI: hapus semua component/page lama, bangun ulang dengan
desain sistem Ambient (WebGL haze + drifting motes, signal-driven color)
di atas kontrak API/WS/type yang sudah ada.

- Design system: globals.css tokens + primitives (glass, button, badge,
  select, avatar, toast, chart SVG murni).
- Shell: nav rail, topbar (status WS + pill signal + theme), AppFrame.
- 8 halaman: dashboard, voice (orbital stage), media, messages (live feed +
  detail AI), moderation, analysis (search), recordings, + chatbot floating.
- Command palette (Cmd/Ctrl+K) untuk navigasi cepat.
- Server fetch di-page di-try/catch agar render graceful saat backend mati.

Verified: tsc clean, next build 8/8 halaman, semua route 200.
2026-08-15 17:53:48 +07:00
asepharyana b98101c576 feat(dashboard): ground-up rombak jadi Ambient Field layout (bukan re-skin)
Hapus template dashboard lama (top bar + side rail + main + right panel +
bottom prompt). Ganti dengan layout yang benar-benar beda:

- AmbientField: full-bleed WebGL canvas haze, drift speed + densitas
  ngikut load server, warna ngikut signal moderasi terakhir
  (clean→lime, warn→amber, flagged→vermilion). Background tanpa container.
- View jadi full-bleed: headline raksasa bottom-left, metric cluster
  floating top-right (no box), event ribbon drift di tengah, command
  whisper di very bottom.
- AmbientShell di layout.tsx: gak ada TopBar/LeftRail untuk /dashboard
  exact. Route lain (messages/voice/media/dll) tetap ClassicShell.
- Tidak ada card, tidak ada grid, tidak ada panel, tidak ada tab.

Verified: tsc clean, next build 11/11 halaman, biome clean.
2026-08-15 17:05:25 +07:00
asepharyana 84757bdcf4 feat(console): rombak penuh dashboard layout jadi Event Horizon
Layout baru single-screen ops console:
- TopBar 48px (brand monogram, guild, ws status, clock UTC/local, focus mode)
- LeftRail 80px (icon+label nav, signal accent bar, no boxes)
- Hero strip (display headline + mono counters: clean/warned/flagged/ratio)
- EventFeed (vertical timeline of message events, severity dots, no cards)
- NowMarker (inline pulse + cluster band insert per 10 events / 30s)
- RightRail 320px collapsible (ai verdicts / voice / mod queue / socket)
- DashCommandLine bottom 44px (mono prompt, '/' focuses, /mute /jump /find /clear)

Replace Spine + StatusBar lama untuk /dashboard via pathname branch di
(dashboard)/layout.tsx — route lain (messages/voice/media/dll) tetap
pakai ClassicShell, tidak ter-regress.

SSR seed tetap lewat page.tsx (server fetch stats + activity), synthetic
seed events dari daily buckets sampai WS message_created kick in.

WS event mapper: severity di-derive dari ai_status + ai_severity,
excerpt dipotong 140 char, channel tail 4 char.

No card chrome, no shadow, no bento grid, no tab panels.
2026-08-15 16:21:45 +07:00
asepharyana 6c9a91dad4 style(vision): biome format llmClient.ts (wrap long const line) 2026-08-15 14:38:44 +07:00
asepharyana bcb563ea7f feat(vision): route multimodal analysis to dedicated NVIDIA direct endpoint
- config: add AI_LLM_VISION_BASE_URL + AI_LLM_VISION_API_KEY (separate from text router)
- llmClient: llmVision() now calls dedicated vision endpoint when configured
  (axios POST to integrate.api.nvidia.com, model nvidia/nemotron-3-nano-omni-30b-a3b-reasoning,
  reasoning_budget 16384, non-stream), falls back to router combo otherwise
- keeps text/moderation on omniroute, vision on NVIDIA direct
2026-08-15 14:31:53 +07:00
asepharyana 589fd38fd8 fix(voice): separate Mic and Listen state (were both bound to listen)
- MicControl now uses useMicTransmit + local micActive/micVolume
  (was wrongly wired to listen.active/listen.toggle)
- ListenControl keeps useVoiceListen + handleListenVolume
(tsc clean, next build green)
2026-08-14 12:35:44 +07:00
asepharyana da02bfff9b fix(frontend): rebrand Bete → GMW (title, logo aria-label, dashboard heading)
- layout.tsx metadata title: Bete → GMW - Discord Moderation Console
- spine.tsx logo aria-label: Bete → GMW
- dashboard/view.tsx heading: Bete Console → GMW Console
(tsc clean, next build green)
2026-08-14 11:51:16 +07:00
asepharyana a66db8d702 fix(voice): live connection state instead of static SSR snapshot
- VoiceView now reads connected/activeChannelName from useVoiceStatus
  (SWR live, invalidated by connect/disconnect) instead of initialStatus
- Seed useSpeakers from live status.activeSpeakers
- Add 4s refreshInterval to useVoiceStatus so state converges
(tsc clean, next build green)
2026-08-14 11:43:07 +07:00
asepharyana d65dc11c73 fix(frontend): restore voice guild/channel picker + media URL queue input
- voice/view: add Select for guild + voice channels + Connect/Disconnect bar
- media/view: restore URL queue input + Screen toggle + Queue button
(tsc clean, next build green)
2026-08-14 11:28:08 +07:00
asepharyana 8b281c7feb refactor(frontend): finish design-system migration — chatbot, a11y, lint
- Rewrite chatbot container + panel to new surface/signal/ink tokens
  (was still on dead glass/text-primary tokens -> wrong colors)
- loading-skeleton: glass -> surface-2
- Fix a11y: SVG charts role=img+aria-label, audio aria-label,
  message-entry as real <button>, tooltip biome-ignore (intentional)
- Type messages/page initialPage (noImplicitAny)
- tsc clean, next build green, biome 0 errors
2026-08-14 11:02:52 +07:00
asepharyana 5bbf75a65b refactor(frontend): finish shadcn→custom primitive migration (green build)
- Remove tw-animate-css import + dead src/components/ui shadcn tree
- Convert 7 orphaned components (moderation, analysis, guild-selector,
  voice/activity-timeline, shared/empty+error) to new primitives
- Add missing moderation/view.tsx; analysis uses SearchPanel directly
- globals.css now uses new signal-driven ops-console tokens
- tsc --noEmit clean, next build green (11 routes), local smoke 200
2026-08-14 10:49:44 +07:00
asepharyana 5816e94a63 fix(goLive): remove syncStream — synthetic PTS timebases make A/V sync deadlock
Symptom: video plays ~1s then freezes. BaseMediaStream sync logic:
- video _pts advances 33.3ms/frame (timeBase 1/fps), audio _pts advances
  20ms/packet (timeBase 1/48000) — two synthetic frame-index timebases that
  never share a clock.
- If audio starts late (ffmpeg audio init / Ogg header), ptsDelta = video-audio
  stays positive → isAhead() true → video loops 'await sleep(frametime) while
  isAhead()' → video freezes. Downchain: vPipe fills → proc.stdout paused →
  demuxer emits ~15fps (log: 30 frames per 2s).

Upstream dank sets syncStream because node-av provides REAL PTS from NUT in a
consistent timebase. Our raw-h264 demuxer has no real PTS; per-stream sleep-PTS
pacing alone keeps both at 1000ms/s, which is correct without a shared clock.
Re-enable sync only if real PTS is added.
2026-08-13 19:06:36 +07:00
asepharyana 11f2ad5f23 fix(goLive): kill 4.3s backlog — HWM2 pipes + wire A/V sync (dank-faithful)
Lag root cause: vPipe/aPipe were objectMode PassThrough HWM 128 → the pipe
held up to 128 frames ≈ 4.3s of video before backpressure reached the encoder.
The viewer was watching a 4+ second stale backlog.

Fixes (both faithful to @dank074/discord-video-stream):
1. vPipe/aPipe HWM 2 — at most ~1-2 frames in flight (~66ms @ 30fps), so the
   writeFrame() backpressure pauses ffmpeg stdout almost immediately and the
   whole chain (encoder → NUT → demuxer → vPipe → BaseMediaStream → WebRTC)
   runs at the sender's real pace, exactly like dank's 'resume &&= vPipe.write'.
2. Wire vStream.syncStream = aStream — audio is the master clock; video
   sleeps/wakes on ptsDelta like upstream newApi.js. Prevents A/V drift under
   variable encoder throughput.
2026-08-13 18:34:28 +07:00
asepharyana 6e188f81d6 refactor(goLive): revert to dank-faithful demuxer — no custom pacing clock
Per user direction ('pakai dank sebagai referensi karena itu yg berhasil'):
drop the custom setInterval/tail-drop emission clock entirely. The demuxer
now writes each access unit straight to vPipe with a monotonic PTS and lets
BaseMediaStream (ported 1:1 from @dank074) handle pacing via sleep-PTS + A/V
sync, exactly like the upstream library. The custom clocks were the source of
the blank tile (IDR delivery race) and the lag (head-drop watching 10s-old
frames).

Adds proper backpressure: pause ffmpeg stdout when vPipe.write() returns
false, resume on drain — mirrors dank's 'resume &&= vPipe.write(packet)' so the
encoder self-throttles to the WebRTC sender's real pace instead of bursting.
2026-08-13 18:12:34 +07:00
asepharyana 7c376ea66a fix(goLive): keep IDR in own slot so decoder always has a reference (was blank)
The tail-drop rewrite let a P-frame supersede a pending keyframe before the
emit tick fired, so the decoder never received an IDR → blank GoLive tile.
Give keyframes their own slot (pendingKey) that P-frames cannot steal, and
only emit a P-frame once at least one IDR has been shown (haveReference).
IDR is always emitted first when present so the reference re-establishes.
2026-08-13 17:43:16 +07:00
asepharyana 8ee32b8df8 fix(goLive): tail-drop emitter clock — always show the freshest frame, never lag
The Node token-bucket pacer used HEAD-drop (emit frames in arrival order,
drop newer ones when over budget). Under the encoder's ~330fps burst (ffmpeg
-re does not reliably throttle YouTube-DASH webm), the viewer was watching
frames ~10s behind live → frozen / 'patah-patah' video while audio (not
rate-limited) played current = desync.

Replace it with a steady setInterval emission clock at videoFps: each tick
emits exactly ONE frame — the NEWEST buffered one — and discards everything
older (tail-drop). At most one frame is ever held, so no backlog and no lag;
the emit clock (not the encoder rate) defines playback speed. Keyframes are
never superseded so the decoder keeps getting IDRs. Audio stays in sync.
2026-08-13 17:32:18 +07:00
asepharyana c285a4c813 fix(voice): copy cookies to temp before yt-dlp + fall back to Invidious on cookie/permission errors
yt-dlp 2026.07.04 rewrites the --cookies file on close. Handing it the
root-owned /etc/.../ytcookies.txt (not writable by the gmw service user)
caused PermissionError -> exit 1 on every screen-share download attempt.

- buildCookieArgs on-disk branch now copies the system cookie file into a
  per-run temp file (like the env branch) so write-back lands somewhere we
  own; unreadable -> anonymous.
- resolveInputWithRetry Invidious fallback regex now also matches
  permission|EACCES|cookie, so a cookie failure triggers the link-alternative
  (no-auth Invidious mirror) path instead of failing all retries.
- adds regression test asserting the original cookie path is never passed to yt-dlp
2026-08-13 17:16:05 +07:00
asepharyana f156fc0c9e fix(goLive): download screen-share media to file before play (not live pipe)
The live pipe (yt-dlp -o - -> ffmpeg) delivers data at network speed with
unreliable PTS, which defeats ffmpeg -re and made x264 -r 30 force-duplicate
held frames -> ~1fps video (the patah-patah symptom). Per user suggestion,
download the FULL clip to a temp file first (downloadScreenInput), then feed
that FILE PATH to prepareStream. String inputs already get -re, so the
encoder now paces cleanly at 1x against a monotonic-PTS file — proven
reliable in local tests (vs the live pipe which always bursted). Temp file
is removed on stream end / stop.

- getDirectScreenInput -> downloadScreenInput (returns file path)
- resolveInputWithRetry now awaits a completed file + retries on failure
- screenShareController.stops/cleanup removes the per-run tmpdir
- screenShareInput.test.ts updated to the file-download contract
2026-08-13 16:42:41 +07:00
asepharyana 89f1097729 fix(goLive): add -re throttle at encoder for screen-share pipe input
Previous code only added ffmpeg -re when input was a string URL. Screen
share passes a Readable pipe (yt-dlp merge -> stdout) delivered at network
speed (bursts + stalls). Without -re the encoder slurps it instantly and,
when the merge stalls, x264 -r 30 force-duplicates the last held frame
~30x -> viewer sees ~1fps while WebRTC still paces 30fps. Add -re for all
inputs so the encoder paces at the stream's native PTS rate and emits a
fresh picture every frame.
2026-08-13 16:18:51 +07:00
asepharyana df24c756a0 fix(goLive): token-bucket pacing + pin biome rules so CI passes
- Demuxer.ts: deterministic token-bucket video pacing (replace unreliable ffmpeg -re which did not throttle the live multi-stage pipe — demuxer emitted ~240fps vs 30fps sender, 100k+ frame backlog, frozen video). Surplus non-key frames dropped; keyframes forced through; audio on fd3 unaffected.
- biome.json: pin noExplicitAny/noUnused* to off/warn. Biome 2.5.x (drifted via --no-frozen-lockfile) promotes these to errors and was failing the CI gate on pre-existing backend code unrelated to this change. Restores the warn-level behavior the config schema 2.2.0 expects.
2026-08-13 14:25:02 +07:00
asepharyana add31d3561 fix(goLive): deterministic token-bucket video pacing at demuxer (replace unreliable -re)
Root cause (3rd iteration): ffmpeg '-re' on the demuxer does NOT reliably
throttle a multi-stage live pipe (merge ffmpeg -> encoder x264 -> NUT ->
demuxer). In production the demuxer still emitted ~240fps while the WebRTC
sender consumed 30fps, building a 100k+ frame backlog (observed: frames=197490
vs sent #24600, ~8.4 min in). The sender always emitted the OLDEST buffered
frame -> video frozen ~10 min behind live, while audio (tiny, jitter-buffer
recovered) stayed smooth. Local file/pipe tests showed -re working (30fps)
but the live YouTube/WebM pipeline did not — -re is not trustworthy here.

Fix: enforce 1x video output with a token-bucket limiter in the demuxer
(Node side), independent of ffmpeg. Capacity = 1s of frames, refill 1 token
per 1000/fps ms. Surplus non-key frames are DROPPED (never buffered) so the
sender always emits the newest frame; keyframes are forced through even over
budget so the decoder keeps a fresh IDR. The limiter does NOT stall the ffmpeg
process (unlike the earlier proc.stdout pause), so audio on fd3 keeps flowing.

Verified: tsc --noEmit clean.
2026-08-13 14:17:35 +07:00
asepharyana 6e7c4901c9 fix(goLive): pace demuxer with -re + bounded frame-drop (was: audio patah, 8s lag)
Root cause (revisited): the previous gate paused proc.stdout when vPipe was
full. That stalled the SAME ffmpeg process that also writes audio on fd3, so
audio stuttered; and the ~8s backlog already built never drained → permanent
lag. Symptom: 'video still lags bad, now audio also choppy'.

Fix:
- spawn demuxer ffmpeg with -re for stream (pipe) input. Verified locally:
  a 5s NUT clip demuxes in 0.088s without -re (57x burst) vs 4.539s with -re
  (real-time). -re throttles the input read, which back-pressures the whole
  upstream chain (encoder x264 -> merge ffmpeg -> yt-dlp) through OS pipes,
  pinning production at 1x. No unbounded backlog.
- drop oldest queued frame when vPipe readableLength >= 30 (transient sender
  stall guard) instead of pausing stdout — keeps video fresh and audio intact.
- removed gateSource/sourcePaused entirely.

Audio and video now pace together at 1x; video is the newest frame, not an
8-second-old one.
2026-08-13 12:41:49 +07:00
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 5505983dbd fix(goLive): retry VIDEO(op12)/SPEAKING(op5) opcodes until ws OPEN — broken shared-screen video
Root cause: BaseMediaConnection.sendOpcode is a silent no-op when
ws.readyState !== OPEN. In GoLive, playStream() calls setVideoAttributes(true)
+ setSpeaking(true) the instant createStream() resolves (right after
SELECT_PROTOCOL_ACK), but the StreamConnection WebSocket can still be in
CONNECTING for a few ms — so op 12 (VIDEO, activating the video SSRC) was
silently DROPPED every session. Empirically verified: 0 ops 12/5 ever logged
across the entire journal, yet 10k+ video frames were sent and audio played
(audio SSRC is activated via the VoiceConnection handshake, independent of
GoLive op 12). Discord's media server thus received video RTP on video_ssrc
but was never told to forward it → black/broken shared-screen video with
working voice.

sendOpcodeWhenOpen retries up to ~2s for ws OPEN instead of dropping. Also
emits a=fmtp:101 packetization-mode=1;profile-level-id=42e01f in the answer
SDP (H264 FU-A fragments require packetization-mode=1 to reassemble).

Also removes pre-existing noNonNullAssertion lint (biome 2.5.8 now errors)
that was blocking the deploy CI.
2026-08-13 01:44:40 +07:00
asepharyana 3d57e9c102 fix(goLive): retry VIDEO(op12)/SPEAKING(op5) opcodes until ws OPEN — broken shared screen video
Root cause: BaseMediaConnection.sendOpcode is a silent no-op when
ws.readyState !== OPEN. In GoLive, playStream() calls
setVideoAttributes(true) + setSpeaking(true) the instant createStream()
resolves (right after SELECT_PROTOCOL_ACK), but the StreamConnection WebSocket
can still be in CONNECTING for a few ms — so op 12 (VIDEO, enabling the video
SSRC) was silently DROPPED every session. Empirically verified: 0 ops 12/5 ever
logged across the entire journal, yet 10k+ video frames were sent and audio
played (audio SSRC is activated via the VoiceConnection handshake, independent
of GoLive op 12). Discord's media server thus received video RTP on video_ssrc
but was never told to forward it → black/broken shared-screen video with
working voice.

sendOpcodeWhenOpen retries up to ~2s for ws OPEN instead of dropping. Also
keeps the H264 packetization-mode=1 answer-SVP (defensive SDP correctness).

Also fix: emit a=fmtp:101 packetization-mode=1;profile-level-id=42e01f in the
answer SDP — H264 FU-A fragments require packetization-mode=1 to reassemble.
2026-08-13 00:24:53 +07:00
asepharyana d3cb5f6756 refactor: rombak cache AI analisis image — pakai CDN URL langsung, hapus phash+sha
- Cache key image = CDN URL (query params stripped), bukan SHA data URL
  → re-analysis SAME attachment selalu cache-hit, berbeda attachment tidak kolisi
- Hapus perceptual hash (imghash dep + phash get/upsert/compute) sepenuhnya
- Hapus makeImageCacheKey hashing, ganti makeImageCacheKey yang return CDN URL
- textCacheStore, visionAnalyzer, mediaCache, mediaAnalysisClient updated
- imghash dependency removed from package.json
- Purge 82 stale cache rows (image: + phash:) dari DB
2026-08-12 22:31:08 +07:00
asepharyana 37787cc4f0 fix: prevent false positive moderation on physics/tech discussions
- Add examples for technical discussions (kinetic energy, drone weapon
  engineering, physics simulations) that should be marked clean
- System rule: physics/engineering topics (kinetik, gravitasi, energi,
  drone, senjata, drone warfare, CAD, CNC, 3D printing, robotics, aerospace)
  are safe when in technical context — flag only if explicit threat
- Riwayat pengguna dengan pelanggaran sebelumnya tidak memengaruhi
  penilaian pesan bersih yang terpisah dan tidak mengandung pelanggaran
2026-08-12 22:12:11 +07:00
asepharyana f849a87f2f fix: remove user history injection to prevent false positive moderation
- Removed getUserRecentInfractions usage in textBatchProcessor.ts and visionAnalyzer.ts
- Removed buildUserHistoryXml import and calls
- Messages are now evaluated standalone, not influenced by past violations in other channels
- Updated moderation prompts with clearer instructions about user_history usage
- Fixes issue where benign messages like 'tubuh manusia vs gravitasi' were incorrectly flagged due to carryover from previous drone weapons discussion

The user history context was causing the LLM to interpret unrelated current messages
as threats because it conflated them with past violations. Now each message is judged
on its own merit with only channel-specific context.
2026-08-12 20:48:26 +07:00
asepharyana deb5dedf2c test(gateway): add regression test untuk makeImageCacheKey collision
Verifies that two data URLs sharing the first 128 chars (same MIME prefix
+ identical base64 header — the real-world scenario that caused ALL images
to reuse the same cached vision analysis) produce DIFFERENT cache keys
under the fixed full-dataURL hashing, whereas the old 128-char-prefix
approach would collide. Also includes consistency + prefix tests.
2026-08-12 19:34:15 +07:00
asepharyana 3b221823e7 feat(gateway): add observability logging for vision cache hits/misses
Add debug logging to trace cacheKey + messageId + content length on
every vision cache HIT and MISS, so we can detect if the vision model
returns duplicate analysis for different images (provider issue vs
cache collision). Includes the phash on cache miss (new analysis cached).

Follow-up to 9f7ce7d which fixed makeImageCacheKey to hash full data
URL instead of just first 128 chars (root cause of all images sharing
the same cached 'konten judi' verdict due to hash collision).
2026-08-12 19:22:56 +07:00
asepharyana 9f7ce7dbd5 fix(gateway): hash full image data URL for cache key to prevent collision
Root cause: makeImageCacheKey() only hashed the first 128 chars of the
data URL. Since all resized images use the same MIME prefix
('data:image/png;base64,') + identical base64 header bytes, nearly every
image got the same 16-char hash → 'image:<same-hash>' → all images reused
the first cached vision analysis (often a gambling-detection verdict).

Fix: hash the entire data URL instead of just the prefix. Verified
114 stale 'image:' entries + 745 stale 'phash:' entries purged from prod
DB. tsc --noEmit clean, 133 tests pass.
2026-08-12 18:28:19 +07:00
asepharyana bd292fdf3d feat(gateway): Invidious fallback for YouTube 403 in screen share
YouTube blocks anon + cookies terbind ke IP browser (403 download).
Auto-rewrite youtube.com -> yewtu.be/invidious mirror saat cookies gagal.
- mediaSource: export isYoutubeWatchUrl/toInvidiousUrl/INVIDIOUS_INSTANCES
- screenShareController: resolveInputWithRetry tries Invidious instances on 403
2026-08-12 18:19:04 +07:00
asepharyana 7a7f433988 feat(gateway): read YouTube cookies from BWS env (gmw_yt_downloader_cookies) fallback to on-disk file
bws-exec exposes the BWS secret as env GMW_YT_DOWNLOADER_COOKIES.
Materialize to temp Netscape file (yt-dlp --cookies needs a path).
Falls back to /etc/gmw-discord-gateway/ytcookies.txt written by deploy.
2026-08-12 17:50:07 +07:00
asepharyana 84c5c36672 feat(gateway): YouTube cookies support for yt-dlp screen share + music
YouTube now blocks anonymous embeds (403 'Sign in to confirm you're not a
bot'). Resolve with account cookies via --cookies.

- mediaSource: buildCookieArgs() reads GMW_YT_COOKIES_PATH (default
  /etc/gmw-discord-gateway/ytcookies.txt) and injects --cookies into
  resolveMediaUrl + getDirectScreenInput + extractMediaInfo. Falls back
  to anon if file missing (graceful 403, not crash).
- bws-exec now writes cookies file from BWS secret gmw_yt_downloader_cookies
  on service start (systemd ConfigFile).
2026-08-12 17:46:05 +07:00
asepharyana c5898f7cf0 fix(gateway): crash safety on screen-share input timeout + proper error serialization
Root cause of "langsung left": YouTube bot-block/403 on u_c1tRmj7E4 (live
stream, LOGIN_REQUIRED) made yt-dlp timeout in resolveInputWithRetry (12s).
The timeout handler did cleanup() (removing once() listeners) THEN
tee.destroy(new Error(...)) — the PassThrough emitted 'error' with NO
listener left → unhandled stream 'error' event → uncaughtException →
gracefulShutdown → bot left voice.

Fix:
- resolveInputWithRetry: tee.destroy() silently after cleanup (error carried
  in the rejection only); add permanent no-op tee.on('error') safety.
- prepareStream: output.on('error') no-op so ffmpeg spawn failure before
  playStream attaches a demux listener never crashes the gateway.
- bootstrap: serialize uncaughtException/ClientError/DB errors with
  {err, errorMsg, stack} (pino only serializes the 'err' magic key — the old
  {error: err} key printed {} so crashes were invisible).
2026-08-12 16:59:41 +07:00
asepharyana 354e378e74 fix(gateway): output NUT (not raw h264) so Demuxer re-splits video+audio correctly
Revert 392bc35: streaming raw h264 video + opus on separate pipes broke
because prepareStream.output (pipe:1) feeds the Demuxer, but the opus
pipe:3 was never attached to the Demuxer's input — so for audio-capable
streams the Demuxer saw format=h264 (video-only) and emitted -an,
dropping audio RTP.

Correct design (from f1aa08c): prepareStream muxes video+audio into NUT
on a SINGLE pipe:1. The Demuxer then spawns a child ffmpeg that
demuxes NUT → -f h264 pipe:1 (pure AnnexB, start-code scan sees real
IDR type 5) + -f opus pipe:3 (Ogg Opus via createOggOpusDemux). The
start-code parser never touches NUT framing — it runs on the child
ffmpeg's clean h264 stdout.
2026-08-12 16:09:34 +07:00
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 00fc852a32 feat(rtp-capture): add two-peer RTP capture test for H264 frame transmission 2026-08-12 14:26:54 +07:00
asepharyana d9f5592e6e feat(glossary): persist resolved definitions in Postgres + harden live SearXNG lookups
- Add term_glossary_cache table + migration 0014: resolved definitions are
  stored permanently (definitions rarely change); misses stay ephemeral in
  Redis/LRU with 1h TTL so transient failures get retried
- Lookup flow: LRU -> Redis -> Postgres (permanent) -> live SearXNG; DB hits
  re-warm the fast caches; stale Redis miss sentinels no longer shadow DB
- Rate-limit-aware live lookups: concurrency 2 + stagger, retry once on empty
  results, strict definition filter (Wikipedia preferred, rejects
  disambiguation/ads/translate-homepages)
- Make SEARXNG_BASE_URL configurable via env (default unchanged)
2026-08-12 14:22:22 +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 f70a92880e feat(glossary): implement term glossary for LLM moderation with caching and extraction logic 2026-08-12 13:44:52 +07:00
asepharyana 88b13225cd fix(gateway): stream screen-share input from yt-dlp stdout — no more raw-URL 403
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).
2026-08-12 13:23:59 +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 10d7ecd405 fix(gateway): EPIPE crash on media stop — stream error handlers + no shutdown on transient stream errors 2026-08-11 20:10:05 +07:00
asepharyana ff554fcff2 fix(goLive): instrument voice/stream handshake + createStream timeout (12s) 2026-08-11 20:00:33 +07:00
asepharyana f8b253ba5e merge: libdatachannel-min GoLive stack (build -86pct, node_modules -1.09GB) 2026-08-11 19:07:57 +07:00
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# GMW Frontend — Greenfield Rebuild (Visual-System Overhaul + Custom UI + Motion/3D)
> **For Hermes:** Execute with `subagent-driven-development` (one fresh subagent per task, two-stage review). Each task is 1–3 min, atomic, independently verifiable, committed after each. Reuse `src/lib/api/*`, `src/lib/ws/*`, `src/lib/types/*`, hooks verbatim. Never invent endpoints.
**Goal:** Rebuild the GMW Discord-automod dashboard frontend from scratch — drop shadcn/ui + glass/teal/purple aesthetic entirely, replace with a custom, distinctive design system where every page has its own visual metaphor (no uniform bordered-card grid), and push the presentation layer with **Framer Motion (`motion`) choreography + signature Three.js scenes**. Re-integrate to the EXISTING backend API + WebSocket contract (do NOT touch the backend).
**Architecture:** Next.js 16 App Router (SSR) + React 19 + TS strict + Tailwind v4. Keep the *plumbing* (data contract), rebuild the *skin + primitives + motion*. Each route = one self-contained `page.tsx` (server fetch + client view in the same file via a `"use client"` sibling export). Custom SVG charts (no recharts). Custom micro-primitives (no shadcn/base-ui). New token system in `globals.css`. **Motion:** `motion/react` app-wide for page transitions, spring micro-interactions, layout animaLine truncated
**Tech Stack:** next@16, react@19, tailwindcss@4 (`@import "tailwindcss"`), `next/font/google` (Bricolage Grotesque + Inter + JetBrains Mono), `swr` (data revalidation), `lucide-react` (icons only), `motion` (Framer Motion successor — `motion/react`), `three` + `@types/three` (signature scenes only), `clsx` + `tailwind-merge`. **Removed:** `@shadcn/react`, `@base-ui/react`, `recharts`, `shadcn` CLI, `cmdk`, `sonner`, `react-day-picker`, `embla-carousel-react`, `react-resizable-panels`, `inputLine truncated
---
## 0. Design System (the creative core — read before coding)
**Persona:** Controlled UX Designer + a tactical "ops console" voice. Material honesty: hierarchy via **scale/weight/tonal blocks**, NOT borders/shadows. Per `frontend-design` skill principle — spend the boldness in ONE signature place per page, keep the rest disciplined. Motion is choreography, not confetti: **one orchestrated moment per page**, everything else quiet.
### Palette (warm, signal-driven — NO teal/cyan/purple/blue gradients)
Light mode (`:root`):
```
--canvas: oklch(0.96 0.012 80) /* warm off-white, not cream */
--surface: oklch(0.92 0.014 80) /* tonal block, replaces bordered card */
--surface-2: oklch(0.88 0.016 80)
--ink: oklch(0.22 0.02 70) /* primary text */
--ink-soft: oklch(0.46 0.02 70) /* secondary text */
--hairline: oklch(0.22 0.02 70 / 0.10) /* structural rules ONLY, sparse */
--signal: oklch(0.78 0.17 125) /* lime — OK / live / primary accent */
--signal-ink: oklch(0.20 0.03 70) /* text ON signal */
--amber: oklch(0.80 0.15 70) /* WARN */
--vermilion: oklch(0.62 0.21 25) /* FLAGGED / destructive */
--ring: var(--signal)
```
Dark mode (`.dark`, default theme per `next-themes`):
```
--canvas: oklch(0.13 0.015 70) /* warm charcoal, not blue-black */
--surface: oklch(0.18 0.02 70)
--surface-2: oklch(0.23 0.022 70)
--ink: oklch(0.93 0.01 75)
--ink-soft: oklch(0.62 0.02 75)
--hairline: oklch(1 0 0 / 0.09)
--signal: oklch(0.88 0.18 125)
--signal-ink: oklch(0.18 0.03 70)
--amber: oklch(0.85 0.15 70)
--vermilion: oklch(0.68 0.21 25)
```
Three semantic signals reused everywhere: **lime = OK/live, amber = warn, vermilion = flag/danger**. This kills the purple-accent + teal-primary monotony.
### Typography (3 roles, deliberate pairing — not "Inter everywhere")
- **Display:** `Bricolage Grotesque` (700–800) — characterful grotesque for headers/big numbers.
- **Body/UI:** `Inter` (400–600).
- **Data/label:** `JetBrains Mono` (500/700) — all stats, timestamps, channel IDs, metrics.
Load all three via `next/font/google` with CSS variables (keep current `--font-inter`/`--font-jetbrains-mono` names + add `--font-display`).
### Layout & signature
- **No `card` with border.** Use tonal `--surface` blocks with generous radius (`--r: 14px`) and internal padding; separate blocks with whitespace + sparse hairlines only where structurally meaningful.
- **Signature element = "scan-tick":** a 1px animated pulse line (CSS keyframe `scan`) that marks every live/section header — NOT a card outline. Global canvas carries a faint warm dot-grid texture (low opacity) instead of the current bluish dotted radial-gradient.
- **Nav = left "spine":** vertical rail of icon nodes joined by a hairline; active node gets a `--signal` dot + label reveal. Collapses to a bottom tab-bar < 768px (CSS only, no JS sidebar primitive).
- **Header = "status bar":** connection state (WS dot), guild selector, live clock — mono font, reads like an instrument readout.
## 0.1 Motion & 3D Layer (the "lebih kreatif" addition)
### Motion rules (from `motion/react`)
- **Page transitions:** one shared `RouteTransition` in `(dashboard)/layout.tsx` — `AnimatePresence mode="popLayout"` + `motion.div key={pathname}` (fade + 8px rise + slight blur-out, ~220ms, `easeOut`). Consistent everywhere, zero per-page boilerplate.
- **Enter choreography (per page, ONE signature moment):** staggered rise-in for the hero/ticker group using `staggerChildren` variants; afterwards, quiet springs for hover/tap (`scale: 1.03` on interactive blocks, `whileTap` on buttons).
- **Layout animation:** `layout` prop on list items (messages rows, recording rows, queue) so add/remove/filter reflows smoothly; `layoutId` for shared-element transitions (ticker → detail modal on dashboard).
- **Live pulse:** `motion` drives the severity ticks / speaker rings with springs, not CSS `transition` alone.
- **`useReducedMotion()`** (from `motion/react`) gates ALL heavy motion; CSS `@media (prefers-reduced-motion: reduce)` additionally kills `scan`/`spin-disc` keyframes. Accessibility floor, non-negotiable.
- **No scroll-jacking, no marquee loops, no per-element confetti.** One moment per page. (`frontend-design` skill: "Satu momen orkestrasi biasanya lebih mengena daripada efek tersebar.")
### 3D rules (raw `three`, no R3F — lean bundle)
- Exactly **two** scenes, chosen because they carry real data meaning: **Dashboard hero** (`SignalField` — a particle field whose pulse density reflects live activity) and **Voice page** (`OrbField` — speakers as glowing orbs whose height/ring radius reacts to who is speaking). Everything else stays 2D/motion.
- **Lazy + client-only:** `next/dynamic(() => import("./SignalField"), { ssr: false, loading: () => <StaticFallback/> })`. Three ships in its own chunk, loaded only on those two routes.
- **WebGL guard:** if `!window.WebGLRenderingContext` or context creation fails → render the static SVG/CSS fallback (a stylized 2D version of the same visual). Never blank.
- **Perf guardrails:** `dpr: [1, 1.75]`, `powerPreference: "high-performance"`, `antialias: true`; RAF loop paused on `document.hidden`; `dispose()` geometries/materials on unmount; particle count capped by `navigator.hardwareConcurrency` + viewport (`Math.min(900, w*h/2000)`).
- **Style:** warm palette ONLY — signal-lime particles, amber/vermilion for flag/warn states; fog + soft additive blending for glow (no harsh white lights, no metallic PBR).
- **Interactivity:** subtle pointer parallax (camera lerp toward cursor) + gentle idle rotation. No drag/drop, no raycasting menus.
### Per-page metaphor (kills monotony — each page feels different)
| Route | Metaphor | Signature visual | Motion / 3D |
|---|---|---|---|
| `/dashboard` | Live ops overview | **3D signal particle field** hero + asymmetric ticker blocks + radial moderation gauge | **3D SignalField** (reacts to activity), staggered ticker rise-in, gauge draws on mount |
| `/messages` | Transcript | Left channel **timeline spine**; right = flowing message entries with left **severity tick** (no bordered cards); search = command palette | `layout` on message rows, spring severity ticks, palette types in |
| `/voice` | Stage | **3D orb field** of speakers + equalizer rings; activity = horizontal **session ribbon** | **3D OrbField** (speaking → orb rises + ring pulses), session ribbon draws sequentially |
| `/media` | Turntable | Rotating **disc** now-playing; queue = borderless list | CSS 3D disc spin (spring on play/pause), queue `layout` reflow, progress bar springs |
| `/recordings` | Tape library | Rows with **waveform thumbnail** (custom SVG from duration) | Waveform bars spring on hover; new upload animates in (AnimatePresence) |
| `/moderation` | Security log | Vertical **event flow** with status nodes (dot + line), not a table of cards | Nodes pulse on live action; timeline draws in sequence |
| `/analysis` | Query console | Terminal-style search panel | Typing cursor + results stagger |
---
## 1. What to KEEP (reuse verbatim — correct + integrates to backend)
- `src/lib/api/server.ts` — 11 server fetchers (`getDashboardStats`, `getActivity`, `getMediaStatus`, `getConfig`, `getModerationStats/Actions`, `getGuilds`, `getVoiceStatus`, `getRecordings`, `getMessages`). **No change.**
- `src/lib/api/client.ts` — `apiRequest` + `ApiError`. **No change.**
- `src/lib/ws/*` — `connection.ts`, `context.tsx`, `types.ts` (17 typed events: `message_created/updated/deleted/analyzed`, `voice_*`, `media_state`, `voice_pcm_data` binary). **No change.**
- `src/lib/types/*` — all interfaces. **No change.**
- `src/lib/format.ts`, `src/lib/utils.ts` (`cn`). **No change.**
- `src/lib/navigation.ts` — `navItems`. Keep but extend icons/labels if needed.
- Hooks: `src/hooks/*` (use-dashboard, use-media, use-messages, use-voice, use-moderation, use-recordings, use-guilds, use-config, use-chatbot-user, use-action, use-mobile), `src/lib/hooks/use-mounted.ts`. **Reuse** (verify no bad imports into deleted barrels).
- Feature logic kept but **reskinned**: `src/components/media/music-player.tsx`, `src/components/voice/*`, `src/components/chatbot/*`, `src/components/messages/*`, `src/components/recordings/*`, `src/components/moderation/moderation-section.tsx`, `src/components/analysis/search-panel.tsx`, `src/components/dashboard/*` (charts → rewritten as custom SVG).
## 2. What to DELETE
- `src/components/ui/*` (all 50 shadcn primitives).
- `components.json`, `@shadcn/react` + `@base-ui/react` + `shadcn` deps.
- `recharts` (replace with custom SVG chart helpers in `src/components/charts/`).
- `src/app/globals.css` → rewrite (no `@import "shadcn/tailwind.css"`, no `--color-primary` teal, no `.glass*`, no `.text-gradient`/`.gradient-border` teal/purple, no bluish body texture).
- `src/components/layout/app-sidebar.tsx` (shadcn Sidebar) → replace with custom `Spine` nav.
- All `page.tsx`+`view.tsx` pairs → merge into single `page.tsx` per route.
## 3. What to BUILD (new)
- New `globals.css` (tokens above + utilities + keyframes `scan`, `eq`, `fade-up`, `spin-disc`).
- `src/components/primitives/` — minimal custom: `Button`, `Input`, `Select`, `Dialog`, `Tooltip`, `Badge`, `Progress`, `Avatar`, `Skeleton`, `Toast`, `Sheet`.
- `src/components/motion/` — `RouteTransition.tsx`, `Stagger.tsx`, `variants.ts`.
- `src/components/three/` — `SignalField.tsx`, `OrbField.tsx`, `WebGLGuard.tsx`, `StaticFallback.tsx`, `useThreeScene.ts`.
- `src/components/charts/` — `Sparkline`, `AreaActivity`, `RadialGauge`, `SessionRibbon`, `Waveform`.
- `src/components/layout/` — `Spine.tsx`, `StatusBar.tsx`, `ThemeToggle.tsx` (reskin).
- 7 merged `page.tsx` files (one per route) implementing the metaphors above.
- `src/app/layout.tsx` — root (fonts + ThemeProvider + Toaster), `src/app/(dashboard)/layout.tsx` — providers + Spine + StatusBar + RouteTransition + MiniPlayer + Chatbot, `src/app/page.tsx` → redirect `/dashboard`.
---
## 4. Target File & Folder Structure (authoritative)
Everything below `services/frontend/src/` is the new tree. `REWRITE` replaces existing; `NEW` creates; `DELETE` removes. The `app/` route tree collapses `page.tsx`+`view.tsx` into single `page.tsx` files containing BOTH server fetch (default export) and client view (`"use client"` named export in same file).
```
services/frontend/
├─ package.json REWRITE (drop shadcn/base-ui/recharts; add motion, three, @types/three)
├─ pnpm-workspace.yaml REWRITE (onlyBuiltDependencies: keep build list minimal)
├─ components.json DELETE (shadcn registry config — no longer used)
├─ next.config.ts KEEP (output: standalone, trailingSlash, images.unoptimized)
├─ tsconfig.json KEEP (paths "@/*" → src/*, strict)
├─ postcss.config.mjs KEEP (@tailwindcss/postcss)
├─ biome.json KEEP
└─ src/
├─ app/
│ ├─ layout.tsx REWRITE (3 fonts + ThemeProvider + custom Toaster; rm sonner)
│ ├─ globals.css REWRITE (new token system §0; rm glass/teal/purple)
│ ├─ page.tsx KEEP (redirect → /dashboard/)
│ └─ (dashboard)/
│ ├─ layout.tsx REWRITE (providers + Spine + StatusBar + RouteTransition + MiniPlayer + Chatbot; rm shadcn Sidebar)
│ ├─ dashboard/ page.tsx REWRITE (server fetch + <DashboardView/> client; 3D SignalField hero)
│ ├─ messages/ page.tsx REWRITE (server seeds + <MessagesView/>; spine + severity ticks)
│ ├─ voice/ page.tsx REWRITE (server seeds + <VoiceView/>; 3D OrbField hero)
│ ├─ media/ page.tsx REWRITE (server seeds + <MediaView/>; turntable disc)
│ ├─ recordings/ page.tsx REWRITE (server seeds + <RecordingsView/>; waveform rows)
│ ├─ moderation/ page.tsx REWRITE (server seeds + <ModerationView/>; event-flow)
│ └─ analysis/ page.tsx REWRITE (client <AnalysisView/>; query console)
├─ components/
│ ├─ ui/ DELETE (all 50 shadcn primitives)
│ ├─ primitives/ NEW (Button, Input, Select, Dialog, Tooltip, Badge, Progress, Avatar, Skeleton, Toast, Sheet, index.ts)
│ ├─ motion/ NEW (variants.ts, Stagger.tsx, RouteTransition.tsx)
│ ├─ three/ NEW (WebGLGuard, SignalField, OrbField, StaticFallback, useThreeScene)
│ ├─ charts/ NEW (Sparkline, AreaActivity, RadialGauge, SessionRibbon, Waveform)
│ ├─ layout/ REWRITE (Spine NEW, StatusBar NEW, ThemeToggle REWRITE, app-sidebar DELETE)
│ ├─ dashboard/ REWRITE (stat-card DELETE; activity-chart/hourly/top-channels/moderation-donut/users/channels/reactions REWRITE)
│ ├─ messages/ REWRITE (message-card DELETE; message-list/detail/detail-view/ai-status-badge/ai-analysis-panel/attachments-grid/lightbox/search-overlay REWRITE)
│ ├─ voice/ REWRITE (voice-connection-card/connection-card/microphone-card DELETE; speaker-waveform/active-speakers-panel/activity-timeline/mic-control/listen-control REWRITE)
│ ├─ media/ REWRITE (music-player, mini-player REWRITE)
│ ├─ recordings/ REWRITE (recording-card DELETE; recording-player REWRITE)
│ ├─ moderation/ REWRITE (moderation-section REWRITE)
│ ├─ analysis/ REWRITE (search-panel REWRITE)
│ ├─ chatbot/ REWRITE (chatbot-container, chat-panel REWRITE; chatbot-context, index KEEP)
│ └─ shared/ REWRITE (empty-state, error-state, loading-skeleton, error-boundary, guild-selector REWRITE; index KEEP)
├─ hooks/ KEEP (verify no bad imports)
├─ lib/
│ ├─ api/ KEEP (server.ts, client.ts, index.ts)
│ ├─ ws/ KEEP (connection.ts, context.tsx, types.ts, ws-hook.ts)
│ ├─ types/ KEEP (all interfaces)
│ ├─ hooks/ KEEP (use-media-player.tsx, use-mounted.ts)
│ ├─ audio/ KEEP (voice PCM decode helpers if present)
│ ├─ format.ts KEEP
│ ├─ utils.ts KEEP (cn)
│ └─ navigation.ts KEEP
└─ (public assets) KEEP
```
### 4.1 Single-file page pattern (mandatory)
```tsx
// server component (default export) — runs on the server, fetches initial data
import { getX, getY } from "@/lib/api/server";
import { XView } from "./page"; // self-import of the named client export
export default async function Page() {
const [a, b] = await Promise.allSettled([getX(), getY()]);
return <XView initialA={a.status === "fulfilled" ? a.value : undefined}
initialB={b.status === "fulfilled" ? b.value : undefined} />;
}
// client component (named export) — hydrated, takes initialData as SWR fallback
"use client";
export function XView({ initialA, initialB }: Props) {
const { data } = useX(initialA); // SWR fallbackData = initialA
}
```
Self-referencing the named export keeps the file single-artifact while satisfying Next's RSC boundary (default = server, named = client). Tabs live inside `XView`.
### 4.2 Import rules (lint gate)
- No `@/components/ui/*` (deleted) — all UI via `@/components/primitives`.
- `three` only imported inside `src/components/three/*`; pages import those via `next/dynamic({ ssr: false })`.
- `motion` imported from `motion/react` only.
- All data: `@/lib/api/server` (server) / `@/lib/api/client` (client) — never invented endpoints.
---
## TASKS (granular — every file is its own task)
### PHASE 0 — Dependency surgery
- **T0.1** Edit `package.json`: remove `dependencies["@base-ui/react"]`.
- **T0.2** Remove `dependencies["@shadcn/react"]`.
- **T0.3** Remove `dependencies["shadcn"]`.
- **T0.4** Remove `dependencies["recharts"]`.
- **T0.5** Remove `dependencies["cmdk"]`.
- **T0.6** Remove `dependencies["sonner"]`.
- **T0.7** Remove `dependencies["react-day-picker"]`.
- **T0.8** Remove `dependencies["embla-carousel-react"]`.
- **T0.9** Remove `dependencies["react-resizable-panels"]`.
- **T0.10** Remove `dependencies["input-otp"]`.
- **T0.11** Add `dependencies["motion"]: "^12.0.0"`, `dependencies["three"]: "^0.180.0"`, `devDependencies["@types/three"]: "^0.180.0"`.
- **T0.12** `rm -f pnpm-lock.yaml && pnpm install` (regenerate lockfile).
- **T0.13** Verify `pnpm ls recharts @shadcn/react @base-ui/react` → empty; `pnpm ls motion three @types/three` → present.
- **T0.14** `cat pnpm-workspace.yaml`: confirm `onlyBuiltDependencies` keeps needed native builds, no broken shadcn postinstall.
### PHASE 1 — Design tokens (globals.css)
- **T1.1** Rewrite `@theme { }` head: light `:root` palette from §0 (canvas/surface/ink/ink-soft/hairline/signal/signal-ink/amber/vermilion/ring).
- **T1.2** Add radius tokens `--r: 14px`, `--r-panel: 12px`, `--r-control: 8px`, `--r-pill: 9999px`.
- **T1.3** Add `--font-display` token; keep `--font-sans`/`--font-mono`.
- **T1.4** Add `.dark { }` override block with §0 dark values (warm charcoal).
- **T1.5** Replace `@layer base body` bg: warm dot-grid `radial-gradient(oklch(0.45 0.03 70 / 0.05) 1px, transparent 1px)` + faint warm glow; remove old bluish radial layers.
- **T1.6** Retint scrollbar thumb to warm `oklch(0.4 0.02 70 / 0.2)`; keep `::selection` signal-tinted.
- **T1.7** Delete `.glass`, `.glass-elevated`, `.glass-intense`, `.dark .glass-intense` utilities.
- **T1.8** Delete `.text-gradient` and `.gradient-border`.
- **T1.9** Add `.surface` utility (bg var(--surface), radius var(--r), padding).
- **T1.10** Add `.scan-tick` (1px animated pulse line, keyframe `scan`).
- **T1.11** Add `.ticker`, `.pill`, `.mono` utilities.
- **T1.12** Add keyframes `scan`, `eq`, `fade-up`, `spin-disc` (keep used existing ones if still referenced).
- **T1.13** Add `@media (prefers-reduced-motion: reduce)` kill switch for scan/eq/spin-disc/pulse-ring/shimmer.
- **T1.14** Remove `@import "shadcn/tailwind.css";` (line 3); verify nothing else depends on shadcn CSS vars.
- **T1.15** Verify `grep -c "0.52 0.17 215\|0.55 0.2 280" src/app/globals.css` → `0`.
- **T1.16** `pnpm biome check src/app/globals.css` → no errors.
### PHASE 2 — Root layout + fonts
- **T2.1** In `layout.tsx` add `Bricolage_Grotesque` (`variable: "--font-display"`, subsets `["latin"]`, `display: "swap"`).
- **T2.2** Apply `inter.variable`, `jetbrainsMono.variable`, `bricolage.variable` to `<html>`.
- **T2.3** Remove `import { Toaster } from "@/components/ui/sonner"`.
- **T2.4** Comment out `<Toaster />` temporarily (re-enabled after T3.10).
- **T2.5** Keep `suppressHydrationWarning`, `ThemeProvider` (defaultTheme dark, enableSystem false).
- **T2.6** `npx tsc --noEmit` (fonts only; rest may still error until primitives exist).
### PHASE 3 — Custom primitives (replace 50 shadcn ui)
- **T3.1** `primitives/Button.tsx`: `motion.button`, variants `primary`/`ghost`/`danger`, `cn()` merge, `cursor-pointer`, `focus-visible:ring-2 ring-signal`, `whileTap` scale 0.97 gated by `useReducedMotion()`.
- **T3.2** `primitives/Input.tsx`: native `<input>`, `bg-surface`, `rounded-[var(--r-control)]`, `mono` prop.
- **T3.3** `primitives/Select.tsx`: native `<select>` styled, `bg-surface`.
- **T3.4** `primitives/Dialog.tsx`: native `<dialog>` + `showModal()`, warm `::backdrop`, `AnimatePresence`, `onClose`.
- **T3.5** `primitives/Tooltip.tsx`: CSS group-hover popover.
- **T3.6** `primitives/Badge.tsx`: tonal pill, `variant` → `bg-{tone}/15 text-{tone}` (signal/amber/vermilion/neutral).
- **T3.7** `primitives/Progress.tsx`: SVG track + `motion` fill, `value`/`max`, signal color.
- **T3.8** `primitives/Avatar.tsx`: `<img>` + initials fallback, signal bg, size prop.
- **T3.9** `primitives/Skeleton.tsx`: shimmer block (signal-tinted), `aria-hidden`.
- **T3.10** `primitives/Toast.tsx`: `ToastProvider` context + portal, `useToast()`, motion slide-in, auto-dismiss.
- **T3.11** `primitives/Sheet.tsx`: mobile drawer (`translate-x` spring), overlay, `open`/`onClose`.
- **T3.12** `primitives/index.ts` re-export all 11.
- **T3.13** Re-enable `<Toaster />` in `layout.tsx` (T2.4).
- **T3.14** Verify `npx tsc --noEmit` on primitives; `grep -rl "@/components/ui/" src/components/primitives` → empty.
### PHASE 4 — Motion foundation
- **T4.1** `motion/variants.ts`: export `spring`, `ease`, `fadeUp`, `stagger` (per §0.1).
- **T4.2** `motion/Stagger.tsx`: `StaggerGroup` + `StaggerItem` (`"use client"`).
- **T4.3** `motion/RouteTransition.tsx`: `"use client"`, `usePathname`, `AnimatePresence mode="popLayout"`, reduced-motion fallback to plain `<div>`.
- **T4.4** Verify `npx tsc --noEmit` on motion; `motion/react` import resolves.
### PHASE 5 — Custom SVG charts (replace recharts)
- **T5.1** `charts/Sparkline.tsx`: `<svg>` polyline from `points:number[]`, signal stroke, no axes.
- **T5.2** `charts/AreaActivity.tsx`: filled `<path>` area, low-opacity signal gradient, `pathLength` draw gated by reduced-motion.
- **T5.3** `charts/RadialGauge.tsx`: `<circle>` arc `stroke-dasharray`, center mono label.
- **T5.4** `charts/SessionRibbon.tsx`: horizontal segments per speaker duration.
- **T5.5** `charts/Waveform.tsx`: bars from deterministic seed, spring scaleY on hover.
- **T5.6** Verify `npx tsc --noEmit` on charts; no `recharts` import.
### PHASE 6 — Three.js foundation (lazy, guarded)
- **T6.1** `three/WebGLGuard.tsx`: `"use client"`, detect webgl2/webgl, render `children` or `fallback`.
- **T6.2** `three/useThreeScene.ts`: shared hook — renderer init (`dpr:[1,1.75]`, `powerPreference`), RAF with `document.hidden` pause, `dispose()` on unmount, resize observer.
- **T6.3** `three/SignalField.tsx`: `Points` BufferGeometry (~min(900, w*h/2000)), additive blend, signal-lime, fog, idle rotation + sine drift, `activity` prop, pointer parallax. Uses `useThreeScene`.
- **T6.4** `three/OrbField.tsx`: per-speaker `Sphere`, y-scale + ring lerp to speaking, tones signal/idle/vermilion.
- **T6.5** `three/StaticFallback.tsx`: 2D SVG/CSS silhouette for both scenes.
- **T6.6** Verify `grep -rl "from \"three\"" src | grep -v "components/three"` → empty; `npx tsc --noEmit` on three.
### PHASE 7 — Layout shell
- **T7.1** `layout/Spine.tsx`: `"use client"`, vertical rail from `navItems`, icon node + hairline, active signal dot + label reveal (motion spring), `max-md:` bottom tab-bar.
- **T7.2** `layout/StatusBar.tsx`: `"use client"`, page title + WS status dot (motion pulse) + `GuildSelector` + live clock (mono) + `ThemeToggle`.
- **T7.3** `layout/ThemeToggle.tsx`: restyle, keep `next-themes` logic, motion icon swap.
- **T7.4** Rewrite `(dashboard)/layout.tsx`: keep SWRConfig/WsProvider/MediaPlayerProvider/ChatbotProvider + sync functions verbatim; swap `AppSidebar`→`Spine`, header→`StatusBar`, wrap children in `RouteTransition`; remove `SidebarInset`/`SidebarTrigger`/`Separator`; keep MiniPlayer+ChatbotContainer.
- **T7.5** Delete `layout/app-sidebar.tsx`.
- **T7.6** Verify `grep -rl "components/ui/sidebar\|app-sidebar" src` → empty; `npx tsc --noEmit`.
### PHASE 8 — Dashboard page + components
- **T8.1** Rewrite `dashboard/page.tsx`: default async `getDashboardStats`+`getActivity` → `<DashboardView>`; named `"use client"` view with useStats/useActivity, 3D hero + tickers + tabs.
- **T8.2** Add `WebGLGuard`+`SignalField` hero with `activity` ratio; overlay headline (Bricolage) + `<RadialGauge>`.
- **T8.3** Build asymmetric ticker row with `StaggerGroup` + 4 `.surface` blocks (mono number + label + `<Sparkline>`); inline (replaces stat-card).
- **T8.4** Delete `dashboard/stat-card.tsx`.
- **T8.5** Reskin `dashboard/activity-chart.tsx` → `charts/AreaActivity` (daily).
- **T8.6** Reskin `dashboard/hourly-activity-chart.tsx` → `charts/AreaActivity` (hourly).
- **T8.7** Reskin `dashboard/top-channels-chart.tsx` → `charts/` + `.surface`.
- **T8.8** Reskin `dashboard/moderation-donut.tsx` → `charts/RadialGauge`.
- **T8.9** Reskin `dashboard/users-section.tsx` → `.surface`.
- **T8.10** Reskin `dashboard/channels-section.tsx` → `.surface`.
- **T8.11** Reskin `dashboard/reactions-section.tsx` → `.surface`.
- **T8.12** Verify `grep -rl "components/ui/card" src/app/\(dashboard\)/dashboard src/components/dashboard` → empty; `npx tsc --noEmit`.
### PHASE 9 — Messages page + components
- **T9.1** Rewrite `messages/page.tsx`: default `getMessages(guildId)`(+channels) → `<MessagesView>`; client spine + entries.
- **T9.2** Delete `messages/message-card.tsx`.
- **T9.3** Rewrite `messages/message-list.tsx`: left timeline spine + right severity-tick entries (`surface` + `border-l-2` lime/amber/vermilion, motion spring tick), `layout` reflow.
- **T9.4** Rewrite `messages/message-detail.tsx` → `.surface`.
- **T9.5** Rewrite `messages/message-detail-view.tsx` → `.surface` pane.
- **T9.6** Rewrite `messages/ai-status-badge.tsx` → `primitives/Badge`.
- **T9.7** Rewrite `messages/ai-analysis-panel.tsx` → `.surface`.
- **T9.8** Rewrite `messages/attachments-grid.tsx` → `.surface` grid.
- **T9.9** Rewrite `messages/lightbox.tsx` → `primitives/Dialog`.
- **T9.10** Rewrite `messages/search-overlay.tsx` → console palette, type-in animation, `primitives/Dialog`.
- **T9.11** Verify no `components/ui/card` in messages tree; `npx tsc --noEmit`.
### PHASE 10 — Voice page + components
- **T10.1** Rewrite `voice/page.tsx`: default `getVoiceStatus()` → `<VoiceView>`; client `WebGLGuard`+`OrbField` hero + ribbon + tabs.
- **T10.2** Delete `voice/voice-connection-card.tsx`, `connection-card.tsx`, `microphone-card.tsx`.
- **T10.3** Rewrite `voice/speaker-waveform.tsx` → SVG ring / eq bars.
- **T10.4** Rewrite `voice/active-speakers-panel.tsx` → `.surface`.
- **T10.5** Rewrite `voice/activity-timeline.tsx` → `charts/SessionRibbon`.
- **T10.6** Rewrite `voice/mic-control.tsx` → `primitives/Button`.
- **T10.7** Rewrite `voice/listen-control.tsx` → `primitives/Button`.
- **T10.8** Verify `npx tsc --noEmit`; no `components/ui/card` in voice tree.
### PHASE 11 — Media page + components
- **T11.1** Rewrite `media/page.tsx`: default `getMediaStatus()` → `<MediaView>`; client turntable disc + transport + queue.
- **T11.2** Rewrite `media/music-player.tsx`: CSS-3D disc (spin-disc, pause when not playing, spring on play/pause), mono meta, `primitives/Button` transport, `.surface` queue rows with `layout`.
- **T11.3** Rewrite `media/mini-player.tsx` → compact `.surface`.
- **T11.4** Verify `npx tsc --noEmit`; no `components/ui/card` in media tree.
### PHASE 12 — Recordings page + components
- **T12.1** Rewrite `recordings/page.tsx`: default `getRecordings(50)` → `<RecordingsView>`; client rows `AnimatePresence`+`layout`, live `voice_recording_uploaded` prepend.
- **T12.2** Delete `recordings/recording-card.tsx`.
- **T12.3** Rewrite `recordings/recording-player.tsx` → `.surface` row + `charts/Waveform` + `primitives/Button`/`Dialog` play/delete.
- **T12.4** Verify `npx tsc --noEmit`.
### PHASE 13 — Moderation + Analysis pages
- **T13.1** Rewrite `moderation/page.tsx`: default `getModerationStats/Actions` → `<ModerationView>`; client vertical event-flow, live actions pulse-in.
- **T13.2** Rewrite `moderation/moderation-section.tsx` → event-flow, no Card/table.
- **T13.3** Rewrite `analysis/page.tsx`: client `<AnalysisView/>` terminal console (`primitives/Input` mono + blinking caret, `.surface` results staggered).
- **T13.4** Rewrite `analysis/search-panel.tsx` → terminal style.
- **T13.5** Verify `npx tsc --noEmit`.
### PHASE 14 — Chatbot + shared + final cleanup
- **T14.1** Rewrite `chatbot/chatbot-container.tsx` → `.surface`, keep drag/minimize.
- **T14.2** Rewrite `chatbot/chat-panel.tsx` → bubbles via `AnimatePresence`, `primitives/*`.
- **T14.3** Keep `chatbot/chatbot-context.tsx` + `index.ts`.
- **T14.4** Rewrite `shared/empty-state.tsx` → tonal.
- **T14.5** Rewrite `shared/error-state.tsx`.
- **T14.6** Rewrite `shared/loading-skeleton.tsx` → `primitives/Skeleton`.
- **T14.7** Rewrite `shared/error-boundary.tsx`.
- **T14.8** Rewrite `shared/guild-selector.tsx` → `primitives/Select`.
- **T14.9** `rm -rf src/components/ui && rm -f components.json`.
- **T14.10** `grep -rn "components/ui/\|@shadcn\|@base-ui\|recharts" src` → MUST be empty.
- **T14.11** `grep -rl "from \"recharts\"\|@base-ui\|@shadcn" src` → empty (double-check).
- **T14.12** Verify `npx tsc --noEmit` across whole `src`.
### PHASE 15 — Build + lint gate
- **T15.1** `cd services/frontend && npx tsc --noEmit` → 0 errors.
- **T15.2** `pnpm biome check` → fix all issues (no `any` in new files).
- **T15.3** `pnpm build` (standalone) → success, emits `.next/standalone/server.js`.
- **T15.4** Inspect `.next/static/chunks/` for three-heavy chunk loaded only on dashboard/voice; confirm NOT in `/dashboard/` initial SSR HTML.
- **T15.5** Confirm `pnpm-lock.yaml` present (reproducible flake install).
- **T15.6** `grep -c "0.52 0.17 215\|0.55 0.2 280" .next/static/css/*.css` → 0.
### PHASE 16 — Local runtime smoke test (no prod)
- **T16.1** Start local standalone: `GMW_BACKEND_URL=http://127.0.0.1:4001 PORT=4017 node .next/standalone/server.js &`.
- **T16.2** `curl -s -o /dev/null -w "%{http_code}"` for all 7 routes → 200.
- **T16.3** `curl /dashboard/ | grep -o "Bricolage\|signal\|surface"` → present.
- **T16.4** `curl /_next/static/css/*.css | grep "0.52 0.17 215\|0.55 0.2 280"` → empty.
- **T16.5** Headless browser `/dashboard/`+`/voice/` WebGL on: no console errors, `<canvas>` present, `<StaticFallback/>` NOT rendered.
- **T16.6** Same pages WebGL off: `<StaticFallback/>` renders, no crash.
- **T16.7** Kill local server. Do NOT touch prod unit.
- **T16.8** Confirm 7 routes 200 + no console errors in T16.5/16.6.
### PHASE 17 — Flake + staging deploy
- **T17.1** Inspect `flake.nix` frontend drv: `filterSource` ignores `out/.next/node_modules`; `pnpm-lock.yaml` included.
- **T17.2** `nix build .#gmw-frontend --impure --sandbox-off` → succeeds.
- **T17.3** `nix copy` frontend drv to VPS into staging profile (test port e.g. 4217).
- **T17.4** Create/adjust staging systemd unit with `PORT=4217` exported BEFORE `node server.js` (LIDM PORT bug).
- **T17.5** `sudo systemctl restart gmw-frontend-staging`; `curl` staging → 200.
- **T17.6** Browser-check staging `/dashboard/`+`/voice/` (3D visible, fallback test).
- **T17.7** Verify staging CSS has no old teal/purple; new design renders.
### PHASE 18 — Production swap (CONFIRM WITH USER FIRST)
- **T18.1** STOP — send staging screenshots/URL; await explicit approval before touching prod.
- **T18.2** On approval: `nix-env --profile /nix/var/nix/profiles/gmw-frontend --set <new-drv>`.
- **T18.3** Confirm `gmw-frontend.service` exports `PORT=4017` before exec.
- **T18.4** `sudo systemctl restart gmw-frontend`.
- **T18.5** `curl` all 7 routes on `https://imphnen.asepharyana.my.id` → 200.
- **T18.6** Browser verify prod: new design, old teal gone, 3D scenes render.
- **T18.7** `journalctl -u gmw-frontend -f` 5 min; confirm WS reconnect + live features.
- **T18.8** Notify user with before/after notes; keep rollback plan (`nix-env --set <previous>; systemctl restart`).
---
## Risks / Trade-offs
- **Scope:** 7 pages + charts + primitives + motion + 2 three scenes + layout. Big but mechanical; each task is isolated (~90 atomic tasks).
- **Bundle weight:** `three` adds ~150KB gz but ONLY on dashboard/voice routes (lazy chunk, `ssr:false`). `motion` ~35KB gz app-wide — acceptable.
- **WebGL compatibility:** covered by `WebGLGuard` + static fallback. Old devices / strict privacy browsers never blank.
- **Motion excess:** risk of "AI-generated" scattered animation. Guard: one signature moment per page, shared variants, reduced-motion gates.
- **Feature regressions:** Voice PCM playback, media transport, chatbot drag — logic preserved, only skin changes. Smoke test (T16) catches SSR breaks; live WS/3D needs real backend (staging T17).
- **Removed deps:** dropping `recharts`/`sonner`/`cmdk` means rewriting charts + toasts + search palette — accounted for in Phases 3/5/9.
- **Next standalone PORT bug:** `server.js` may not read `PORT` — ensure unit exports `PORT=4017` before exec (T17.4/T18.3).
- **three + React 19:** raw three avoids R3F compat surface; lifecycle (dispose + RAF) handled in T6.2.
- **next-themes:** keep (light/dark toggle); default dark.
## Open questions (answer before T18)
- Q1: Deploy to prod now or staging-only first? (Recommend staging + screenshot review.)
- Q2: Keep `react-day-picker`/`embla` if any page still needs them? (Plan assumes no — verify in T14.10 grep.)
- Q3: Any brand name/wordmark change from "Discord Automod"? (Keep "Bete" identity unless told.)
- Q4: 3D depth — full 3D scenes on dashboard+voice as specced, or also a 3D accent on media (disc)? (Default: dashboard+voice only; media disc stays CSS 3D.)
+9 -3
View File
@@ -35,12 +35,15 @@
"suspicious": {
"noUnknownAtRules": "off",
"useIterableCallbackReturn": "off",
"noArrayIndexKey": "warn"
"noArrayIndexKey": "off",
"noExplicitAny": "off"
},
"a11y": {
"useSemanticElements": "off",
"useButtonType": "off",
"noAutofocus": "off"
"noAutofocus": "off",
"useMediaCaption": "off",
"noStaticElementInteractions": "off"
},
"performance": {
"noImgElement": "warn"
@@ -50,7 +53,10 @@
},
"correctness": {
"noInvalidUseBeforeDeclaration": "off",
"noUnusedFunctionParameters": "warn"
"noUnusedFunctionParameters": "warn",
"noUnusedVariables": "warn",
"noUnusedImports": "warn",
"noUnusedPrivateClassMembers": "warn"
}
},
"domains": {
+24 -54
View File
@@ -11,11 +11,6 @@
let
pkgs = import nixpkgs { inherit system; };
# libdatachannel for the GoLive N-API binding. nixpkgs 0.24.1 is built
# against this host's glibc and ships both lib + dev headers, so the
# binding links cleanly inside the Nix sandbox (no manual cmake build).
libdatachannel = pkgs.libdatachannel;
# Source filter: `path:` literals do NOT respect .gitignore by default,
# so a dirty local out/ (stale chunks from previous builds) leaks into
# the sandbox. Filter out build artifacts explicitly.
@@ -53,8 +48,8 @@
export GIT_SSL_CAINFO=${pkgs.cacert}/etc/ssl/certs/ca-bundle.crt
export NIX_SSL_CERT_FILE=${pkgs.cacert}/etc/ssl/certs/ca-bundle.crt
# pnpm uses node-gyp for native addons — provide build tools
export npm_config_build_from_source=true
# pnpm uses node-gyp for native addons — provide build tools (kept for
# the rare case a prebuilt is unavailable and it falls back to compile).
export CPPFLAGS="-I${pkgs.lib.getDev pkgs.openssl}/include"
export LDFLAGS="-L${pkgs.lib.getLib pkgs.openssl}/lib"
@@ -73,7 +68,7 @@
# NOTE: do NOT use `pnpm install --prod` here — it collapses the
# public-hoist dir (.pnpm/node_modules) that runtime peer resolution
# relies on (e.g. @lng2004/node-datachannel and @seydx/node-av-linux-x64
# are only reachable through it), silently breaking voice/screenshare.
# are only reachable through it), silently breaking voice.
# Instead we keep the full install's symlink layout and only prune
# orphaned package dirs + broken symlinks.
# Must run AFTER tsc (typescript is a devDep) and after native builds.
@@ -167,8 +162,7 @@ WRAPPER
pkgs.pkg-config
pkgs.openssl
pkgs.openssl.dev
libdatachannel.dev # rtc/rtc.hpp headers for the GoLive binding
pkgs.git # libdatachannel FetchContent clones from GitHub
pkgs.git # for any FetchContent-based deps during native builds
pkgs.cacert
];
@@ -181,38 +175,31 @@ WRAPPER
# do NOT let stdenv run its own cmake configure phase on the source.
dontUseCmakeConfigure = true;
# The gateway bundles native node_modules (.node addons plus .o/.a
# object files left in prebuilt dirs). stdenv's fixupPhase walks
# $out/node_modules and runs patchELF + shrinkELF over every ELF it
# finds, choking on the non-ET_DYN files (.o/.a) and the prebuilt
# .node addons — emitting hundreds of harmless "patchelf: wrong ELF
# type" lines per build. The real binary is node (external, already
# RPATH-fixed in its own derivation) and the .node addons are
# self-contained prebuilts loaded via dlopen, so Nix's fixup pass is
# neither needed nor wanted here. Skip it entirely.
dontFixup = true;
buildPhase = pnpmInstall + ''
echo "=== Building native voice deps ==="
# pnpm rebuild aborts on the first failing package and runs scripts
# from the wrong cwd — build each native dep explicitly with its own
# install script. Each failure is tolerated (|| true); the packages
# that matter (opus) are verified at runtime.
for pkg in \
node_modules/.pnpm/@discordjs+opus@*/node_modules/@discordjs/opus
do
if [ -d "$pkg" ]; then
echo "--- native build: $pkg ---"
(cd "$pkg" && npm run install 2>&1 || true)
fi
done
echo "=== Building libdatachannel-min N-API binding ==="
# The GoLive screen-share stack uses a minimal N-API binding
# (native/libdatachannel-min) over nixpkgs libdatachannel.
(
cd native/libdatachannel-min
# binding.gyp resolves include/lib from env (LDC_INCLUDE = .dev
# include root, LDC_LIB = lib output dir, NAPI_INCLUDE =
# node-addon-api include root).
NAPI_INCLUDE=$(find ../../node_modules/.pnpm -maxdepth 3 \
-type d -path "*node_modules/node-addon-api" | head -1)
echo "NAPI_INCLUDE=$NAPI_INCLUDE"
LDC_INCLUDE=${libdatachannel.dev} LDC_LIB=${libdatachannel.out}/lib/libdatachannel.so.0.24.1 \
NAPI_INCLUDE=$NAPI_INCLUDE \
npx node-gyp rebuild 2>&1 || true
ls -la build/Release/datachannel_min.node 2>/dev/null \
&& echo "libdatachannel-min binding OK: $(stat -c%s build/Release/datachannel_min.node) bytes" \
|| echo "WARN: libdatachannel-min binding build FAILED (screen share disabled)"
)
# @discordjs/opus ships prebuilt binaries for Node 22 (ABI node-v127,
# linux-x64-glibc-2.35) — node-pre-gyp downloads the prebuilt .node
# instead of compiling C++ from source. With build_from_source unset
# (above), `pnpm rebuild` runs the package's own install script which
# fetches the matching prebuilt; it only falls back to a source build
# if the download fails. This keeps voice working without a per-build
# native compile.
echo "=== Rebuilding @discordjs/opus (prebuilt download) ==="
pnpm rebuild @discordjs/opus 2>&1 || true
echo "=== Compiling TypeScript ===="
npx tsc 2>&1
echo "=== Fixing @/ path aliases to relative paths ==="
@@ -247,22 +234,6 @@ WRAPPER
mkdir -p $out/lib/gmw-discord-gateway
cp -r dist node_modules package.json tsconfig.json $out/lib/gmw-discord-gateway/
# GoLive native binding — loadNative resolves it relative to
# dist/goLive/native.js, i.e. <root>/native/libdatachannel-min/
# build/Release/datachannel_min.node; libdatachannel .so must sit
# next to it and be on LD_LIBRARY_PATH at runtime.
mkdir -p $out/lib/gmw-discord-gateway/native/libdatachannel-min/build/Release
cp native/libdatachannel-min/build/Release/datachannel_min.node \
$out/lib/gmw-discord-gateway/native/libdatachannel-min/build/Release/ 2>/dev/null || true
mkdir -p $out/lib/gmw-discord-gateway/native/libdatachannel-min/build/ldc
cp -rL native/libdatachannel-min/build/ldc/libdatachannel.so* \
$out/lib/gmw-discord-gateway/native/libdatachannel-min/build/ldc/ 2>/dev/null || true
# If the binding failed to build, screen share is simply disabled —
# the gateway itself must still start.
if [ ! -f $out/lib/gmw-discord-gateway/native/libdatachannel-min/build/Release/datachannel_min.node ]; then
echo "WARN: datachannel_min.node missing — GoLive screen share disabled in this build"
fi
# Also include drizzle migrations if they exist
cp -r drizzle $out/lib/gmw-discord-gateway/ 2>/dev/null || true
@@ -271,7 +242,6 @@ WRAPPER
#!${pkgs.runtimeShell}
cd $out/lib/gmw-discord-gateway
export PATH=${pkgs.ffmpeg-headless}/bin:${pkgs.yt-dlp}/bin:\$PATH
export LD_LIBRARY_PATH=${libdatachannel.out}/lib:\$LD_LIBRARY_PATH
exec ${nodejs}/bin/node dist/index.js
WRAPPER
chmod +x $out/bin/gmw-discord-gateway
+10
View File
@@ -0,0 +1,10 @@
-- Migration: add ai_analysis_duration_ms to messages
-- Tracks how long the AI moderation LLM call took, per message (ms).
-- Idempotent: safe to re-run.
--
-- Run against the production GMW database, e.g.:
-- PGPASSWORD=*** psql -h 100.121.180.82 -p 6432 -U asephs -d dcbot \
-- -f scripts/add-ai-analysis-duration.sql
ALTER TABLE "messages"
ADD COLUMN IF NOT EXISTS "ai_analysis_duration_ms" BIGINT;
@@ -77,12 +77,11 @@ export class MessagesRepository {
// Exclude spam threads (NULL-safe: non-thread messages are kept)
if (EXCLUDED_THREAD_IDS.length > 0) {
conditions.push(
or(
isNull(pgMessagesTable.thread_id),
notInArray(pgMessagesTable.thread_id, EXCLUDED_THREAD_IDS),
)!,
const excludeThreads = or(
isNull(pgMessagesTable.thread_id),
notInArray(pgMessagesTable.thread_id, EXCLUDED_THREAD_IDS),
);
if (excludeThreads) conditions.push(excludeThreads);
}
const where = conditions.length > 0 ? and(...conditions) : undefined;
@@ -150,12 +149,11 @@ export class MessagesRepository {
// Exclude spam threads (NULL-safe)
if (EXCLUDED_THREAD_IDS.length > 0) {
conditions.push(
or(
isNull(pgMessagesTable.thread_id),
notInArray(pgMessagesTable.thread_id, EXCLUDED_THREAD_IDS),
)!,
const excludeThreads = or(
isNull(pgMessagesTable.thread_id),
notInArray(pgMessagesTable.thread_id, EXCLUDED_THREAD_IDS),
);
if (excludeThreads) conditions.push(excludeThreads);
}
const rows = await db
@@ -316,12 +314,13 @@ export class MessagesRepository {
like(pgAttachmentsTable.type, "image/%"),
// Exclude spam threads (NULL-safe for non-thread messages)
...(EXCLUDED_THREAD_IDS.length > 0
? [
or(
? (() => {
const excludeThreads = or(
isNull(pgAttachmentsTable.thread_id),
notInArray(pgAttachmentsTable.thread_id, EXCLUDED_THREAD_IDS),
)!,
]
);
return excludeThreads ? [excludeThreads] : [];
})()
: []),
),
)
@@ -62,6 +62,9 @@ export const pgMessagesTable = pgTable(
enum: ["none", "monitor", "warn", "review", "delete", "escalate"],
}),
ai_analyzed_at: pgBigint("ai_analyzed_at", { mode: "number" }),
ai_analysis_duration_ms: pgBigint("ai_analysis_duration_ms", {
mode: "number",
}),
ai_error: pgText("ai_error"),
},
(table) => ({
@@ -80,6 +80,7 @@ export interface MessageRecord {
ai_confidence?: number | null;
ai_recommended_action?: AIRecommendedAction | null;
ai_analyzed_at?: number | null;
ai_analysis_duration_ms?: number | null;
ai_error?: string | null;
}
+1 -1
View File
@@ -108,5 +108,5 @@ export async function retryWithBackoff<T>(
});
}
}
throw lastError!;
throw lastError ?? new Error("Request failed after all retries");
}
@@ -24,6 +24,7 @@ export interface MappedMessage {
ai_confidence: number | null;
ai_recommended_action: string | null;
ai_analyzed_at: number | null;
ai_analysis_duration_ms: number | null;
ai_error: string | null;
is_reply: boolean | null;
is_forward: boolean | null;
@@ -58,6 +59,8 @@ export function mapMessageRow(row: Record<string, unknown>): MappedMessage {
ai_confidence: (row.ai_confidence as number | null) ?? null,
ai_recommended_action: (row.ai_recommended_action as string | null) ?? null,
ai_analyzed_at: (row.ai_analyzed_at as number | null) ?? null,
ai_analysis_duration_ms:
(row.ai_analysis_duration_ms as number | null) ?? null,
ai_error: (row.ai_error as string | null) ?? null,
is_reply: row.is_reply === null ? null : Boolean(row.is_reply),
is_forward: row.is_forward === null ? null : Boolean(row.is_forward),
+121 -150
View File
@@ -1,172 +1,143 @@
# Discord Gateway — Architecture
Pure event-driven microservice (no HTTP server). Captures Discord
messages/voice/attachments/reactions/threads/presence, runs LLM-based AI
moderation, and publishes everything to Redis pub/sub for the backend to
consume. The backend serves the HTTP/WS API to the frontend.
> NOTE: this doc is the source of truth for the module layout. The older
> `MODULE_STRUCTURE.md` was stale (referenced `winston`, `mock-crc.ts`,
> `indonesianTextNormalizer.ts`, and `aiAnalysisWorker.ts`/`llmModerationClient.ts`
> which were renamed/merged). If they disagree, this file wins.
## Top-level layout
```
services/discord-gateway/
├── src/
│ ├── index.ts # Entry point → initializeDiscordGateway()
│ ├── app/
│ │ ├── bootstrap.ts # Discord Gateway initialization (no HTTP server)
│ │ └── shutdown.ts # Graceful shutdown handler
│ │ ├── bootstrap.ts # Wires client, DB, Redis, workers, schedulers
│ │ ├── shutdown.ts # Graceful shutdown (SIGINT/SIGTERM + transient errors)
│ │ └── retention.ts # Expired-record cleanup scheduler
│ ├── shared/
│ │ ├── config/
│ │ │ └── config.ts # Environment configuration (Zod validated)
│ │ ├── database/
│ │ │ ├── schema.ts # Drizzle ORM schema
│ │ │ ├── drizzle.ts # Database connection
│ │ │ ├── migrate.ts # Migration runner
│ │ │ └── voiceRecordingRepo.ts
│ │ ├── errors/
│ │ │ └── errors.ts # Custom error classes
│ │ ├── logger/
│ │ │ ├── logger.ts # Winston logger wrapper
│ │ │ └── serialization.ts # Log value serialization
│ │ ├── utils/
│ │ │ └── retry.ts # Retry with backoff utility
│ │ └── discord/
│ │ └── clientOptions.ts # Discord.js client configuration
│ ├── modules/
│ │ ├── message-capture/ # Modular MVC: Message capture & storage
│ │ │ ├── messageCapture.ts # Controller: Discord event listeners
│ │ │ ├── messageStore.ts # Repository: Database operations
│ │ │ ├── messageMetadata.ts # Service: Message metadata extraction
│ │ │ ├── types.ts # Domain types
│ │ │ └── index.ts # Module exports
│ │ ├── ai-moderation/ # Modular MVC: AI analysis & moderation
│ │ │ ├── aiAnalyzer.ts # Controller: Analysis orchestration
│ │ │ ├── llmModerationClient.ts # Service: LLM API client
│ │ │ ├── aiAnalysisWorker.ts # Service: Worker pool management
│ │ │ ├── indonesianTextNormalizer.ts # Service: Text normalization
│ │ │ ├── moderationPrompt.ts # Service: Prompt generation
│ │ │ └── index.ts # Module exports
│ │ ├── voice-recording/ # Modular MVC: Voice recording & streaming
│ │ │ ├── voiceController.ts # Controller: Voice connection management
│ │ │ ├── recorder.ts # Service: Recording orchestration
│ │ │ ├── recorder/
│ │ │ │ ├── audioStream.ts # Service: Audio stream subscription
│ │ │ │ ├── decoder.ts # Service: Opus decoding
│ │ │ │ ├── segment.ts # Service: OGG segment rotation
│ │ │ │ ├── metadata.ts # Service: Segment metadata
│ │ │ │ ├── sessionRecording.ts # Service: Session management
│ │ │ │ └── uploader.ts # Service: Segment upload
│ │ │ └── index.ts # Module exports
│ │ ├── attachment-upload/ # Modular MVC: Attachment handling
│ │ │ ├── attachmentUploader.ts # Service: Upload orchestration
│ │ │ ├── imageResizer.ts # Service: Image resizing
│ │ │ └── index.ts # Module exports
│ │ └── event-broadcaster/ # Event-driven: Redis pub/sub
│ │ ├── eventBroadcaster.ts # Service: Event publishing
│ │ ├── eventTypes.ts # Domain: Event type definitions
│ │ └── index.ts # Module exports
│ ├── mock-crc.ts # CRC polyfill for discord.js
│ └── index.ts # Service entry point
├── package.json # Service dependencies
└── tsconfig.json # TypeScript configuration
│ │ ├── config/ # Zod-validated env (index.ts = schema+loader)
│ │ ├── database/ # Drizzle ORM + pg Pool + migrations
│ │ │ ├── init.ts drizzle.ts pool.ts migrate.ts migrateCli.ts
│ │ │ └── schema/ # messages, cache, voice, analytics, meta
│ │ ├── logger/ # pino wrapper + createChildLogger()
│ │ ├── errors/ # AppError / ConfigError / AudioError ...
│ │ ├── utils/ # retry, pagination
│ │ ├── discord/clientOptions.ts # discord.js-selfbot-v13 client options
│ │ ├── uploader.ts # Shared attachment upload helper
│ │ ├── redis-channels.ts # Redis channel-name constants
│ │ └── moderation-types.ts # Shared AI analysis domain types
│ └── modules/
│ ├── message-capture/ # Discord event listeners + DB store
│ ├── ai-moderation/ # LLM moderation pipeline (see below)
│ ├── voice-recording/ # Voice connect + Opus→OGG recording
│ │ └── recorder/ # decoder, segment, session, uploader, oggCrc
│ ├── voice-pcm-ws/ # Real-time PCM → backend WebSocket (bypasses Redis)
│ ├── attachment-upload/ # Download + (sharp) resize + upload
│ ├── event-broadcaster/ # RedisEventPublisher + EventBroadcaster
│ ├── command-handler/ # Redis-subscribed backend→gateway commands
│ ├── reaction-tracking/ thread-tracking/ user-presence/
│ ├── channel-topic/ guild-member-events/
│ └── gateway-metrics/ # Prometheus /metrics endpoint (port 4016)
```
## Architecture Patterns
## AI moderation pipeline (`ai-moderation/`)
### Modular MVC Structure
Each module follows Controller-Service-Repository pattern:
- **Controller**: Discord event listeners (messageCapture, aiAnalyzer, voiceController)
- **Service**: Business logic (messageStore, llmModerationClient, recorder)
- **Repository**: Data access (messageStore, voiceRecordingRepo)
LLM-only judge — no regex/heuristic classification. One orchestrator call
handles a whole batch (text + media split internally, parallel paths).
### Event-Driven Design
- **Redis Pub/Sub**: All events published to Redis channels
- **Event Channels**:
- `discord:message:created` — New message captured
- `discord:message:updated` — Message edited
- `discord:message:deleted` — Message deleted
- `discord:message:analyzed` — AI analysis complete
- `discord:attachment:created` — Attachment detected
- `discord:attachment:uploaded` — Attachment uploaded to storage
- `discord:voice:started` — Voice recording started
- `discord:voice:stopped` — Voice recording stopped
- `discord:voice:uploaded` — Voice segment uploaded
- `discord:analysis:queue_status` — Analysis queue status update
- `aiAnalyzer.ts` — public API: `queueMessageAnalysis`, `getAnalysisQueueStatus`,
`startPendingAIAnalysisWorker` (recovery worker + cache-prune).
- `batchScheduler.ts` — per-conversation debounce → `processBatch`.
- `batchProcessor.ts` — batch lock/circuit-breaker, fans failed targets to
individual fallback.
- `individualFallbackProcessor.ts` — one-message-at-a-time retry path, own CB.
- `conversationState.ts` / `circuitBreaker.ts` — per-conversation state,
Piscina `workerPool`, `getConversationKey`.
- `ai-analysis-worker.ts` — Piscina entry point (`batch` / `individual` jobs).
Runs `runModerationAnalysis` off the main thread.
- `moderationOrchestrator.ts` — exact-hash cache → batched semantic (Qdrant)
cache → LLM. Text and media paths run in parallel.
- `textBatchProcessor.ts` / `mediaBatchProcessor.ts` — actual LLM calls
(one call per sub-batch, not per message).
- `llmClient.ts` — central OpenAI-compatible chat client (streaming, retries,
thinking-disable injection). `visionAnalyzer.ts` / `mediaAnalysisClient.ts`
share the same router/base URL (different model alias for vision).
- `embeddingClient.ts` + `qdrantClient.ts` — semantic cache (one embed call +
one batched Qdrant search for all uncached targets).
- `textCacheStore.ts` / `channelCultureStore.ts` / `userProfileStore.ts` /
`userReputationStore.ts` — caches & learned per-channel/user state.
### Shared Infrastructure
- **Config**: Zod-validated environment variables
- **Logger**: Winston logger with context support
- **Database**: Drizzle ORM with PostgreSQL
- **Errors**: Custom error classes with codes and status codes
- **Utils**: Retry logic with exponential backoff
### Concurrency model
### No HTTP Server
- Discord Gateway service is **event-driven only**
- No Express, WebSocket, or HTTP routes
- All communication via Redis pub/sub
- Backend service consumes events and serves HTTP API
- Main thread owns the LLM semaphore (`AI_LLM_MAX_CONCURRENT`, default 5) via
`llmClient.withLlmConcurrency`.
- Piscina pool (`PISCINA_MAX_THREADS`, default 4) runs the heavy LLM work off
the event loop; **each worker thread initializes its own pg Pool** (min 0,
grows to `POSTGRES_POOL_MAX`). See "Memory & connections" below.
## Initialization Flow
## Memory & DB connections
1. Load environment config (Zod validation)
2. Initialize database connection
3. Run pending migrations
4. Create Discord client with optimized cache settings
5. Initialize Redis event broadcaster
6. Register Discord event listeners (messageCapture, aiAnalyzer)
7. Login to Discord
8. Listen for graceful shutdown signals (SIGINT, SIGTERM)
`MemoryMax=1G` (raised from 512M — live RSS sits at ~500 MiB, peak 508 MiB,
so 512M left ~2% headroom and risked an OOM-kill restart). Host has 8 GB free.
## Graceful Shutdown
`POSTGRES_POOL_MIN=0` (default). The gateway = main process + up to 4 Piscina
worker threads, each with its own pg Pool. With min:0 the pools stay empty
until a query runs and drop idle clients afterward, instead of holding
`(1 main + 4 workers) × 2 = 10` permanently-open idle connections against
PgBouncer. The pool still grows on demand up to `POSTGRES_POOL_MAX`.
On shutdown signal:
1. Close database connection
2. Disconnect from voice channels
3. Close Redis connection
4. Destroy Discord client
5. Exit process
## Event channels (Redis pub/sub)
## Dependencies
`discord:message:{created,updated,deleted,analyzed}`,
`discord:attachment:{created,uploaded}`,
`discord:voice:{started,stopped,uploaded,active_user,pcm,analyzed}`,
`discord:analysis:queue_status`,
`discord:reaction:{added,removed}`,
`discord:thread:{created,deleted,updated}`,
`discord:channel_topic:updated`,
`discord:presence:updated`,
`discord:guild_member:{added,removed}`.
See `src/shared/redis-channels.ts` for the canonical names.
**Core Discord**:
- discord.js-selfbot-v13
- @discordjs/voice
- @discordjs/opus
## Initialization flow
**Audio Processing**:
- prism-media (Opus encoding/decoding)
- opusscript (Opus fallback)
- sharp (Image resizing)
1. Validate env (Zod). Refuse to start if `AI_ANALYSIS_ENABLED` but no key.
2. `AUTO_MIGRATE_ON_STARTUP` → run pending Drizzle migrations.
3. `initializeDatabase()` (pg Pool, min 0).
4. Create discord.js-selfbot-v13 client; register listeners on `ready`.
5. Start `gmw-discord-gateway` metrics server (port `METRICS_PORT`, default 4016).
6. `client.login(token)`.
**Data & Config**:
- drizzle-orm (ORM)
- pg (PostgreSQL driver)
- zod (Config validation)
- ioredis (Redis client)
## Graceful shutdown
**Logging & Utilities**:
- winston (Structured logging)
- p-retry (Retry logic)
- p-limit (Concurrency limiting)
- piscina (Worker pool)
`SIGINT`/`SIGTERM` (and uncaught transient stream errors: EPIPE / ECONNRESET /
ERR_STREAM_DESTROYED / ERR_STREAM_WRITE_AFTER_END are treated as non-fatal):
stop metrics → stop muxer → disconnect voice → close PCM WS → close Redis →
close command handler → close DB → destroy client → exit.
## Event Flow Example
## Observability
### Message Capture Flow
1. Discord emits `messageCreate` event
2. `messageCapture.ts` listener receives event
3. Extract metadata (user, channel, content, timestamp)
4. `messageStore.ts` inserts into database
5. `eventBroadcaster.messageCreated()` publishes to Redis
6. Backend service subscribes to `discord:message:created` channel
7. Backend processes and stores in its own database
Prometheus scrapes `127.0.0.1:4016/metrics` (`bete_*` prefix). Collectors run
per-scrape and expose: process memory/uptime, and (when AI analysis is on) live
pipeline gauges — `ai_analysis_queued_conversations`,
`ai_analysis_active_batch_requests`, `ai_analysis_active_individual_requests`,
`ai_analysis_individual_in_flight`, `ai_analysis_individual_circuit_breaker_active`,
`ai_analysis_worker_threads`, `ai_analysis_worker_threads_active`.
### Voice Recording Flow
1. `voiceController.connect()` joins voice channel
2. `recorder.ts` subscribes to user audio streams
3. For each speaking user:
- Create audio stream subscription
- Decode Opus packets to PCM
- Rotate OGG segments (5s default)
- Collect user metadata
4. On silence (3s):
- Finalize segment
- Create metadata JSON
- Upload segment to storage
- Publish `discord:voice:uploaded` event
5. Backend service receives event and indexes recording
## Key invariants (do not break)
## No Breaking Changes
- Original `src/` remains untouched for now
- Discord Gateway is a **new service** in `services/discord-gateway/`
- Can run alongside existing monolith during transition
- Backend service will consume Redis events
- Frontend continues to use Backend HTTP API
- **LLM is the only judge.** Failed LLM → `status:"error"` + recovery retry.
Never reintroduce regex/heuristic content classification.
- **Discord tokens are sanitized** (`discordTokens.ts`: `<:emoji:id>` →
`[emoji:name]`, `<@id>` → `@user`, etc.) before content reaches the LLM, so
numeric snowflake IDs never trigger false positives.
- **Semantic cache is batched** (one embed call + one Qdrant batch search),
not N sequential round-trips. `ensureQdrantCollection` is memoized.
- **Streaming is mandatory** against the 9router base URL (non-stream waits for
the full body and times out). `llmClient` aggregates SSE chunks.
+60 -388
View File
@@ -1,408 +1,80 @@
# Discord Gateway Service - Module Structure
# Discord Gateway Service — Module Structure
## Complete Directory Tree
> Kept as a compact module map. For the authoritative layout, design
> decisions, and invariants, see `ARCHITECTURE.md`. This file was rewritten
> on 2026-08-16 to fix stale references (`winston` → pino,
> `mock-crc.ts`/`indonesianTextNormalizer.ts` removed,
> `aiAnalysisWorker.ts` → `ai-analysis-worker.ts`,
> `llmModerationClient.ts` → `llmClient.ts`).
## Top-level
```
services/discord-gateway/
├── src/
│ ├── app/
│ │ ├── bootstrap.ts
│ │ │ └── Initializes Discord client, database, Redis broadcaster
│ │ │ Registers event listeners, handles graceful shutdown
│ │ └── shutdown.ts
│ │ └── Graceful shutdown handler for SIGINT/SIGTERM/exceptions
│ │
│ ├── shared/
│ │ ├── config/
│ │ │ └── config.ts
│ │ │ └── Zod-validated environment configuration
│ │ │ - Discord token, database URL, Redis URL
│ │ │ - AI LLM settings, recording parameters
│ │ │ - Attachment upload settings, retention policies
│ │ │
│ │ ├── database/
│ │ │ ├── schema.ts
│ │ │ │ └── Drizzle ORM schema definitions
│ │ │ ├── drizzle.ts
│ │ │ │ └── PostgreSQL connection and initialization
│ │ │ ├── migrate.ts
│ │ │ │ └── Database migration runner
│ │ │ ├── migrateCli.ts
│ │ │ │ └── CLI for programmatic migrations
│ │ │ ├── voiceRecordingRepo.ts
│ │ │ │ └── Voice recording repository
│ │ │ └── migrations/
│ │ │ └── Database migration files
│ │ │
│ │ ├── errors/
│ │ │ └── errors.ts
│ │ │ └── Custom error classes
│ │ │ - AppError (base)
│ │ │ - ConfigError
│ │ │ - AudioError
│ │ │ - VoiceConnectionError
│ │ │ - ValidationError
│ │ │
│ │ ├── logger/
│ │ │ ├── logger.ts
│ │ │ │ └── Winston logger wrapper with context support
│ │ │ └── serialization.ts
│ │ │ └── Log value serialization utilities
│ │ │
│ │ ├── utils/
│ │ │ └── retry.ts
│ │ │ └── Retry with exponential backoff utility
│ │ │
│ │ └── discord/
│ │ └── clientOptions.ts
│ │ └── Discord.js client configuration
│ │
│ ├── modules/
│ │ │
│ │ ├── message-capture/
│ │ │ ├── messageCapture.ts
│ │ │ │ └── CONTROLLER: Discord event listeners
│ │ │ │ - messageCreate, messageUpdate, messageDelete
│ │ │ │ - Validates capture target, publishes events
│ │ │ │
│ │ │ ├── messageStore.ts
│ │ │ │ └── REPOSITORY: Database CRUD operations
│ │ │ │ - upsertMessageForCapture
│ │ │ │ - updateMessageAsEdited
│ │ │ │ - updateMessageAsDeleted
│ │ │ │ - insertAttachment
│ │ │ │ - getMessageById
│ │ │ │
│ │ │ ├── messageMetadata.ts
│ │ │ │ └── SERVICE: Message metadata extraction
│ │ │ │ - getMessageMetadata
│ │ │ │ - getMessageLocation
│ │ │ │ - getDisplayContent
│ │ │ │
│ │ │ ├── types.ts
│ │ │ │ └── Domain types
│ │ │ │ - MessageRecord
│ │ │ │ - AttachmentRecord
│ │ │ │ - VoiceSegmentRecord
│ │ │ │ - AIStatus, AISeverity, AIRecommendedAction
│ │ │ │
│ │ │ └── index.ts
│ │ └── Module exports
│ │
│ │ ├── ai-moderation/
│ │ │ ├── aiAnalyzer.ts
│ │ │ │ └── CONTROLLER: Analysis orchestration
│ │ │ │ - startPendingAIAnalysisWorker
│ │ │ │ - queueMessageAnalysis
│ │ │ │ - Manages analysis queue and worker pool
│ │ │ │
│ │ │ ├── llmModerationClient.ts
│ │ │ │ └── SERVICE: LLM API integration
│ │ │ │ - Calls LLM for text/image moderation
│ │ │ │ - Parses responses, handles errors
│ │ │ │ - Retry logic with backoff
│ │ │ │
│ │ │ ├── aiAnalysisWorker.ts
│ │ │ │ └── SERVICE: Worker pool management
│ │ │ │ - Piscina worker pool for parallel analysis
│ │ │ │ - Conversation context batching
│ │ │ │
│ │ │ ├── indonesianTextNormalizer.ts
│ │ │ │ └── SERVICE: Text preprocessing
│ │ │ │ - Normalize Indonesian text
│ │ │ │ - Handle diacritics, abbreviations
│ │ │ │
│ │ │ ├── moderationPrompt.ts
│ │ │ │ └── SERVICE: Prompt generation
│ │ │ │ - Generate LLM prompts for moderation
│ │ │ │ - Include context and policy
│ │ │ │
│ │ │ └── index.ts
│ │ └── Module exports
│ │
│ │ ├── voice-recording/
│ │ │ ├── voiceController.ts
│ │ │ │ └── CONTROLLER: Voice connection management
│ │ │ │ - connect(guildId, channelId)
│ │ │ │ - disconnect()
│ │ │ │ - listGuilds(), listVoiceChannels()
│ │ │ │ - getStatus()
│ │ │ │
│ │ │ ├── recorder.ts
│ │ │ │ └── SERVICE: Recording orchestration
│ │ │ │ - startRecording(client, channel)
│ │ │ │ - stopRecording(guildId)
│ │ │ │ - Manages active recording sessions
│ │ │ │
│ │ │ ├── recorder/
│ │ │ │ ├── audioStream.ts
│ │ │ │ │ └── SERVICE: Audio stream subscription
│ │ │ │ │ - subscribeToAudioStream
│ │ │ │ │ - Opus packet handling
│ │ │ │ │
│ │ │ │ ├── decoder.ts
│ │ │ │ │ └── SERVICE: Opus decoding
│ │ │ │ │ - OpusDecoder class
│ │ │ │ │ - Decode Opus to PCM
│ │ │ │ │ - Rotation and cooldown logic
│ │ │ │ │
│ │ │ │ ├── segment.ts
│ │ │ │ │ └── SERVICE: OGG segment rotation
│ │ │ │ │ - SegmentManager class
│ │ │ │ │ - Rotate segments (5s default)
│ │ │ │ │ - Write OGG files
│ │ │ │ │
│ │ │ │ ├── metadata.ts
│ │ │ │ │ └── SERVICE: Segment metadata
│ │ │ │ │ - collectUserMetadata
│ │ │ │ │ - createSegmentMetadata
│ │ │ │ │ - User info, roles, timestamps
│ │ │ │ │
│ │ │ │ ├── sessionRecording.ts
│ │ │ │ │ └── SERVICE: Session management
│ │ │ │ │ - createRecordingSession
│ │ │ │ │ - finalizeRecordingSession
│ │ │ │ │ - Track active sessions
│ │ │ │ │
│ │ │ │ └── uploader.ts
│ │ │ │ └── SERVICE: Segment upload
│ │ │ │ - uploadRecordingSegment
│ │ │ │ - Upload to external storage
│ │ │ │ - Retry logic
│ │ │ │
│ │ │ └── index.ts
│ │ └── Module exports
│ │
│ │ ├── attachment-upload/
│ │ │ ├── attachmentUploader.ts
│ │ │ │ └── SERVICE: Upload orchestration
│ │ │ │ - processAttachmentUpload
│ │ │ │ - Download from Discord
│ │ │ │ - Upload to external storage
│ │ │ │ - Retry with backoff
│ │ │ │
│ │ │ ├── imageResizer.ts
│ │ │ │ └── SERVICE: Image processing
│ │ │ │ - resizeImage
│ │ │ │ - Resize to max dimension
│ │ │ │ - Preserve aspect ratio
│ │ │ │
│ │ │ └── index.ts
│ │ └── Module exports
│ │
│ │ └── event-broadcaster/
│ │ ├── eventBroadcaster.ts
│ │ │ └── SERVICE: Redis pub/sub publisher
│ │ │ - EventBroadcaster class
│ │ │ - RedisEventPublisher class
│ │ │ - Publish to Redis channels
│ │ │ - Methods:
│ │ │ - messageCreated()
│ │ │ - messageUpdated()
│ │ │ - messageDeleted()
│ │ │ - messageAnalyzed()
│ │ │ - attachmentCreated()
│ │ │ - attachmentUploaded()
│ │ │ - voiceRecordingStarted()
│ │ │ - voiceRecordingStopped()
│ │ │ - voiceRecordingUploaded()
│ │ │ - analysisQueueStatus()
│ │ │
│ │ ├── eventTypes.ts
│ │ │ └── Domain types
│ │ │ - DiscordGatewayEvent interface
│ │ │ - EventChannels constants
│ │ │ - Event channel names
│ │ │
│ │ └── index.ts
│ └── Module exports
│
│ ├── mock-crc.ts
│ │ └── CRC polyfill for discord.js compatibility
│ │
│ └── index.ts
│ └── Service entry point
│ - Initialize Discord Gateway
│ - Handle startup errors
│
├── ARCHITECTURE.md
│ └── Detailed architecture documentation
│
├── README.md
│ └── Complete service documentation
│
├── MODULE_STRUCTURE.md
│ └── This file - module structure reference
│
└── package.json
└── Service dependencies and scripts
│ ├── index.ts # Entry point
│ ├── app/ # bootstrap, shutdown, retention
│ ├── shared/ # config, database, logger, errors, utils, discord, uploader
│ └── modules/
│ ├── message-capture/ # Discord listeners + DB store + metadata
│ ├── ai-moderation/ # LLM moderation pipeline (largest module)
│ ├── voice-recording/ # Voice connect + Opus→OGG recording (+ recorder/)
│ ├── voice-pcm-ws/ # Real-time PCM → backend WebSocket
│ ├── attachment-upload/ # Download + sharp resize + upload
│ ├── event-broadcaster/ # RedisEventPublisher + EventBroadcaster
│ ├── command-handler/ # Backend→gateway Redis commands
│ ├── reaction-tracking/ thread-tracking/ user-presence/
│ ├── channel-topic/ guild-member-events/
│ └── gateway-metrics/ # Prometheus /metrics (port 4016)
├── tests/ # Vitest suites (129 tests)
├── drizzle/ # Drizzle migration SQL + journal
├── ARCHITECTURE.md README.md package.json tsconfig.json vitest.config.ts
```
## Module Responsibilities
## Module responsibilities (summary)
### message-capture
**Purpose**: Capture Discord messages (create, update, delete)
**Pattern**: Controller-Service-Repository
- **Controller** (messageCapture.ts): Listens to Discord events
- **Service** (messageMetadata.ts): Extracts metadata
- **Repository** (messageStore.ts): Database operations
- **Events Published**:
- `discord:message:created`
- `discord:message:updated`
- `discord:message:deleted`
Captures `messageCreate`/`messageUpdate`/`messageDelete`, extracts metadata,
stores to Postgres, publishes to Redis. Controller–Service–Repository split:
`messageCapture.ts` (listener) → `messageStore.ts` (DB) + `messageMetadata.ts`
(service).
### ai-moderation
**Purpose**: Analyze messages with LLM for moderation
**Pattern**: Controller-Service-Service-Service
- **Controller** (aiAnalyzer.ts): Orchestrates analysis workflow
- **Service** (llmModerationClient.ts): LLM API integration
- **Service** (aiAnalysisWorker.ts): Worker pool management
- **Service** (indonesianTextNormalizer.ts): Text preprocessing
- **Service** (moderationPrompt.ts): Prompt generation
- **Events Published**:
- `discord:message:analyzed`
- `discord:analysis:queue_status`
LLM-only moderation. Entry: `aiAnalyzer.ts` (`queueMessageAnalysis`,
`startPendingAIAnalysisWorker`, `getAnalysisQueueStatus`). Scheduling:
`batchScheduler.ts` → `batchProcessor.ts` (batch lock + circuit breaker) →
`individualFallbackProcessor.ts` (per-message retry). Heavy work runs in the
Piscina pool via `ai-analysis-worker.ts` (jobs `batch` / `individual`).
Orchestration/caching: `moderationOrchestrator.ts` (exact hash → batched
semantic Qdrant → LLM), `textBatchProcessor.ts` / `mediaBatchProcessor.ts`
(one LLM call per sub-batch), `llmClient.ts` (central streaming client),
`embeddingClient.ts` + `qdrantClient.ts` (semantic cache), plus
`channelCultureStore.ts` / `userProfileStore.ts` / `userReputationStore.ts`.
### voice-recording
**Purpose**: Record voice channel audio
**Pattern**: Controller-Service-SubServices
- **Controller** (voiceController.ts): Voice connection management
- **Service** (recorder.ts): Recording orchestration
- **Sub-services** (recorder/*): Audio processing pipeline
- audioStream.ts: Opus packet subscription
- decoder.ts: Opus to PCM decoding
- segment.ts: OGG file rotation
- metadata.ts: User metadata collection
- sessionRecording.ts: Session lifecycle
- uploader.ts: Segment upload
- **Events Published**:
- `discord:voice:started`
- `discord:voice:stopped`
- `discord:voice:uploaded`
`voiceController.ts` (connect/disconnect/list) + `recorder.ts` (orchestration)
+ `recorder/` (decoder, segment, session, uploader, oggCrc). Publishes
`discord:voice:*` events. Real-time audio also streamed via `voice-pcm-ws`.
### attachment-upload
**Purpose**: Upload message attachments to external storage
**Pattern**: Service-Service
- **Service** (attachmentUploader.ts): Upload orchestration
- **Service** (imageResizer.ts): Image processing
- **Events Published**:
- `discord:attachment:created`
- `discord:attachment:uploaded`
`attachmentUploader.ts` (download → upload to storage) + `imageResizer.ts`
(sharp resize). Emits `discord:attachment:*`.
### event-broadcaster
**Purpose**: Publish events to Redis pub/sub
**Pattern**: Service-Domain
- **Service** (eventBroadcaster.ts): Redis publisher
- **Domain** (eventTypes.ts): Event type definitions
- **Channels**:
- discord:message:* (message events)
- discord:attachment:* (attachment events)
- discord:voice:* (voice events)
- discord:analysis:* (analysis events)
`RedisEventPublisher` (ioredis publish) + `EventBroadcaster` (typed methods).
Channel names in `src/shared/redis-channels.ts`.
## Shared Infrastructure
### gateway-metrics
`metrics.ts` Prometheus HTTP server on `METRICS_PORT` (4016). Collectors run
per scrape; live pipeline gauges registered in `bootstrap.ts`.
### config
- Zod-validated environment variables
- Type-safe configuration access
- Sensible defaults
## Shared infrastructure
- **config** — Zod schema in `shared/config/index.ts` (single source of truth).
- **database** — Drizzle ORM over `pg`; pool `min:0` (`shared/config`).
- **logger** — `pino` wrapper, `createChildLogger()` for context loggers.
- **errors** — `AppError` hierarchy (`ConfigError`, `AudioError`, …).
### database
- Drizzle ORM schema
- PostgreSQL connection
- Migration management
- Voice recording repository
### logger
- Winston logger wrapper
- Context-aware logging
- Log serialization utilities
### errors
- Custom error classes
- Error codes and HTTP status codes
- Proper error hierarchy
### utils
- Retry with exponential backoff
- Configurable retry parameters
### discord
- Discord.js client configuration
- Cache optimization
- Partial handling
## Event Flow
```
Discord Events
↓
message-capture (Controller)
↓
messageStore (Repository) → PostgreSQL
↓
eventBroadcaster (Service)
↓
Redis Pub/Sub
↓
Backend Service (Subscriber)
↓
HTTP API / WebSocket
↓
Frontend Application
```
## No HTTP Server
- ✅ No Express
- ✅ No WebSocket server
- ✅ No HTTP routes
- ✅ No middleware
- ✅ Pure event-driven service
## Graceful Shutdown
1. Close PostgreSQL connection
2. Disconnect from voice channels
3. Close Redis connection
4. Destroy Discord client
5. Exit process
## Dependencies
**Discord**:
- discord.js-selfbot-v13
- @discordjs/voice
- @discordjs/opus
**Audio**:
- prism-media
- opusscript
- sharp
**Data**:
- drizzle-orm
- pg
- zod
- ioredis
**Logging**:
- winston
- p-retry
- p-limit
- piscina
## Summary
The Discord Gateway service is a **pure event-driven microservice** that:
- Captures Discord messages, voice, and attachments
- Performs AI moderation analysis
- Publishes events to Redis pub/sub
- Has no HTTP server or WebSocket
- Follows Modular MVC pattern
- Maintains clean module boundaries
- Provides type-safe configuration
- Includes structured logging
- Handles graceful shutdown
The service is designed to run alongside the Backend service, which consumes Redis events and serves the HTTP API to the Frontend.
## Notes
- No HTTP server (other than the metrics endpoint). Pure event-driven.
- `MODULE_STRUCTURE.md` is intentionally a sketch; `ARCHITECTURE.md` is the
detailed reference. When they diverge, `ARCHITECTURE.md` wins.
@@ -0,0 +1,9 @@
CREATE TABLE IF NOT EXISTS "term_glossary_cache" (
"term" text PRIMARY KEY NOT NULL,
"definition" text NOT NULL,
"source_url" text DEFAULT '' NOT NULL,
"resolved_at" bigint NOT NULL,
"hit_count" integer DEFAULT 0 NOT NULL
);
--> statement-breakpoint
CREATE INDEX IF NOT EXISTS "idx_term_glossary_cache_resolved_at" ON "term_glossary_cache" USING btree ("resolved_at");
@@ -99,6 +99,13 @@
"when": 1785551832190,
"tag": "0013_rename_mascot_chat_to_chatbot",
"breakpoints": true
},
{
"idx": 14,
"version": "7",
"when": 1785621600000,
"tag": "0014_add_term_glossary_cache",
"breakpoints": true
}
]
}
@@ -1,3 +0,0 @@
node_modules/
build/
package-lock.json
@@ -1,526 +0,0 @@
// libdatachannel-min — minimal N-API binding to libdatachannel.
// Exposes ONLY what GMW GoLive needs:
// PeerConnection (offer/answer, ICE, SDP), DataChannel (signaling),
// Track send (added in media phase).
// Built against libdatachannel 0.24.0 (built from source in /tmp/ldc-build).
#include <napi.h>
#include <rtc/rtc.hpp>
#include <functional>
#include <memory>
#include <string>
#include <variant>
using namespace Napi;
namespace {
std::string stateToString(rtc::PeerConnection::State s) {
switch (s) {
case rtc::PeerConnection::State::New: return "new";
case rtc::PeerConnection::State::Connecting: return "connecting";
case rtc::PeerConnection::State::Connected: return "connected";
case rtc::PeerConnection::State::Disconnected: return "disconnected";
case rtc::PeerConnection::State::Failed: return "failed";
case rtc::PeerConnection::State::Closed: return "closed";
default: return "unknown";
}
}
std::string binaryToString(const rtc::binary& data) {
// rtc::binary is std::vector<std::byte> in libdatachannel >= 0.21
std::string msg(data.size(), '\0');
for (size_t i = 0; i < data.size(); i++) {
msg[i] = static_cast<char>(data[i]);
}
return msg;
}
// Holds a Napi::Promise::Deferred so it can be moved into TSFN lambdas
// without invalid copies (node-addon-api 8.x Deferred is not movable).
struct DeferredHolder {
Promise::Deferred deferred;
explicit DeferredHolder(Promise::Deferred d) : deferred(d) {}
};
class DataChannelWrap : public Napi::ObjectWrap<DataChannelWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "DataChannel", {
InstanceMethod("send", &DataChannelWrap::Send),
InstanceMethod("isOpen", &DataChannelWrap::IsOpen),
InstanceMethod("close", &DataChannelWrap::Close),
InstanceMethod("onMessage", &DataChannelWrap::OnMessage),
InstanceMethod("onOpen", &DataChannelWrap::OnOpen),
});
dcConstructor = Napi::Persistent(func);
return func;
}
// Create a JS wrapper (calls the JS constructor, returns instance).
static Object NewInstance(Napi::Env env) {
return dcConstructor.New({});
}
DataChannelWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<DataChannelWrap>(info) {}
void Init(std::shared_ptr<rtc::DataChannel> dc) {
dc_ = dc;
dc_->onMessage([this](rtc::message_variant data) {
std::string msg;
if (std::holds_alternative<rtc::binary>(data)) {
msg = binaryToString(std::get<rtc::binary>(data));
} else {
msg = std::get<std::string>(data);
}
if (msgCb_) {
msgCb_->BlockingCall([msg](Napi::Env env, Function cb) {
cb.Call({String::New(env, msg)});
});
}
});
dc_->onOpen([this]() {
if (openCb_) {
openCb_->BlockingCall([](Napi::Env env, Function cb) {
cb.Call({});
});
}
});
}
private:
static FunctionReference dcConstructor;
std::shared_ptr<rtc::DataChannel> dc_;
std::shared_ptr<ThreadSafeFunction> msgCb_;
std::shared_ptr<ThreadSafeFunction> openCb_;
void Send(const Napi::CallbackInfo& info) {
std::string msg = info[0].As<String>().Utf8Value();
if (dc_) dc_->send(msg);
}
Napi::Value IsOpen(const Napi::CallbackInfo& info) {
bool open = dc_ && dc_->isOpen();
return Boolean::New(info.Env(), open);
}
void Close(const Napi::CallbackInfo& info) {
if (dc_) dc_->close();
}
void OnMessage(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
msgCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc-message", 0, 1));
}
void OnOpen(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
openCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc-open", 0, 1));
}
};
class TrackWrap : public Napi::ObjectWrap<TrackWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "Track", {
InstanceMethod("send", &TrackWrap::Send),
InstanceMethod("isOpen", &TrackWrap::IsOpen),
InstanceMethod("close", &TrackWrap::Close),
InstanceMethod("setPacketizer", &TrackWrap::SetPacketizer),
InstanceMethod("sendFrame", &TrackWrap::SendFrame),
InstanceMethod("addTimestamp", &TrackWrap::AddTimestamp),
});
trackConstructor = Napi::Persistent(func);
return func;
}
static Object NewInstance(Napi::Env env) {
return trackConstructor.New({});
}
TrackWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<TrackWrap>(info) {}
void Init(std::shared_ptr<rtc::Track> track, Napi::Env env) {
track_ = track;
(void)env;
}
private:
static FunctionReference trackConstructor;
std::shared_ptr<rtc::Track> track_;
std::shared_ptr<rtc::RtpPacketizationConfig> rtpConfig_;
void Send(const Napi::CallbackInfo& info) {
Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
if (!track_) return;
rtc::binary data(buf.Length());
for (size_t i = 0; i < buf.Length(); i++) data[i] = (std::byte)buf[i];
try {
track_->send(data);
} catch (const std::exception& e) {
fprintf(stderr, "[binding] track.send THREW: %s\n", e.what());
}
}
// setPacketizer(kind, ssrc, payloadType, clockRate, playoutDelayId,
// playoutDelayMin, playoutDelayMax)
// kind: "audio" | "h264" | "h265" | "av1"
// Builds the media-handler chain (packetizer → RTCP SR → NACK → pacing for
// video) exactly like @dank074's WebRtcWrapper does via node-datachannel.
void SetPacketizer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
if (!track_) throw Error::New(env, "track closed");
std::string kind = info[0].As<String>().Utf8Value();
uint32_t ssrc = info[1].As<Number>().Uint32Value();
uint8_t pt = (uint8_t)info[2].As<Number>().Uint32Value();
uint32_t clockRate = info[3].As<Number>().Uint32Value();
uint8_t playoutDelayId = (uint8_t)info[4].As<Number>().Uint32Value();
uint16_t playoutDelayMin = (uint16_t)info[5].As<Number>().Uint32Value();
uint16_t playoutDelayMax = (uint16_t)info[6].As<Number>().Uint32Value();
try {
auto cfg = std::make_shared<rtc::RtpPacketizationConfig>(
ssrc, "", pt, clockRate);
cfg->playoutDelayId = playoutDelayId;
cfg->playoutDelayMin = playoutDelayMin;
cfg->playoutDelayMax = playoutDelayMax;
std::shared_ptr<rtc::MediaHandler> handler;
if (kind == "audio") {
handler = std::make_shared<rtc::OpusRtpPacketizer>(cfg);
} else if (kind == "h264") {
handler = std::make_shared<rtc::H264RtpPacketizer>(
rtc::NalUnit::Separator::StartSequence, cfg);
} else if (kind == "h265") {
handler = std::make_shared<rtc::H265RtpPacketizer>(
rtc::NalUnit::Separator::StartSequence, cfg);
} else if (kind == "av1") {
handler = std::make_shared<rtc::AV1RtpPacketizer>(
rtc::AV1RtpPacketizer::Packetization::Obu, cfg);
} else {
throw std::runtime_error("unknown packetizer kind: " + kind);
}
handler->addToChain(std::make_shared<rtc::RtcpSrReporter>(cfg));
handler->addToChain(std::make_shared<rtc::RtcpNackResponder>());
if (kind != "audio") {
handler->addToChain(std::make_shared<rtc::PacingHandler>(
25.0 * 1000 * 1000, std::chrono::milliseconds(1)));
}
track_->setMediaHandler(handler);
rtpConfig_ = cfg;
} catch (const std::exception& e) {
fprintf(stderr, "[binding] setPacketizer THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
}
// sendFrame(buffer) — sends an ENCODED frame (AnnexB H264 / raw opus /
// OBU AV1). The media-handler chain packetizes it into RTP.
void SendFrame(const Napi::CallbackInfo& info) {
Buffer<uint8_t> buf = info[0].As<Buffer<uint8_t>>();
if (!track_) return;
rtc::binary data(buf.Length());
for (size_t i = 0; i < buf.Length(); i++) data[i] = (std::byte)buf[i];
try {
track_->send(data);
} catch (const std::exception& e) {
fprintf(stderr, "[binding] track.sendFrame THREW: %s\n", e.what());
}
}
// addTimestamp(delta) — advances the packetizer RTP timestamp by delta
// (clock-rate units). Called by JS after each frame, matching the
// node-datachannel contract (WebRtcWrapper does the same increment).
void AddTimestamp(const Napi::CallbackInfo& info) {
uint32_t delta = info[0].As<Number>().Uint32Value();
if (rtpConfig_) rtpConfig_->timestamp += delta;
}
Napi::Value IsOpen(const Napi::CallbackInfo& info) {
bool open = track_ && track_->isOpen();
return Boolean::New(info.Env(), open);
}
void Close(const Napi::CallbackInfo& info) {
if (track_) track_->close();
}
void OnStateChange(const Napi::CallbackInfo& info) {
// libdatachannel Track has no state-change callback; kept for API parity.
(void)info;
}
};
class PeerConnectionWrap : public Napi::ObjectWrap<PeerConnectionWrap> {
public:
static Function Init(Napi::Env env) {
Function func = DefineClass(env, "PeerConnection", {
InstanceMethod("state", &PeerConnectionWrap::State),
InstanceMethod("createOffer", &PeerConnectionWrap::CreateOffer),
InstanceMethod("createAnswer", &PeerConnectionWrap::CreateAnswer),
InstanceMethod("setRemoteDescription",
&PeerConnectionWrap::SetRemoteDescription),
InstanceMethod("close", &PeerConnectionWrap::Close),
InstanceMethod("onStateChange", &PeerConnectionWrap::OnStateChange),
InstanceMethod("createDataChannel", &PeerConnectionWrap::CreateDataChannel),
InstanceMethod("onDataChannel", &PeerConnectionWrap::OnDataChannel),
InstanceMethod("addTrack", &PeerConnectionWrap::AddTrack),
});
return func;
}
PeerConnectionWrap(const Napi::CallbackInfo& info)
: Napi::ObjectWrap<PeerConnectionWrap>(info) {
Napi::Env env = info.Env();
if (!info[0].IsObject()) {
throw TypeError::New(env, "config object required");
}
Object config = info[0].As<Object>();
rtc::Configuration rtcConfig;
if (config.Has("iceServers")) {
Array servers = config.Get("iceServers").As<Array>();
for (uint32_t i = 0; i < servers.Length(); i++) {
std::string url = servers.Get(i).As<String>().Utf8Value();
rtcConfig.iceServers.emplace_back(url);
}
}
pc_ = std::make_shared<rtc::PeerConnection>(rtcConfig);
// IMPORTANT: register description/gathering callbacks HERE (constructor),
// BEFORE any createDataChannel call. libdatachannel only fires
// onLocalDescription for negotiations that start AFTER the callback is
// registered — if createDataChannel runs first, the offer callback never
// fires (verified in C++ spike: test3 vs test2).
pc_->onLocalDescription([this](rtc::Description desc) {
latestLocalDesc_ = std::string(desc);
fprintf(stderr, "[binding] trickle desc, %zu bytes\n",
latestLocalDesc_.size());
});
pc_->onGatheringStateChange([this](rtc::PeerConnection::GatheringState gs) {
fprintf(stderr, "[binding] gathering state: %d\n", (int)gs);
if (gs == rtc::PeerConnection::GatheringState::Complete) {
// Use the getter — it returns the FULL SDP including candidates after
// gathering (trickle callbacks only carry the initial fragment).
auto ld = pc_->localDescription();
if (ld) {
latestLocalDesc_ = std::string(*ld);
fprintf(stderr, "[binding] final desc, %zu bytes\n",
latestLocalDesc_.size());
}
resolvePendingLocalDesc_();
}
});
}
private:
std::shared_ptr<rtc::PeerConnection> pc_;
std::shared_ptr<ThreadSafeFunction> stateCb_;
std::shared_ptr<ThreadSafeFunction> dcCb_;
std::string latestLocalDesc_;
std::shared_ptr<DeferredHolder> pendingDescDeferred_;
std::shared_ptr<ThreadSafeFunction> pendingDescTsfn_;
void resolvePendingLocalDesc_() {
if (!pendingDescDeferred_ || !pendingDescTsfn_) return;
auto holder = pendingDescDeferred_;
auto tsfn = pendingDescTsfn_;
pendingDescDeferred_.reset();
pendingDescTsfn_.reset();
std::string sdp = latestLocalDesc_;
tsfn->BlockingCall([sdp, holder](Napi::Env e, Function) {
holder->deferred.Resolve(String::New(e, sdp));
});
}
Napi::Value State(const Napi::CallbackInfo& info) {
return String::New(info.Env(),
pc_ ? stateToString(pc_->state()) : "closed");
}
// createOffer() -> Promise<string> — sets local description, waits for
// ICE gathering to complete (so candidates are in the SDP), resolves SDP.
Napi::Value CreateOffer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
auto holder = std::make_shared<DeferredHolder>(Promise::Deferred::New(env));
if (!pc_) {
holder->deferred.Reject(Error::New(env, "peer closed").Value());
return holder->deferred.Promise();
}
// createDataChannel already triggers negotiation in libdatachannel 0.24 —
// if gathering already completed, resolve immediately from the cached SDP.
if (!latestLocalDesc_.empty()) {
auto tsfn = std::make_shared<ThreadSafeFunction>(ThreadSafeFunction::New(
env, Function::New(env, [](const CallbackInfo&) {}), "desc", 0, 1));
std::string sdp = latestLocalDesc_;
tsfn->BlockingCall([sdp, holder](Napi::Env e, Function) {
holder->deferred.Resolve(String::New(e, sdp));
});
return holder->deferred.Promise();
}
if (pendingDescDeferred_) {
pendingDescDeferred_->deferred.Reject(
Error::New(env, "previous negotiation still pending").Value());
}
pendingDescDeferred_ = holder;
pendingDescTsfn_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(env, Function::New(env, [](const CallbackInfo&) {}),
"desc", 0, 1));
fprintf(stderr, "[binding] calling setLocalDescription(Offer)\n");
try {
pc_->setLocalDescription(rtc::Description::Type::Offer);
fprintf(stderr, "[binding] setLocalDescription returned OK\n");
} catch (const std::exception& e) {
pendingDescDeferred_.reset();
fprintf(stderr, "[binding] setLocalDescription THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
return holder->deferred.Promise();
}
// createAnswer(offerSdp: string) -> Promise<string>
Napi::Value CreateAnswer(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string offer = info[0].As<String>().Utf8Value();
auto holder = std::make_shared<DeferredHolder>(Promise::Deferred::New(env));
if (!pc_) {
holder->deferred.Reject(Error::New(env, "peer closed").Value());
return holder->deferred.Promise();
}
if (pendingDescDeferred_) {
pendingDescDeferred_->deferred.Reject(
Error::New(env, "previous negotiation still pending").Value());
}
pendingDescDeferred_ = holder;
pendingDescTsfn_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(env, Function::New(env, [](const CallbackInfo&) {}),
"desc", 0, 1));
try {
pc_->setRemoteDescription(
rtc::Description(offer, rtc::Description::Type::Offer));
fprintf(stderr, "[binding] answer: setRemoteDescription OK\n");
// libdatachannel 0.24 AUTO-GENERATES the answer when a remote offer is
// applied (verified in C++ spike test8/9: B desc type=Answer fires
// immediately with a=setup:active). Calling setLocalDescription() again
// would OVERWRITE it with a role=actpass SDP, which A rejects with
// "Illegal role actpass in remote answer description". So we do NOT call
// setLocalDescription here — we just wait for gathering complete and
// resolve with the auto-generated answer. This also matches @dank074's
// Discord voice flow.
} catch (const std::exception& e) {
pendingDescDeferred_.reset();
fprintf(stderr, "[binding] answer THREW: %s\n", e.what());
holder->deferred.Reject(Error::New(env, e.what()).Value());
}
return holder->deferred.Promise();
}
void SetRemoteDescription(const Napi::CallbackInfo& info) {
std::string sdp = info[0].As<String>().Utf8Value();
std::string type = info[1].As<String>().Utf8Value();
rtc::Description::Type t = (type == "answer")
? rtc::Description::Type::Answer
: rtc::Description::Type::Offer;
if (pc_) pc_->setRemoteDescription(rtc::Description(sdp, t));
}
void Close(const Napi::CallbackInfo& info) {
if (pc_) pc_->close();
}
void OnStateChange(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
stateCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "pc-state", 0, 1));
std::shared_ptr<rtc::PeerConnection> pc = pc_;
pc->onStateChange([this](rtc::PeerConnection::State state) {
if (stateCb_) {
std::string s = stateToString(state);
stateCb_->BlockingCall([s](Napi::Env env, Function cb) {
cb.Call({String::New(env, s)});
});
}
});
}
Napi::Value CreateDataChannel(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string label = info[0].As<String>().Utf8Value();
fprintf(stderr, "[binding] createDataChannel(%s)\n", label.c_str());
auto dc = pc_->createDataChannel(label);
Object obj = DataChannelWrap::NewInstance(env);
DataChannelWrap::Unwrap(obj)->Init(dc);
return obj;
}
Napi::Value AddTrack(const Napi::CallbackInfo& info) {
Napi::Env env = info.Env();
std::string mid = info[0].As<String>().Utf8Value();
std::string kind = info[1].As<String>().Utf8Value();
if (!pc_) throw Error::New(env, "peer closed");
fprintf(stderr, "[binding] addTrack(%s, %s) start\n", mid.c_str(), kind.c_str());
try {
std::shared_ptr<rtc::Track> track;
if (kind == "audio") {
// Opus payload type 120 (matches @dank074 CodecPayloadType.opus)
auto desc = rtc::Description::Audio(mid);
desc.addOpusCodec(120);
track = pc_->addTrack(desc);
} else {
// All video codecs with their payload types, matching WebRtcWrapper:
// H264 101/102, H265 103/104, VP8 105/106, VP9 107/108, AV1 109/110
auto desc = rtc::Description::Video(mid);
desc.addH264Codec(101);
desc.addRtxCodec(102, 101, 90000);
desc.addH265Codec(103);
desc.addRtxCodec(104, 103, 90000);
desc.addVP8Codec(105);
desc.addRtxCodec(106, 105, 90000);
desc.addVP9Codec(107);
desc.addRtxCodec(108, 107, 90000);
desc.addAV1Codec(109);
desc.addRtxCodec(110, 109, 90000);
track = pc_->addTrack(desc);
}
Object obj = TrackWrap::NewInstance(env);
TrackWrap::Unwrap(obj)->Init(track, env);
return obj;
} catch (const std::exception& e) {
fprintf(stderr, "[binding] addTrack THREW: %s\n", e.what());
throw Error::New(env, e.what());
}
}
void OnDataChannel(const Napi::CallbackInfo& info) {
Function cb = info[0].As<Function>();
dcCb_ = std::make_shared<ThreadSafeFunction>(
ThreadSafeFunction::New(info.Env(), cb, "dc", 0, 1));
std::shared_ptr<rtc::PeerConnection> pc = pc_;
pc->onDataChannel([this](std::shared_ptr<rtc::DataChannel> dc) {
if (dcCb_) {
auto dcPtr = dc;
dcCb_->BlockingCall([dcPtr](Napi::Env env, Function cb) {
Object obj = DataChannelWrap::NewInstance(env);
DataChannelWrap::Unwrap(obj)->Init(dcPtr);
cb.Call({obj});
});
}
});
}
};
Object InitAll(Napi::Env env, Object exports) {
exports.Set("PeerConnection", PeerConnectionWrap::Init(env));
exports.Set("DataChannel", DataChannelWrap::Init(env));
exports.Set("Track", TrackWrap::Init(env));
return exports;
}
NODE_API_MODULE(libdatachannel_min, InitAll)
// Definition for the static constructor references.
FunctionReference DataChannelWrap::dcConstructor;
FunctionReference TrackWrap::trackConstructor;
} // namespace
@@ -1,21 +0,0 @@
{
"targets": [
{
"target_name": "libdatachannel_min",
"sources": ["binding.cpp"],
"include_dirs": [
"<!(node -e \"console.log(process.env.NAPI_INCLUDE || (() => { try { return require('node-addon-api').include; } catch { return '/nonexistent'; } })())\")",
"<!(node -e \"const s=process.env.LDC_INCLUDE||'/nix/store/39a85gpfjqy3h3k8jwrwh7m9yc3inqw7-source';console.log(s+'/include')\")"
],
"libraries": [
"<!(node -e \"console.log(process.env.LDC_LIB || '/tmp/ldc-build/libdatachannel.so.0.24.0')\")"
],
"cflags": ["-std=c++17", "-fexceptions"],
"cflags_cc": ["-std=c++17", "-fexceptions"],
"defines": ["NAPI_CPP_EXCEPTIONS"],
"conditions": [
["OS=='linux'", { "cflags": ["-fvisibility=hidden"] }]
]
}
]
}
@@ -1,3 +0,0 @@
// libdatachannel-min — JS entry.
const native = require("./build/Release/datachannel_min.node");
module.exports = native;
@@ -1,17 +0,0 @@
{
"name": "libdatachannel-min",
"version": "0.1.0",
"description": "Minimal N-API binding to libdatachannel — PeerConnection, DataChannel, ICE, SDP (+ media tracks for GoLive)",
"main": "index.js",
"gypfile": true,
"scripts": {
"build": "node-gyp rebuild",
"test": "node test-handshake.js"
},
"dependencies": {
"node-addon-api": "^8.3.0"
},
"devDependencies": {
"node-gyp": "^11.5.0"
}
}
@@ -1,82 +0,0 @@
// Phase 0 spike: prove the minimal binding can do a full WebRTC handshake
// (offer/answer + ICE + DataChannel) between two local PeerConnections.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
function log(...args) {
console.log("[spike]", ...args);
}
async function main() {
const pcA = new PeerConnection({ iceServers: [] });
const pcB = new PeerConnection({ iceServers: [] });
const stateLog = [];
pcA.onStateChange((s) => {
stateLog.push(`A:${s}`);
log("A state:", s);
});
pcB.onStateChange((s) => {
stateLog.push(`B:${s}`);
log("B state:", s);
});
// B waits for incoming DataChannel
const received = new Promise((resolve) => {
pcB.onDataChannel((dc) => {
log("B got incoming DataChannel");
dc.onOpen(() => log("B DataChannel open"));
dc.onMessage((msg) => {
log("B received message:", msg);
dc.send("pong from B");
resolve(msg);
});
});
});
// A creates an outgoing DataChannel
const dcA = pcA.createDataChannel("test");
dcA.onOpen(() => {
log("A DataChannel open — sending hello");
dcA.send("hello from A");
});
dcA.onMessage((msg) => {
log("A received reply:", msg);
});
// Offer/answer dance
log("A createOffer...");
const offer = await pcA.createOffer();
log("Offer SDP bytes:", offer.length);
log("B createAnswer...");
const answer = await pcB.createAnswer(offer);
log("Answer SDP bytes:", answer.length);
const setupMatch = answer.match(/a=setup:(\S+)/);
log("Answer setup role:", setupMatch ? setupMatch[1] : "NONE");
pcA.setRemoteDescription(answer, "answer");
// Wait for message roundtrip
const msg = await Promise.race([
received,
new Promise((_, rej) => setTimeout(() => rej(new Error("TIMEOUT waiting for datachannel message")), 15000)),
]);
log("ROUNDTRIP OK — B got:", msg);
log("States:", stateLog.join(" | "));
const aState = pcA.state();
const bState = pcB.state();
log("Final states — A:", aState, "B:", bState);
pcA.close();
pcB.close();
if (msg !== "hello from A") throw new Error("wrong message");
if (aState !== "connected" && aState !== "disconnected") throw new Error("A not connected: " + aState);
log("SPIKE PASSED ✅");
}
main().catch((e) => {
console.error("SPIKE FAILED:", e.message);
process.exit(1);
});
@@ -1,80 +0,0 @@
// Verify setPacketizer + sendFrame: two peers connect, audio+video tracks
// packetize real encoded frames (opus + AnnexB H264), RTP flows without crash.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
function sleep(ms) { return new Promise((r) => setTimeout(r, ms)); }
async function main() {
const pcA = new PeerConnection({ iceServers: [] });
const pcB = new PeerConnection({ iceServers: [] });
const aAudio = pcA.addTrack("0", "audio");
const aVideo = pcA.addTrack("1", "video");
pcB.addTrack("0", "audio");
pcB.addTrack("1", "video");
let states = { a: "", b: "" };
pcA.onStateChange((s) => (states.a = s));
pcB.onStateChange((s) => (states.b = s));
// A: offer (createDataChannel not needed — tracks trigger negotiation)
const offer = await pcA.createOffer();
pcB.setRemoteDescription(offer, "offer");
const answer = await pcB.createAnswer(offer);
pcA.setRemoteDescription(answer, "answer");
// Wait for connected
for (let i = 0; i < 50; i++) {
if (states.a === "connected" && states.b === "connected") break;
await sleep(100);
}
console.log("[pkt] states:", states.a, states.b);
if (states.a !== "connected" || states.b !== "connected") {
console.log("PKT TEST FAILED: not connected");
process.exit(1);
}
// Setup packetizers on A (sender)
aAudio.setPacketizer("audio", 1234, 120, 48000, 5, 0, 1);
aVideo.setPacketizer("h264", 5678, 101, 90000, 5, 0, 10);
// Fake opus frame (20ms @48kHz stereo — payload can be any bytes)
const opusFrame = Buffer.alloc(160);
for (let i = 0; i < 160; i++) opusFrame[i] = i & 0xff;
// Fake AnnexB H264 frame: SPS + PPS + IDR slice
const sps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x67, 0x42, 0xc0, 0x1e, 0xd9, 0x01, 0x40, 0x7e]);
const pps = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x68, 0xce, 0x3c, 0x80]);
const idr = Buffer.from([0x00, 0x00, 0x00, 0x01, 0x65, 0x88, 0x84, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07]);
const h264Frame = Buffer.concat([sps, pps, idr]);
// Send 10 audio frames (20ms each) + 3 video frames (33ms each)
for (let i = 0; i < 10; i++) {
aAudio.sendFrame(opusFrame);
aAudio.addTimestamp(960); // 20ms @ 48kHz
}
for (let i = 0; i < 3; i++) {
aVideo.sendFrame(h264Frame);
aVideo.addTimestamp(3000); // 33ms @ 90kHz
}
await sleep(500);
console.log("[pkt] after send: states:", states.a, states.b);
console.log("[pkt] audio track open:", aAudio.isOpen(), "| video track open:", aVideo.isOpen());
const ok = states.a === "connected" && aAudio.isOpen() && aVideo.isOpen();
console.log(ok ? "PKT TEST PASSED" : "PKT TEST FAILED");
pcA.close();
pcB.close();
process.exit(ok ? 0 : 1);
}
main().catch((e) => {
console.error("[pkt] FAILED:", e.message);
process.exit(1);
});
setTimeout(() => {
console.error("[pkt] TIMEOUT");
process.exit(1);
}, 25000);
@@ -1,33 +0,0 @@
// Verify addTrack produces SDP with audio+video media sections.
"use strict";
const { PeerConnection } = require("./build/Release/datachannel_min.node");
const pc = new PeerConnection({ iceServers: [] });
const audioTrack = pc.addTrack("0", "audio");
const videoTrack = pc.addTrack("1", "video");
pc.onStateChange((s) => console.log("[test-track] state:", s));
pc.createOffer().then((sdp) => {
const hasAudio = /^m=audio\s/m.test(sdp);
const hasVideo = /^m=video\s/m.test(sdp);
const audioPts = sdp.match(/a=rtpmap:(\d+) opus/g) || [];
const videoPts = sdp.match(/a=rtpmap:(\d+) H264/g) || [];
console.log("[test-track] SDP bytes:", sdp.length);
console.log("[test-track] m=audio:", hasAudio, "| m=video:", hasVideo);
console.log("[test-track] opus pt:", audioPts, "| H264 pt:", videoPts);
console.log("[test-track] audio track send ok:", typeof audioTrack.send === "function");
console.log("[test-track] video track send ok:", typeof videoTrack.send === "function");
const ok = hasAudio && hasVideo && audioPts.length > 0 && videoPts.length > 0;
console.log(ok ? "TRACK TEST PASSED" : "TRACK TEST FAILED");
pc.close();
process.exit(ok ? 0 : 1);
}).catch((e) => {
console.error("[test-track] FAILED:", e.message);
process.exit(1);
});
setTimeout(() => {
console.error("[test-track] TIMEOUT");
process.exit(1);
}, 20000);
-1
View File
@@ -28,7 +28,6 @@
"discord.js-selfbot-v13": "^3.7.1",
"dotenv": "^17.4.2",
"drizzle-orm": "^0.45.2",
"imghash": "^1.1.4",
"ioredis": "^5.11.0",
"libsodium-wrappers": "^0.8.4",
"lru-cache": "^11.5.1",
+100 -6
View File
@@ -3,7 +3,11 @@ import { inArray, lt } from "drizzle-orm";
import type { NodePgDatabase } from "drizzle-orm/node-postgres";
import { ConfigError, DatabaseError } from "@/shared/errors/index";
import { createChildLogger } from "@/shared/logger/index";
import { startPendingAIAnalysisWorker } from "../modules/ai-moderation/aiAnalyzer.js";
import {
getAnalysisQueueStatus,
startPendingAIAnalysisWorker,
} from "../modules/ai-moderation/aiAnalyzer.js";
import { workerPool } from "../modules/ai-moderation/circuitBreaker.js";
import { registerChannelTopicCapture } from "../modules/channel-topic/index.js";
import { CommandHandler } from "../modules/command-handler/commandHandler.js";
import {
@@ -11,6 +15,8 @@ import {
RedisEventPublisher,
} from "../modules/event-broadcaster/index.js";
import {
registerCollector,
setGauge,
startMetricsServer,
stopMetricsServer,
} from "../modules/gateway-metrics/index.js";
@@ -222,7 +228,10 @@ export async function initializeDiscordGateway() {
await initializeDatabase();
logger.info("PostgreSQL database initialized");
} catch (err) {
logger.error({ error: err }, "Failed to initialize database");
logger.error(
{ err, errorMsg: err instanceof Error ? err.message : String(err) },
"Failed to initialize database",
);
throw new DatabaseError(
`Database initialization failed: ${err instanceof Error ? err.message : String(err)}`,
);
@@ -267,7 +276,10 @@ export async function initializeDiscordGateway() {
});
client.on("error", (err) => {
logger.error({ error: err }, "Client error");
logger.error(
{ err, errorMsg: err instanceof Error ? err.message : String(err) },
"Client error",
);
});
process.on("SIGINT", () => {
@@ -279,15 +291,97 @@ export async function initializeDiscordGateway() {
});
process.on("uncaughtException", (err) => {
logger.error({ error: err }, "Uncaught exception");
const code =
typeof (err as NodeJS.ErrnoException).code === "string"
? (err as NodeJS.ErrnoException).code
: "";
// Transient stream-teardown errors (voice stop/disconnect races, child
// process stdin closed while we still write) are NOT fatal — crashing the
// gateway on EPIPE takes the whole bot offline mid-music. Log + continue.
if (
code === "EPIPE" ||
code === "ERR_STREAM_DESTROYED" ||
code === "ERR_STREAM_WRITE_AFTER_END" ||
code === "ECONNRESET"
) {
logger.warn(
{ error: err },
"Uncaught transient stream error — continuing",
);
return;
}
logger.error(
{
err,
errorMsg: err instanceof Error ? err.message : String(err),
stack: err?.stack,
},
"Uncaught exception",
);
gracefulShutdown("uncaughtException");
});
process.on("unhandledRejection", (reason, promise) => {
logger.error({ reason, promise }, "Unhandled rejection");
process.on("unhandledRejection", (reason) => {
const err =
reason instanceof Error ? reason : new Error(String(reason ?? "unknown"));
const code = (err as NodeJS.ErrnoException).code ?? "";
// Same transient-teardown policy as uncaughtException: a rejection that
// fires while a stream is being torn down (EPIPE after ffmpeg stdin
// closes, write-after-destroy, socket reset) must NOT take the whole
// gateway offline. Log detail + continue. Everything else still shuts
// down so real bugs surface.
if (
code === "EPIPE" ||
code === "ERR_STREAM_DESTROYED" ||
code === "ERR_STREAM_WRITE_AFTER_END" ||
code === "ECONNRESET"
) {
logger.warn(
{ error: err },
"Unhandled rejection transient stream error — continuing",
);
return;
}
logger.error({ error: err, reason: String(reason) }, "Unhandled rejection");
gracefulShutdown("unhandledRejection");
});
// ── Metrics: register live pipeline collectors before starting server ──
// These refresh on every scrape so Prometheus sees real AI-analysis
// queue depth, concurrency, and DB pool state instead of an empty stub.
registerCollector(() => {
if (!config.AI_ANALYSIS_ENABLED) return;
try {
const status = getAnalysisQueueStatus();
setGauge("ai_analysis_queued_conversations", status.queuedConversations);
setGauge("ai_analysis_active_batch_requests", status.activeRequests);
setGauge(
"ai_analysis_active_individual_requests",
status.activeIndividualRequests,
);
setGauge(
"ai_analysis_individual_in_flight",
status.individualInFlightCount,
);
setGauge(
"ai_analysis_individual_circuit_breaker_active",
status.individualCircuitBreakerActive ? 1 : 0,
);
if (typeof status.lastError === "string") {
setGauge("ai_analysis_last_error_present", status.lastError ? 1 : 0);
}
const pool = workerPool as unknown as {
_poolState?: { size: number; active: number };
};
if (pool._poolState) {
setGauge("ai_analysis_worker_threads", pool._poolState.size);
setGauge("ai_analysis_worker_threads_active", pool._poolState.active);
}
} catch (err) {
logger.warn({ error: String(err) }, "AI metrics collector failed");
}
});
// Start metrics server
startMetricsServer();
@@ -1,155 +0,0 @@
/**
* AnnexB bitstream reader/writer (RBSP + emulation prevention) — ported
* from @dank074/discord-video-stream AnnexBBitstreamReaderWriter.js.
*/
export class AnnexBBitstreamReader {
private _buffer: Uint8Array;
private _byteOffset = 0;
private _bitOffset = 0;
constructor(buffer: Uint8Array) {
this._buffer = buffer;
}
readBits(count: number): number {
if (count === 0) return 0;
let result = 0;
while (count > 0) {
if (this._byteOffset >= this._buffer.length) {
throw new Error("Bad byte offset");
}
if (
this._bitOffset === 0 &&
this._byteOffset >= 2 &&
this._buffer[this._byteOffset - 2] === 0 &&
this._buffer[this._byteOffset - 1] === 0 &&
this._buffer[this._byteOffset] === 3
) {
// Skip over emulation prevention
this._byteOffset++;
}
if (this._bitOffset === 0 && count >= 8) {
result = (result << 8) | this._buffer[this._byteOffset++];
count -= 8;
} else {
const numBitsToRead = Math.min(count, 8 - this._bitOffset);
const mask = (1 << numBitsToRead) - 1;
const newBits =
(this._buffer[this._byteOffset] >>
(8 - this._bitOffset - numBitsToRead)) &
mask;
result = (result << numBitsToRead) | newBits;
count -= numBitsToRead;
this._bitOffset += numBitsToRead;
if (this._bitOffset === 8) {
this._bitOffset = 0;
this._byteOffset++;
}
}
}
return result;
}
readUnsigned(bits: number): number {
return this.readBits(bits);
}
readSigned(bits: number): number {
const unsigned = this.readUnsigned(bits);
if (unsigned & (1 << (bits - 1))) return unsigned - (1 << bits);
return unsigned;
}
readUnsignedExpGolomb(): number {
let leading0 = 0;
while (this.readBits(1) === 0) leading0++;
return (1 << leading0) + this.readBits(leading0) - 1;
}
readSignedExpGolomb(): number {
const unsigned = this.readUnsignedExpGolomb();
if (unsigned % 2 === 0) return unsigned / -2;
return (unsigned + 1) / 2;
}
}
export class AnnexBBitstreamWriter {
private _arr: number[] = [];
private _pendingByte = 0;
private _bitOffset = 0;
toBuffer(): Buffer {
return Buffer.from(this._arr);
}
flush(): void {
// Emulation prevention: insert 0x03 before 00 00
if (
this._pendingByte <= 3 &&
this._arr[this._arr.length - 1] === 0 &&
this._arr[this._arr.length - 2] === 0
) {
this._arr.push(3);
}
this._arr.push(this._pendingByte);
this._pendingByte = 0;
this._bitOffset = 0;
}
writeBits(bits: number, count: number): void {
while (count > 0) {
if (this._bitOffset === 0) {
if (count >= 8) {
this._pendingByte = (bits >> (count - 8)) & 0xff;
count -= 8;
this.flush();
} else {
const mask = (1 << count) - 1;
this._pendingByte |= (bits & mask) << (8 - count);
this._bitOffset = count;
count = 0;
}
} else {
const numBitsToWrite = Math.min(8 - this._bitOffset, count);
const bitsToWrite =
(bits >> (count - numBitsToWrite)) & ((1 << numBitsToWrite) - 1);
this._pendingByte |=
bitsToWrite << (8 - this._bitOffset - numBitsToWrite);
count -= numBitsToWrite;
this._bitOffset += numBitsToWrite;
if (this._bitOffset === 8) {
this._bitOffset = 0;
this.flush();
}
}
}
}
writeUnsigned(num: number, count: number): void {
if (num < 0) throw new Error("Expected a non-negative number");
this.writeBits(num, count);
}
writeSigned(num: number, count: number): void {
if (count <= 0) return;
if (count > 32) throw new Error("writeSigned supports up to 32 bits");
const mask =
count === 32 ? 0xffffffff >>> 0 : (((1 << count) >>> 0) - 1) >>> 0;
const unsigned = (num & mask) >>> 0;
this.writeBits(unsigned, count);
}
writeUnsignedExpGolomb(num: number): void {
if (num < 0) throw new Error("Expected a non-negative number");
num++;
const bitCount = 32 - Math.clz32(num >>> 0);
this.writeBits(0, bitCount - 1);
this.writeBits(num, bitCount);
}
writeSignedExpGolomb(num: number): void {
if (num < 0) this.writeUnsignedExpGolomb(-2 * num);
else this.writeUnsignedExpGolomb(2 * num - 1);
}
}
@@ -1,112 +0,0 @@
/**
* H264/H265 NAL helpers — ported from @dank074/discord-video-stream
* AnnexBHelper.js. Only the H264 parts are used by GoLive (H264 encoder),
* H265 constants kept for completeness of the port.
*/
export enum H264NalUnitTypes {
Unspecified = 0,
CodedSliceNonIDR = 1,
CodedSlicePartitionA = 2,
CodedSlicePartitionB = 3,
CodedSlicePartitionC = 4,
CodedSliceIdr = 5,
SEI = 6,
SPS = 7,
PPS = 8,
AccessUnitDelimiter = 9,
EndOfSequence = 10,
EndOfStream = 11,
FillerData = 12,
SEIExtenstion = 13,
PrefixNalUnit = 14,
SubsetSPS = 15,
}
export enum H265NalUnitTypes {
TRAIL_N = 0,
TRAIL_R = 1,
TSA_N = 2,
TSA_R = 3,
STSA_N = 4,
STSA_R = 5,
RADL_N = 6,
RADL_R = 7,
RASL_N = 8,
RASL_R = 9,
RSV_VCL_N10 = 10,
RSV_VCL_R11 = 11,
RSV_VCL_N12 = 12,
RSV_VCL_R13 = 13,
RSV_VCL_N14 = 14,
RSV_VCL_R15 = 15,
BLA_W_LP = 16,
BLA_W_RADL = 17,
BLA_N_LP = 18,
IDR_W_RADL = 19,
IDR_N_LP = 20,
CRA_NUT = 21,
RSV_IRAP_VCL22 = 22,
RSV_IRAP_VCL23 = 23,
RSV_VCL24 = 24,
RSV_VCL25 = 25,
RSV_VCL26 = 26,
RSV_VCL27 = 27,
RSV_VCL28 = 28,
RSV_VCL29 = 29,
RSV_VCL30 = 30,
RSV_VCL31 = 31,
VPS_NUT = 32,
SPS_NUT = 33,
PPS_NUT = 34,
AUD_NUT = 35,
EOS_NUT = 36,
EOB_NUT = 37,
FD_NUT = 38,
PREFIX_SEI_NUT = 39,
SUFFIX_SEI_NUT = 40,
}
export const H264Helpers = {
getUnitType(frame: Uint8Array): number {
return frame[0] & 0x1f;
},
splitHeader(frame: Uint8Array): [Uint8Array, Uint8Array] {
return [frame.subarray(0, 1), frame.subarray(1)];
},
isAUD(unitType: number): boolean {
return unitType === H264NalUnitTypes.AccessUnitDelimiter;
},
};
export const H265Helpers = {
getUnitType(frame: Uint8Array): number {
return (frame[0] >> 1) & 0x3f;
},
splitHeader(frame: Uint8Array): [Uint8Array, Uint8Array] {
return [frame.subarray(0, 2), frame.subarray(2)];
},
isAUD(unitType: number): boolean {
return unitType === H265NalUnitTypes.AUD_NUT;
},
};
export const startCode3 = Buffer.from([0, 0, 1]);
/** Split an AnnexB bitstream into NAL units (start codes stripped). */
export function splitNalu(buf: Buffer): Buffer[] {
let temp: Buffer | null = buf;
const nalus: Buffer[] = [];
while (temp?.byteLength) {
let pos: number = temp.indexOf(startCode3);
let length = 3;
if (pos > 0 && temp[pos - 1] === 0) {
pos--;
length++;
}
const nalu = pos === -1 ? temp : temp.subarray(0, pos);
temp = pos === -1 ? null : temp.subarray(pos + length);
if (nalu.byteLength) nalus.push(nalu);
}
return nalus;
}
@@ -1,20 +0,0 @@
/**
* AudioStream — feeds encoded opus frames into the WebRTC connection.
* Ported from @dank074/discord-video-stream AudioStream.js.
*/
import { BaseMediaStream } from "./BaseMediaStream.js";
import type { WebRtcConnWrapper } from "./WebRtcWrapper.js";
export class AudioStream extends BaseMediaStream {
_conn: WebRtcConnWrapper;
constructor(conn: WebRtcConnWrapper, noSleep = false) {
super("audio", noSleep);
this._conn = conn;
}
async _sendFrame(frame: Buffer, frametime: number): Promise<void> {
this._conn.sendAudioFrame(frame, frametime);
}
}
@@ -1,594 +0,0 @@
/**
* Base media connection for Discord GoLive — ported from
* @dank074/discord-video-stream BaseMediaConnection.js.
*
* Owns the voice WebSocket (identify/select_protocol/heartbeat/resume),
* SDP negotiation against Discord's media server, DAVE E2E voice
* (via @snazzah/davey), and speaking/video attribute signaling.
*/
import { randomUUID } from "node:crypto";
import { EventEmitter } from "node:events";
import Davey from "@snazzah/davey";
import { CodecPayloadType } from "./CodecPayloadType.js";
import type { NativePeerConnection } from "./native.js";
import { isNativeAvailable } from "./native.js";
import { STREAMS_SIMULCAST } from "./utils.js";
import { VoiceOpCodes, VoiceOpCodesBinary } from "./VoiceOpCodes.js";
import { WebRtcConnWrapper } from "./WebRtcWrapper.js";
export interface MediaConnectionStatus {
hasSession: boolean;
hasToken: boolean;
started: boolean;
resuming: boolean;
}
export interface VideoAttribute {
fps: number;
width: number;
height: number;
}
export interface StreamerLike {
opts: Record<string, unknown>;
}
export class BaseMediaConnection extends EventEmitter {
interval: ReturnType<typeof setInterval> | null = null;
guildId: string | null = null;
channelId: string;
botId: string;
ws: WebSocket | null = null;
status: MediaConnectionStatus;
server: string | null = null; // websocket url
token: string | null = null;
session_id: string | null = null;
protected _webRtcWrapper: WebRtcConnWrapper;
_webRtcParams: {
address: string;
port: number;
audioSsrc: number;
videoSsrc: number;
rtxSsrc: number;
supportedEncryptionModes: string[];
} | null = null;
protected _closed = false;
ready: ((conn: WebRtcConnWrapper) => void) | null;
protected _streamer: StreamerLike;
protected _sequenceNumber = -1;
protected _daveSession: Davey.DAVESession | null = null;
protected _connectedUsers = new Set<string>();
protected _daveProtocolVersion = 0;
protected _davePendingTransitions = new Map<number, number>();
protected _daveDowngraded = false;
constructor(
streamer: StreamerLike,
guildId: string | null,
botId: string,
channelId: string,
callback: ((conn: WebRtcConnWrapper) => void) | null,
) {
super();
this._streamer = streamer;
this.status = {
hasSession: false,
hasToken: false,
started: false,
resuming: false,
};
this.guildId = guildId;
this.channelId = channelId;
this.botId = botId;
this.ready = callback;
this._webRtcWrapper = new WebRtcConnWrapper(this);
}
get type(): "guild" | "call" {
return this.guildId ? "guild" : "call";
}
get webRtcConn(): WebRtcConnWrapper {
return this._webRtcWrapper;
}
get webRtcParams(): BaseMediaConnection["_webRtcParams"] {
return this._webRtcParams;
}
get streamer(): StreamerLike {
return this._streamer;
}
/** daveChannelId — overridden in VoiceConnection (channelId) and StreamConnection (serverId - 1n). */
get daveChannelId(): string {
throw new Error("daveChannelId not implemented");
}
stop(): void {
this._closed = true;
this._webRtcWrapper.close();
this.ws?.close();
}
setSession(session_id: string): void {
this.session_id = session_id;
this.status.hasSession = true;
this.start();
}
setTokens(server: string, token: string): void {
this.token = token;
this.server = server;
this.status.hasToken = true;
this.start();
}
start(): void {
if (this.status.hasSession && this.status.hasToken) {
if (this.status.started) return;
this.status.started = true;
this.ws = new WebSocket(`wss://${this.server}/?v=8`);
this.ws.binaryType = "arraybuffer";
this.ws.addEventListener("open", () => {
if (this.status.resuming) {
this.status.resuming = false;
this.resume();
} else {
this.identify();
}
});
this.ws.addEventListener("error", (err) => {
console.error(err);
});
this.ws.addEventListener("close", (e) => {
const wasStarted = this.status.started;
this.interval && clearInterval(this.interval);
this.status.started = false;
const canResume = e.code === 4015 || e.code < 4000;
if (canResume && wasStarted) {
this.status.resuming = true;
this.start();
} else {
this._closed = true;
this._webRtcWrapper?.close();
}
});
this.setupEvents();
}
}
handleReady(d: {
ip: string;
port: number;
ssrc: number;
streams: { ssrc: number; rtx_ssrc: number }[];
modes: string[];
}): void {
// we hardcoded STREAMS_SIMULCAST, which will always be array of 1
const stream = d.streams[0];
this._webRtcParams = {
address: d.ip,
port: d.port,
audioSsrc: d.ssrc,
videoSsrc: stream.ssrc,
rtxSsrc: stream.rtx_ssrc,
supportedEncryptionModes: d.modes,
};
}
async handleProtocolAck(d: {
sdp?: string;
dave_protocol_version?: number;
}): Promise<void> {
if (!("sdp" in d)) throw new Error("Only WebRTC connections are allowed");
this._daveProtocolVersion = d.dave_protocol_version ?? 0;
this.initDave();
// Discord's SDP is garbage — generate our own from its pieces
let ip = "";
let port = "";
let iceUsername = "";
let icePassword = "";
let fingerprint = "";
let candidate = "";
for (const line of (d.sdp ?? "").split("\n")) {
if (line.startsWith("c=")) ip = line;
else if (line.startsWith("a=rtcp")) port = line.split(":")[1];
else if (line.startsWith("a=ice-ufrag")) iceUsername = line;
else if (line.startsWith("a=ice-pwd")) icePassword = line;
else if (line.startsWith("a=fingerprint")) fingerprint = line;
else if (line.startsWith("a=candidate")) candidate = line;
}
const audioPayloadType = CodecPayloadType.opus.payload_type;
const audioSection = `
m=audio ${port} UDP/TLS/RTP/SAVPF ${audioPayloadType}
${ip}
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=setup:passive
a=mid:0
a=maxptime:60
a=inactive
${iceUsername}
${icePassword}
${fingerprint}
${candidate}
a=rtcp-mux
a=rtpmap:${audioPayloadType} opus/48000/2
a=fmtp:${audioPayloadType} minptime=10;useinbandfec=1;usedtx=1
a=rtcp-fb:${audioPayloadType} transport-cc
a=rtcp-fb:${audioPayloadType} nack
a=ice-lite
`.trim();
const videoPayloads = Object.values(CodecPayloadType).filter(
(el) => el.type === "video",
);
const videoPayloadTypes = videoPayloads.flatMap((el) => [
el.payload_type,
el.rtx_payload_type ?? 0,
]);
const videoSection = `
m=video ${port} UDP/TLS/RTP/SAVPF ${videoPayloadTypes.join(" ")}
${ip}
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:13 urn:3gpp:video-orientation
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=setup:passive
a=mid:1
a=inactive
${iceUsername}
${icePassword}
${fingerprint}
${candidate}
a=rtcp-mux
a=ice-lite
`.trim();
const videoRtpMap = videoPayloads
.flatMap((el) => [
`a=rtpmap:${el.payload_type} ${el.name}/90000`,
`a=rtpmap:${el.rtx_payload_type} rtx/90000`,
`a=fmtp:${el.rtx_payload_type} apt=${el.payload_type}`,
`a=rtcp-fb:${el.payload_type} ccm fir`,
`a=rtcp-fb:${el.payload_type} nack`,
`a=rtcp-fb:${el.payload_type} nack pli`,
`a=rtcp-fb:${el.payload_type} goog-remb`,
`a=rtcp-fb:${el.payload_type} transport-cc`,
])
.join("\n");
this._webRtcWrapper.webRtcConn?.setRemoteDescription(
[audioSection, videoSection, videoRtpMap].join("\n"),
"answer",
);
this.emit("select_protocol_ack");
}
initDave(): void {
if (this._daveProtocolVersion) {
if (this._daveSession) {
this._daveSession.reinit(
this._daveProtocolVersion,
this.botId,
this.daveChannelId,
);
} else {
this._daveSession = new Davey.DAVESession(
this._daveProtocolVersion,
this.botId,
this.daveChannelId,
);
}
this.sendOpcodeBinary(
VoiceOpCodesBinary.MLS_KEY_PACKAGE,
this._daveSession.getSerializedKeyPackage(),
);
} else if (this._daveSession) {
this._daveSession.reset();
this._daveSession.setPassthroughMode(true, 10);
}
}
processInvalidCommit(transitionId: number): void {
this.sendOpcode(VoiceOpCodes.MLS_INVALID_COMMIT_WELCOME, {
transition_id: transitionId,
});
this.initDave();
}
executePendingTransition(transitionId: number): void {
const newVersion = this._davePendingTransitions.get(transitionId);
if (newVersion === undefined) {
console.error("Unrecognized transition ID", { transitionId });
return;
}
const oldVersion = this._daveProtocolVersion;
this._daveProtocolVersion = newVersion;
if (oldVersion !== newVersion && newVersion === 0) {
// Downgraded
this._daveDowngraded = true;
} else if (transitionId > 0 && this._daveDowngraded) {
this._daveDowngraded = false;
this._daveSession?.setPassthroughMode(true, 10);
}
this._davePendingTransitions.delete(transitionId);
}
setupEvents(): void {
this.ws?.addEventListener("message", async (e) => {
if (e.data instanceof ArrayBuffer) {
this.handleBinaryMessages(Buffer.from(e.data));
return;
}
const { op, d, seq } = JSON.parse(e.data as string) as {
op: number;
// biome-ignore lint/suspicious/noExplicitAny: Discord voice WS payload is dynamically typed
d: any;
seq?: number;
};
if (seq) this._sequenceNumber = seq;
if (op === VoiceOpCodes.READY) {
this.handleReady(d);
this.setProtocols().then(() => this.ready?.(this._webRtcWrapper));
this.setVideoAttributes(false);
} else if (op >= 4000) {
console.error(`${this.constructor.name} connection error`, d);
} else if (op === VoiceOpCodes.HELLO) {
this.setupHeartbeat(d.heartbeat_interval);
} else if (op === VoiceOpCodes.SELECT_PROTOCOL_ACK) {
await this.handleProtocolAck(d);
} else if (op === VoiceOpCodes.SPEAKING) {
// ignore speaking updates
} else if (op === VoiceOpCodes.HEARTBEAT_ACK) {
// ignore heartbeat acknowledgements
} else if (op === VoiceOpCodes.RESUMED) {
this.status.started = true;
} else if (op === VoiceOpCodes.CLIENTS_CONNECT) {
d.user_ids.forEach((id: string) => {
this._connectedUsers.add(id);
});
} else if (op === VoiceOpCodes.CLIENT_DISCONNECT) {
this._connectedUsers.delete(d.user_id);
} else if (op === VoiceOpCodes.DAVE_PREPARE_TRANSITION) {
this._davePendingTransitions.set(d.transition_id, d.protocol_version);
if (d.transition_id === 0) {
this.executePendingTransition(d.transition_id);
} else {
if (d.protocol_version === 0) {
this._daveSession?.setPassthroughMode(true, 120);
}
this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
transition_id: d.transition_id,
});
}
} else if (op === VoiceOpCodes.DAVE_EXECUTE_TRANSITION) {
this.executePendingTransition(d.transition_id);
} else if (op === VoiceOpCodes.DAVE_PREPARE_EPOCH) {
if (d.epoch === 1) {
this._daveProtocolVersion = d.protocol_version;
this.initDave();
}
}
});
}
handleBinaryMessages(msg: Buffer): void {
this._sequenceNumber = msg.readUint16BE(0);
const op = msg.readUint8(2);
switch (op) {
case VoiceOpCodesBinary.MLS_EXTERNAL_SENDER: {
this._daveSession?.setExternalSender(msg.subarray(3));
break;
}
case VoiceOpCodesBinary.MLS_PROPOSALS: {
const optype = msg.readUint8(3);
if (!this._daveSession) break;
const { commit, welcome } = this._daveSession.processProposals(
optype,
msg.subarray(4),
[...this._connectedUsers],
);
if (commit) {
this.sendOpcodeBinary(
VoiceOpCodesBinary.MLS_COMMIT_WELCOME,
welcome ? Buffer.concat([commit, welcome]) : commit,
);
}
break;
}
case VoiceOpCodesBinary.MLS_ANNOUNCE_COMMIT_TRANSITION: {
const transitionId = msg.readUInt16BE(3);
try {
this._daveSession?.processCommit(msg.subarray(5));
if (transitionId) {
this._davePendingTransitions.set(
transitionId,
this._daveProtocolVersion,
);
this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
transition_id: transitionId,
});
}
} catch (e) {
console.debug("MLS commit errored", e);
this.processInvalidCommit(transitionId);
}
break;
}
case VoiceOpCodesBinary.MLS_WELCOME: {
const transitionId = msg.readUInt16BE(3);
try {
this._daveSession?.processWelcome(msg.subarray(5));
if (transitionId) {
this._davePendingTransitions.set(
transitionId,
this._daveProtocolVersion,
);
this.sendOpcode(VoiceOpCodes.DAVE_TRANSITION_READY, {
transition_id: transitionId,
});
}
} catch (e) {
console.debug("MLS welcome errored", e);
this.processInvalidCommit(transitionId);
}
break;
}
}
}
get daveReady(): boolean {
return !!this._daveProtocolVersion && !!this._daveSession?.ready;
}
get daveSession(): Davey.DAVESession | null {
return this._daveSession;
}
setupHeartbeat(interval: number): void {
if (this.interval) {
clearInterval(this.interval);
}
this.interval = setInterval(() => {
try {
this.sendOpcode(VoiceOpCodes.HEARTBEAT, {
t: Date.now(),
seq_ack: this._sequenceNumber,
});
} catch {
/* ignore */
}
}, interval);
}
sendOpcode(code: number, data: unknown): void {
if (this.ws?.readyState !== WebSocket.OPEN) return;
this.ws.send(JSON.stringify({ op: code, d: data }));
}
sendOpcodeBinary(code: number, data: Uint8Array): void {
if (this.ws?.readyState !== WebSocket.OPEN) return;
const buf = Buffer.allocUnsafe(data.length + 1);
buf.writeUInt8(code);
Buffer.from(data).copy(buf, 1);
this.ws.send(buf);
}
/** serverId — overridden in VoiceConnection (guildId ?? channelId) and StreamConnection (rtc_server_id). */
get serverId(): string | null {
throw new Error("serverId not implemented");
}
/** identifies with media server with credentials */
identify(): void {
if (!this.serverId) throw new Error("Server ID is null or empty");
if (!this.session_id) throw new Error("Session ID is null or empty");
if (!this.token) throw new Error("Token is null or empty");
this.sendOpcode(VoiceOpCodes.IDENTIFY, {
server_id: this.serverId,
user_id: this.botId,
session_id: this.session_id,
token: this.token,
video: true,
streams: STREAMS_SIMULCAST,
max_dave_protocol_version: Davey.DAVE_PROTOCOL_VERSION ?? 0,
});
}
resume(): void {
if (!this.serverId) throw new Error("Server ID is null or empty");
if (!this.session_id) throw new Error("Session ID is null or empty");
if (!this.token) throw new Error("Token is null or empty");
this.sendOpcode(VoiceOpCodes.RESUME, {
server_id: this.serverId,
session_id: this.session_id,
token: this.token,
seq_ack: this._sequenceNumber,
});
}
/** Sets protocols and ip data used for video and audio (vp8 video, opus audio). */
async setProtocols(): Promise<void> {
if (!this._webRtcParams) throw new Error("WebRTC parameters not set");
if (!isNativeAvailable()) {
throw new Error(
"libdatachannel-min native binding not built — cannot start GoLive",
);
}
const reconnect = () => {
const webRtcConn = this._webRtcWrapper.initWebRtc();
webRtcConn.onStateChange((state) => {
if (state === "closed" && !this._closed) reconnect();
});
this._webRtcWrapper.onLocalDescription = (sdp) => {
const rtc_connection_id = randomUUID();
this.sendOpcode(VoiceOpCodes.SELECT_PROTOCOL, {
protocol: "webrtc",
codecs: Object.values(CodecPayloadType),
data: sdp,
sdp,
rtc_connection_id,
});
};
// createOffer (binding resolves full SDP incl. candidates after gathering)
void webRtcConn.createOffer().then((sdp) => {
this._webRtcWrapper.onLocalDescription?.(sdp);
});
};
reconnect();
return new Promise((resolve) => {
this.once("select_protocol_ack", () => resolve());
});
}
setVideoAttributes(enabled: boolean, attr?: VideoAttribute): void {
if (!this._webRtcParams) throw new Error("WebRTC parameters not set");
const { audioSsrc, videoSsrc, rtxSsrc } = this._webRtcParams;
if (!enabled) {
this.sendOpcode(VoiceOpCodes.VIDEO, {
audio_ssrc: audioSsrc,
video_ssrc: 0,
rtx_ssrc: 0,
streams: [],
});
} else {
if (!attr) throw new Error("Need to specify video attributes");
this.sendOpcode(VoiceOpCodes.VIDEO, {
audio_ssrc: audioSsrc,
video_ssrc: videoSsrc,
rtx_ssrc: rtxSsrc,
streams: [
{
type: "video",
rid: "100",
ssrc: videoSsrc,
active: true,
quality: 100,
rtx_ssrc: rtxSsrc,
// hardcode the max bitrate because we don't really know anyway
max_bitrate: 10000 * 1000,
max_framerate: enabled ? attr.fps : 0,
max_resolution: {
type: "fixed",
width: attr.width,
height: attr.height,
},
},
],
});
}
}
/** Set speaking status */
setSpeaking(speaking: boolean): void {
if (!this._webRtcParams) throw new Error("WebRTC connection not ready");
this.sendOpcode(VoiceOpCodes.SPEAKING, {
delay: 0,
speaking: speaking ? 1 : 0,
ssrc: this._webRtcParams.audioSsrc,
});
}
}
export type { NativePeerConnection };
@@ -1,175 +0,0 @@
/**
* BaseMediaStream — pacing/sync for GoLive frames. Ported from
* @dank074/discord-video-stream BaseMediaStream.js, minus node-av's
* AVFrame (frames are plain objects here) and debug-level (uses the GMW
* logger instead).
*/
import { Writable } from "node:stream";
import { setTimeout as sleep } from "node:timers/promises";
export interface GoLiveFrame {
data: Buffer | null;
pts: number;
duration: number;
timeBase: { num: number; den: number };
free?: () => void;
}
export class BaseMediaStream extends Writable {
_pts: number | undefined;
_syncTolerance = 20;
_noSleep: boolean;
_startTime: number | undefined;
_startPts: number | undefined;
_sync = true;
_syncStream: BaseMediaStream | undefined;
_type: string;
constructor(type: string, noSleep = false) {
super({ objectMode: true, highWaterMark: 0 });
this._type = type;
this._noSleep = noSleep;
}
get sync(): boolean {
return this._sync;
}
set sync(val: boolean) {
this._sync = val;
}
get syncStream(): BaseMediaStream | undefined {
return this._syncStream;
}
set syncStream(stream: BaseMediaStream | undefined) {
if (stream !== undefined && this === stream.syncStream) {
throw new Error("Cannot sync 2 streams with eachother");
}
this._syncStream = stream;
}
get noSleep(): boolean {
return this._noSleep;
}
set noSleep(val: boolean) {
this._noSleep = val;
if (!val) this.resetTimingCompensation();
}
get pts(): number | undefined {
return this._pts;
}
get syncTolerance(): number {
return this._syncTolerance;
}
set syncTolerance(n: number) {
if (n < 0) return;
this._syncTolerance = n;
}
async _sendFrame(_frame: Buffer, _frametime: number): Promise<void> {
throw new Error("Not implemented");
}
ptsDelta(): number | undefined {
if (this.pts !== undefined && this.syncStream?.pts !== undefined) {
return this.pts - this.syncStream.pts;
}
return undefined;
}
isAhead(): boolean {
const delta = this.ptsDelta();
return (
this.syncStream?.writableEnded === false &&
delta !== undefined &&
delta > this.syncTolerance
);
}
isBehind(): boolean {
const delta = this.ptsDelta();
return (
this.syncStream?.writableEnded === false &&
delta !== undefined &&
delta < -this.syncTolerance
);
}
resetTimingCompensation(): void {
this._startTime = this._startPts = undefined;
}
async _write(
frame: GoLiveFrame,
_encoding: BufferEncoding,
callback: (error?: Error | null) => void,
): Promise<void> {
const { data, pts, duration, timeBase } = frame;
if (!data) {
frame.free?.();
callback();
return;
}
const frametime = (Number(duration) / timeBase.den) * timeBase.num * 1000;
const start_sendFrame = performance.now();
await this._sendFrame(Buffer.from(data), frametime);
const end_sendFrame = performance.now();
this._pts = (Number(pts) / timeBase.den) * timeBase.num * 1000;
this.emit("pts", this._pts);
const sendTime = end_sendFrame - start_sendFrame;
const ratio = sendTime / frametime;
if (ratio > 1) {
// Frame takes longer to send than its frametime — warn once per 100
if (
this._lastWarnedRatio === undefined ||
ratio > this._lastWarnedRatio
) {
this._lastWarnedRatio = ratio;
}
}
this._startTime ??= start_sendFrame;
this._startPts ??= this._pts;
const sleepMs = Math.max(
0,
this._pts -
this._startPts +
frametime -
(end_sendFrame - this._startTime),
);
if (this._noSleep || sleepMs === 0) {
callback(null);
} else if (this.sync && this.isBehind()) {
// Stream is behind — don't sleep for this frame
this.resetTimingCompensation();
callback(null);
} else if (this.sync && this.isAhead()) {
// Stream is ahead — wait until the sync stream catches up
do {
await sleep(frametime);
} while (this.sync && this.isAhead());
this.resetTimingCompensation();
callback(null);
} else {
await sleep(sleepMs);
callback(null);
}
frame.free?.();
}
_lastWarnedRatio: number | undefined;
_destroy(
error: Error | null,
callback: (error?: Error | null) => void,
): void {
super._destroy(error, callback);
this.syncStream = undefined;
}
}
@@ -1,71 +0,0 @@
/** Payload types for Discord GoLive media — ported from @dank074/discord-video-stream. */
export interface CodecPayloadTypeEntry {
name: string;
type: "audio" | "video";
clockRate: number;
priority: number;
payload_type: number;
rtx_payload_type?: number;
encode?: boolean;
decode?: boolean;
}
export const CodecPayloadType: Record<string, CodecPayloadTypeEntry> = {
opus: {
name: "opus",
type: "audio",
clockRate: 48000,
priority: 1000,
payload_type: 120,
},
H264: {
name: "H264",
type: "video",
clockRate: 90000,
priority: 1000,
payload_type: 101,
rtx_payload_type: 102,
encode: true,
decode: true,
},
H265: {
name: "H265",
type: "video",
clockRate: 90000,
priority: 1000,
payload_type: 103,
rtx_payload_type: 104,
encode: true,
decode: true,
},
VP8: {
name: "VP8",
type: "video",
clockRate: 90000,
priority: 1000,
payload_type: 105,
rtx_payload_type: 106,
encode: true,
decode: true,
},
VP9: {
name: "VP9",
type: "video",
clockRate: 90000,
priority: 1000,
payload_type: 107,
rtx_payload_type: 108,
encode: true,
decode: true,
},
AV1: {
name: "AV1",
type: "video",
clockRate: 90000,
priority: 1000,
payload_type: 109,
rtx_payload_type: 110,
encode: true,
decode: true,
},
};
@@ -1,377 +0,0 @@
/**
* Lightweight demuxer — replaces node-av's LibavDemuxer for GoLive.
*
* Spawns ffmpeg to remux input into H264 AnnexB on stdout (video only —
* screen share doesn't need to mux audio into the demuxer; audio goes
* separately). This replaces the 114MB node-av binary with a plain ffmpeg
* spawn.
*
* Each video "frame" emitted is a complete NAL sequence terminated by a
* keyframe boundary (IDR). Audio is not extracted here — for GoLive with
* audio, the NUT mux + full demuxer would be needed; screen share audio is
* handled via a separate ffmpeg instance (see getDirectScreenInput).
*/
import { spawn } from "node:child_process";
import { randomUUID } from "node:crypto";
import { createWriteStream, existsSync, readdirSync } from "node:fs";
import { tmpdir } from "node:os";
import { join } from "node:path";
import { PassThrough } from "node:stream";
/**
* Resolve ffmpeg/ffprobe binary. Prefers explicit env override, then PATH,
* then a Nix-store ffmpeg-headless (the GMW flake provides it in the service
* profile, but dev shells / tests may not have it on PATH).
*/
function resolveBin(name: "ffmpeg"): string {
const override = process.env.FFMPEG_PATH;
if (override && existsSync(override)) return override;
// Nix store scan: <store>/<hash>-ffmpeg-headless-*/bin/<name>
const store = "/nix/store";
if (existsSync(store)) {
const entries = readdirSync(store);
for (const entry of entries) {
if (!entry.includes("ffmpeg-headless-")) continue;
const candidate = join(store, entry, "bin", name);
if (existsSync(candidate)) return candidate;
}
}
return name; // fall back to PATH
}
const FFMPEG = resolveBin("ffmpeg");
export const AVCodecID = {
AV_CODEC_ID_H264: 27,
AV_CODEC_ID_HEVC: 173,
AV_CODEC_ID_VP8: 139,
AV_CODEC_ID_VP9: 167,
AV_CODEC_ID_AV1: 225,
AV_CODEC_ID_OPUS: 86019,
} as const;
export type AVCodecID = (typeof AVCodecID)[keyof typeof AVCodecID];
export const AV_PKT_FLAG_KEY = 1;
export interface Frame {
data: Buffer | null;
pts: number;
duration: number;
timeBase: { num: number; den: number };
flags: number;
streamIndex: number;
free(): void;
}
export interface DemuxedStream {
codec: number;
codecName: string;
width: number;
height: number;
framerate_num: number;
framerate_den: number;
sample_rate: number;
stream: PassThrough;
}
/**
* Probe a media file for stream info using ffmpeg's stderr (the
* ffmpeg-headless Nix package ships ffmpeg but not ffprobe). Returns
* stream descriptors in the same shape ffprobe -show_streams would.
*/
export async function probeStreams(
url: string,
): Promise<Array<Record<string, unknown>>> {
return new Promise((resolve, reject) => {
const proc = spawn(FFMPEG, [
"-hide_banner",
"-loglevel",
"info",
"-i",
url,
"-f",
"null",
"-",
]);
let stderr = "";
proc.stderr.on("data", (d: Buffer) => (stderr += d.toString()));
proc.on("close", () => {
// Parse "Stream #0:0: Video: h264 (High), yuv420p, 640x360, 30 fps"
const streams: Array<Record<string, unknown>> = [];
const re = /Stream #0:(\d+): (Video|Audio): ([^,]+)/g;
let m: RegExpExecArray | null;
// biome-ignore lint/suspicious/noAssignInExpressions: regex loop idiom
while ((m = re.exec(stderr)) !== null) {
const [full, idx, kind, codecRaw] = m;
void full;
const codecName = codecRaw.split(" ")[0].toLowerCase();
const stream: Record<string, unknown> = {
index: Number(idx),
codec_type: kind.toLowerCase(),
codec_name: codecName,
width: 0,
height: 0,
r_frame_rate: "0/1",
sample_rate: 0,
};
// dimensions: "640x360"
const dim = /(\d{2,5})x(\d{2,5})/.exec(stderr.slice(m.index));
if (dim) {
stream.width = Number(dim[1]);
stream.height = Number(dim[2]);
}
// fps: "30 fps" or "29.97 fps"
const fps = /(\d+(?:\.\d+)?) fps/.exec(stderr.slice(m.index));
if (fps) {
const v = Number(fps[1]);
stream.r_frame_rate = `${Math.round(v * 1000)}/1000`;
}
// sample rate for audio: "48000 Hz"
const sr = /(\d+) Hz/.exec(stderr.slice(m.index));
if (sr) stream.sample_rate = Number(sr[1]);
streams.push(stream);
}
resolve(streams);
});
proc.on("error", (err) => reject(err));
});
}
/**
* Demux input (URL string or readable stream) into video frames on a
* PassThrough. Uses ffmpeg -f h264 -c copy for video-only AnnexB output.
* Returns stream info + the video pipe. Audio is not extracted (GoLive
* screen share sends silence / uses Discord's mixed audio).
*/
export async function demux(
input: string | PassThrough,
_opts: { format: string },
): Promise<{
video: DemuxedStream | undefined;
audio: DemuxedStream | undefined;
close: () => void;
}> {
const _label = randomUUID();
const vPipe = new PassThrough({ objectMode: true, highWaterMark: 128 });
const aPipe = new PassThrough({ objectMode: true, highWaterMark: 128 });
// For stream input, spool to a temp file first so ffprobe can inspect it
// (ffprobe needs a seekable file; pipes can't be re-read). The stream is
// fully consumed before ffmpeg starts — acceptable for screen-share
// sources which are already fully buffered by yt-dlp in practice.
let spoolPath: string | null = null;
const cleanupSpool = () => {
if (spoolPath) {
import("node:fs").then(({ unlink }) => unlink(spoolPath!, () => {}));
spoolPath = null;
}
};
let effectiveInput: string;
if (typeof input === "string") {
effectiveInput = input;
} else {
spoolPath = join(tmpdir(), `golive-demux-${_label}.h264`);
const ws = createWriteStream(spoolPath);
await new Promise<void>((resolve, reject) => {
input.pipe(ws);
input.on("error", reject);
ws.on("finish", resolve);
ws.on("error", reject);
});
effectiveInput = spoolPath;
}
// Probe for codec + dimensions
let streams: Array<Record<string, unknown>> = [];
try {
streams = await probeStreams(effectiveInput);
} catch (_e) {
// probe failed (e.g. raw h264 without container) — infer h264 default
streams = [];
}
const v = streams.find((s) => s.codec_type === "video");
const a = streams.find((s) => s.codec_type === "audio");
let vInfo: DemuxedStream | undefined;
let aInfo: DemuxedStream | undefined;
if (v) {
const codecName = (v.codec_name as string) ?? "h264";
const rFrame = (v.r_frame_rate as string) ?? "0/1";
const [num, den] = rFrame.split("/").map((n) => Number(n));
vInfo = {
codec:
AVCodecID[
(codecName.toUpperCase() as keyof typeof AVCodecID) ??
"AV_CODEC_ID_H264"
] ?? AVCodecID.AV_CODEC_ID_H264,
codecName,
width: (v.width as number) ?? 0,
height: (v.height as number) ?? 0,
framerate_num: num ?? 0,
framerate_den: den ?? 1,
sample_rate: 0,
stream: vPipe,
};
} else {
// Probe failed (e.g. raw AnnexB h264 input) — still emit frames on the
// video pipe; playStream infers dimensions from the first frame.
vInfo = {
codec: AVCodecID.AV_CODEC_ID_H264,
codecName: "h264",
width: 0,
height: 0,
framerate_num: 0,
framerate_den: 1,
sample_rate: 0,
stream: vPipe,
};
}
if (a) {
const codecName = (a.codec_name as string) ?? "opus";
aInfo = {
codec:
AVCodecID[
(codecName.toUpperCase() as keyof typeof AVCodecID) ??
"AV_CODEC_ID_OPUS"
],
codecName,
width: 0,
height: 0,
framerate_num: 0,
framerate_den: 0,
sample_rate: Number(a.sample_rate) ?? 0,
stream: aPipe,
};
}
// Spawn ffmpeg — extract raw video (AnnexB for H264) to stdout
const args: string[] = [
"-hide_banner",
"-loglevel",
"error",
"-i",
effectiveInput,
"-c:v",
"copy",
"-an", // no audio in this minimal demuxer
"-f",
"h264",
"pipe:1",
];
const proc = spawn(FFMPEG, args, { stdio: ["ignore", "pipe", "pipe"] });
// Scan stdout for NAL units. Each NAL unit (between start codes) is one frame
// payload. We emit them individually; the packetizer chain handles FU-A.
let videoBuf = Buffer.alloc(0);
let frameCount = 0;
const emitFrame = (nal: Uint8Array, isKeyFrame: boolean) => {
vPipe.write({
data: Buffer.from(nal),
pts: frameCount,
duration: 1,
timeBase: { num: 1, den: 90000 },
flags: isKeyFrame ? AV_PKT_FLAG_KEY : 0,
streamIndex: 0,
free: () => {},
});
frameCount++;
};
if (proc.stdout) {
proc.stdout.on("data", (chunk: Buffer) => {
videoBuf = Buffer.concat([videoBuf, chunk]);
// Find start codes (00 00 01 or 00 00 00 01) and split NALs
let start = 0;
// If buffer starts with zeros, that's the first start code — emit from there
while (start < videoBuf.length) {
let scPos = -1;
for (let i = start + 1; i < videoBuf.length - 2; i++) {
if (
videoBuf[i] === 0 &&
videoBuf[i + 1] === 0 &&
videoBuf[i + 2] === 1
) {
scPos = i + 3;
break;
}
}
if (scPos === -1) break;
// Emit the NAL from `start` to `scPos` (but skip the start code bytes at `start`)
if (start < scPos) {
let nalStart = start;
// Skip start code bytes for the NAL itself (00 00 01)
if (
videoBuf[nalStart] === 0 &&
videoBuf[nalStart + 1] === 0 &&
videoBuf[nalStart + 2] === 1
) {
nalStart += 3;
} else if (
nalStart + 3 < scPos &&
videoBuf[nalStart] === 0 &&
videoBuf[nalStart + 1] === 0 &&
videoBuf[nalStart + 2] === 0 &&
videoBuf[nalStart + 3] === 1
) {
nalStart += 4;
}
const nal = videoBuf.subarray(nalStart, scPos);
// Trim trailing zero bytes (from start code overlap)
let end = nal.length;
while (end > 0 && nal[end - 1] === 0) end--;
if (end > 0) {
const nalTrimmed = nal.subarray(0, end);
const isIdr = (nalTrimmed[0] & 0x1f) === 5; // IDR
emitFrame(nalTrimmed, isIdr);
}
}
// Skip the 00 00 01 at scPos-3 to find next
start = scPos;
// But the next start code needs at least 3 bytes
if (start > videoBuf.length - 3) break;
}
// Keep remaining bytes (potential partial NAL or start code)
if (start > 0 && start < videoBuf.length) {
videoBuf = videoBuf.subarray(start);
} else if (videoBuf.length > 4) {
// No full NAL found, but avoid unbounded growth
// Keep a sliding window
videoBuf = videoBuf.subarray(videoBuf.length - 3);
}
});
proc.stdout.on("end", () => {
if (videoBuf.length > 0) {
let end = videoBuf.length;
while (end > 0 && videoBuf[end - 1] === 0) end--;
if (end > 0) emitFrame(videoBuf.subarray(0, end), false);
}
vPipe.end();
aPipe.end();
});
}
if (proc.stderr) {
proc.stderr.on("data", () => {
/* errors swallowed */
});
}
proc.on("close", () => {
vPipe.end();
aPipe.end();
});
const close = () => {
proc.kill("SIGTERM");
vPipe.end();
aPipe.end();
cleanupSpool();
};
return { video: vInfo, audio: aInfo, close };
}
@@ -1,51 +0,0 @@
/**
* Lightweight encoders config — ported from @dank074/discord-video-stream
* encoders/software.js. Only software (libx264) is needed for GoLive.
*/
export interface EncoderSettings {
name: string;
options: string[];
outFilters?: string[];
globalOptions?: string[];
}
export interface EncoderSet {
H264: EncoderSettings;
H265: EncoderSettings;
VP8: EncoderSettings;
VP9: EncoderSettings;
AV1: EncoderSettings;
}
/** Software x264 encoder. Matches @dank074's software() defaults. */
export function software(
opts: {
x264?: { preset?: string; tune?: string };
x265?: { preset?: string; tune?: string };
} = {},
): () => EncoderSet {
const { x264, x265 } = opts;
const { preset: x264Preset = "superfast", tune: x264Tune = "film" } =
x264 ?? {};
const { preset: x265Preset = "superfast", tune: x265Tune } = x265 ?? {};
return () => ({
H264: {
name: "libx264",
options: ["-forced-idr 1", `-tune ${x264Tune}`, `-preset ${x264Preset}`],
},
H265: {
name: "libx265",
options: [
"-forced-idr 1",
...(x265Tune ? [`-tune ${x265Tune}`] : []),
`-preset ${x265Preset}`,
],
},
VP8: { name: "libvpx", options: ["-deadline 20000"] },
VP9: { name: "libvpx-vp9", options: ["-deadline 20000"] },
AV1: { name: "libsvtav1", options: [] },
});
}
export const Encoders = { software };
@@ -1,41 +0,0 @@
/** Discord gateway opcodes used by Streamer — ported from @dank074/discord-video-stream. */
export enum GatewayOpCodes {
DISPATCH = 0,
HEARTBEAT = 1,
IDENTIFY = 2,
PRESENCE_UPDATE = 3,
VOICE_STATE_UPDATE = 4,
VOICE_SERVER_PING = 5,
RESUME = 6,
RECONNECT = 7,
REQUEST_GUILD_MEMBERS = 8,
INVALID_SESSION = 9,
HELLO = 10,
HEARTBEAT_ACK = 11,
CALL_CONNECT = 13,
GUILD_SUBSCRIPTIONS = 14,
LOBBY_CONNECT = 15,
LOBBY_DISCONNECT = 16,
LOBBY_VOICE_STATES_UPDATE = 17,
STREAM_CREATE = 18,
STREAM_DELETE = 19,
STREAM_WATCH = 20,
STREAM_PING = 21,
STREAM_SET_PAUSED = 22,
REQUEST_GUILD_APPLICATION_COMMANDS = 24,
EMBEDDED_ACTIVITY_LAUNCH = 25,
EMBEDDED_ACTIVITY_CLOSE = 26,
EMBEDDED_ACTIVITY_UPDATE = 27,
REQUEST_FORUM_UNREADS = 28,
REMOTE_COMMAND = 29,
GET_DELETED_ENTITY_IDS_NOT_MATCHING_HASH = 30,
REQUEST_SOUNDBOARD_SOUNDS = 31,
SPEED_TEST_CREATE = 32,
SPEED_TEST_DELETE = 33,
REQUEST_LAST_MESSAGES = 34,
SEARCH_RECENT_MEMBERS = 35,
REQUEST_CHANNEL_STATUSES = 36,
GUILD_SUBSCRIPTIONS_BULK = 37,
GUILD_CHANNELS_RESYNC = 38,
REQUEST_CHANNEL_MEMBER_COUNT = 39,
}
@@ -1,291 +0,0 @@
/**
* H264 SPS VUI rewriter — ported from @dank074/discord-video-stream
* SPSVUIRewriter.js. Rewrites the SPS so Discord's receiver applies
* bitstream restrictions (max_num_reorder_frames=0, max_dec_frame_buffering
* bounded) — required for low-latency GoLive decode.
*/
import {
AnnexBBitstreamReader,
AnnexBBitstreamWriter,
} from "./AnnexBBitstreamReaderWriter.js";
export function rewriteSPSVUI(buffer: Uint8Array): Buffer {
const reader = new AnnexBBitstreamReader(buffer.subarray(1));
const writer = new AnnexBBitstreamWriter();
const readBit = (n = 1) => reader.readBits(n);
const writeBit = (v: number, n = 1) => writer.writeBits(v, n);
const readU = (n: number) => reader.readUnsigned(n);
const writeU = (v: number, n: number) => writer.writeUnsigned(v, n);
const readUE = () => reader.readUnsignedExpGolomb();
const writeUE = (v: number) => writer.writeUnsignedExpGolomb(v);
const readSE = () => reader.readSignedExpGolomb();
const writeSE = (v: number) => writer.writeSignedExpGolomb(v);
// Rewrite the NAL header
writeU(buffer[0], 8);
const profile_idc = readU(8);
writeU(profile_idc, 8);
const constraint_flags = readU(8);
writeU(constraint_flags, 8);
const level_idc = readU(8);
writeU(level_idc, 8);
const seq_parameter_set_id = readUE();
writeUE(seq_parameter_set_id);
// If profile in high profiles, additional fields
const highProfiles = new Set([
100, 110, 122, 244, 44, 83, 86, 118, 128, 138, 144,
]);
if (highProfiles.has(profile_idc)) {
const chroma_format_idc = readUE();
writeUE(chroma_format_idc);
if (chroma_format_idc === 3) {
const separate_colour_plane_flag = readBit(1);
writeBit(separate_colour_plane_flag, 1);
}
const bit_depth_luma_minus8 = readUE();
writeUE(bit_depth_luma_minus8);
const bit_depth_chroma_minus8 = readUE();
writeUE(bit_depth_chroma_minus8);
const qpprime_y_zero_transform_bypass_flag = readBit(1);
writeBit(qpprime_y_zero_transform_bypass_flag, 1);
const seq_scaling_matrix_present_flag = readBit(1);
writeBit(seq_scaling_matrix_present_flag, 1);
if (seq_scaling_matrix_present_flag) {
const scalingCount = chroma_format_idc !== 3 ? 8 : 12;
for (let i = 0; i < scalingCount; i++) {
const seq_scaling_list_present_flag = readBit(1);
writeBit(seq_scaling_list_present_flag, 1);
if (seq_scaling_list_present_flag) {
const size = i < 6 ? 16 : 64;
let lastScale = 8;
let nextScale = 8;
for (let j = 0; j < size; j++) {
const delta = readSE();
writeSE(delta);
nextScale = (lastScale + delta + 256) % 256;
if (nextScale !== 0) lastScale = nextScale;
}
}
}
}
}
const log2_max_frame_num_minus4 = readUE();
writeUE(log2_max_frame_num_minus4);
const pic_order_cnt_type = readUE();
writeUE(pic_order_cnt_type);
if (pic_order_cnt_type === 0) {
const log2_max_pic_order_cnt_lsb_minus4 = readUE();
writeUE(log2_max_pic_order_cnt_lsb_minus4);
} else if (pic_order_cnt_type === 1) {
const delta_pic_order_always_zero_flag = readBit(1);
writeBit(delta_pic_order_always_zero_flag, 1);
const offset_for_non_ref_pic = readSE();
writeSE(offset_for_non_ref_pic);
const offset_for_top_to_bottom_field = readSE();
writeSE(offset_for_top_to_bottom_field);
const num_ref_frames_in_pic_order_cnt_cycle = readUE();
writeUE(num_ref_frames_in_pic_order_cnt_cycle);
for (let i = 0; i < num_ref_frames_in_pic_order_cnt_cycle; i++) {
const offset_for_ref_frame = readSE();
writeSE(offset_for_ref_frame);
}
}
const max_num_ref_frames = readUE();
writeUE(max_num_ref_frames);
const gaps_in_frame_num_value_allowed_flag = readBit(1);
writeBit(gaps_in_frame_num_value_allowed_flag, 1);
const pic_width_in_mbs_minus1 = readUE();
writeUE(pic_width_in_mbs_minus1);
const pic_height_in_map_units_minus1 = readUE();
writeUE(pic_height_in_map_units_minus1);
const frame_mbs_only_flag = readBit(1);
writeBit(frame_mbs_only_flag, 1);
if (frame_mbs_only_flag === 0) {
const mb_adaptive_frame_field_flag = readBit(1);
writeBit(mb_adaptive_frame_field_flag, 1);
}
const direct_8x8_inference_flag = readBit(1);
writeBit(direct_8x8_inference_flag, 1);
const frame_cropping_flag = readBit(1);
writeBit(frame_cropping_flag, 1);
if (frame_cropping_flag) {
const frame_crop_left_offset = readUE();
writeUE(frame_crop_left_offset);
const frame_crop_right_offset = readUE();
writeUE(frame_crop_right_offset);
const frame_crop_top_offset = readUE();
writeUE(frame_crop_top_offset);
const frame_crop_bottom_offset = readUE();
writeUE(frame_crop_bottom_offset);
}
// https://webrtc.googlesource.com/src/+/5f2c9278f35e47ff72eb191669d473b7400c9f3e/common_video/h264/sps_vui_rewriter.cc#283
function addBitstreamRestriction() {
// motion_vectors_over_pic_boundaries_flag: u(1) — Default is 1 when not present.
writeBit(1, 1);
// max_bytes_per_pic_denom: ue(v) — Default is 2 when not present.
writeUE(2);
// max_bits_per_mb_denom: ue(v) — Default is 1 when not present.
writeUE(1);
// log2_max_mv_length_horizontal / vertical — both default to 16.
writeUE(16);
writeUE(16);
// IMPORTANT: max_num_reorder_frames must be 0 for low latency.
writeUE(0);
writeUE(max_num_ref_frames);
}
const vui_parameters_present_flag = readBit(1);
writeBit(1, 1);
// If no VUI exists, write one
if (!vui_parameters_present_flag) {
// aspect_ratio_info_present_flag, overscan_info_present_flag. Both u(1).
writeBit(0, 2);
// video_signal_type_present_flag, u(1) — write 0, ignore color space.
writeBit(0, 1);
// chroma_loc_info_present_flag, timing_info_present_flag,
// nal_hrd_parameters_present_flag, vcl_hrd_parameters_present_flag,
// pic_struct_present_flag — all u(1)
writeBit(0, 5);
// bitstream_restriction_flag: u(1)
writeBit(1, 1);
addBitstreamRestriction();
} else {
// VUI parsing and copying
const aspect_ratio_info_present_flag = readBit(1);
writeBit(aspect_ratio_info_present_flag, 1);
if (aspect_ratio_info_present_flag) {
const aspect_ratio_idc = readU(8);
writeU(aspect_ratio_idc, 8);
if (aspect_ratio_idc === 255) {
const sar_width = readU(16);
writeU(sar_width, 16);
const sar_height = readU(16);
writeU(sar_height, 16);
}
}
const overscan_info_present_flag = readBit(1);
writeBit(overscan_info_present_flag, 1);
if (overscan_info_present_flag) {
const overscan_appropriate_flag = readBit(1);
writeBit(overscan_appropriate_flag, 1);
}
// Read the video signal type, but don't copy it
const video_signal_type_present_flag = readBit(1);
writeBit(0, 1);
if (video_signal_type_present_flag) {
readBit(3); // _video_format
readBit(1); // _video_full_range_flag
const colour_description_present_flag = readBit(1);
if (colour_description_present_flag) {
readU(8); // _colour_primaries
readU(8); // _transfer_characteristics
readU(8); // _matrix_coeffs
}
}
const chroma_loc_info_present_flag = readBit(1);
writeBit(chroma_loc_info_present_flag, 1);
if (chroma_loc_info_present_flag) {
const chroma_sample_loc_type_top_field = readUE();
writeUE(chroma_sample_loc_type_top_field);
const chroma_sample_loc_type_bottom_field = readUE();
writeUE(chroma_sample_loc_type_bottom_field);
}
const timing_info_present_flag = readBit(1);
writeBit(timing_info_present_flag, 1);
if (timing_info_present_flag) {
const num_units_in_tick = readU(32);
writeU(num_units_in_tick, 32);
const time_scale = readU(32);
writeU(time_scale, 32);
const fixed_frame_rate_flag = readBit(1);
writeBit(fixed_frame_rate_flag, 1);
}
const nal_hrd_parameters_present_flag = readBit(1);
writeBit(nal_hrd_parameters_present_flag, 1);
if (nal_hrd_parameters_present_flag) {
// hrd_parameters()
const cpb_cnt_minus1 = readUE();
writeUE(cpb_cnt_minus1);
const bit_rate_scale = readBit(4);
writeBit(bit_rate_scale, 4);
const cpb_size_scale = readBit(4);
writeBit(cpb_size_scale, 4);
for (let i = 0; i <= cpb_cnt_minus1; i++) {
const bit_rate_value_minus1 = readUE();
writeUE(bit_rate_value_minus1);
const cpb_size_value_minus1 = readUE();
writeUE(cpb_size_value_minus1);
const cbr_flag = readBit(1);
writeBit(cbr_flag, 1);
}
const initial_cpb_removal_delay_length_minus1 = readBit(5);
writeBit(initial_cpb_removal_delay_length_minus1, 5);
const cpb_removal_delay_length_minus1 = readBit(5);
writeBit(cpb_removal_delay_length_minus1, 5);
const dpb_output_delay_length_minus1 = readBit(5);
writeBit(dpb_output_delay_length_minus1, 5);
const time_offset_length = readBit(5);
writeBit(time_offset_length, 5);
}
const vcl_hrd_parameters_present_flag = readBit(1);
writeBit(vcl_hrd_parameters_present_flag, 1);
if (vcl_hrd_parameters_present_flag) {
// hrd_parameters()
const cpb_cnt_minus1 = readUE();
writeUE(cpb_cnt_minus1);
const bit_rate_scale = readBit(4);
writeBit(bit_rate_scale, 4);
const cpb_size_scale = readBit(4);
writeBit(cpb_size_scale, 4);
for (let i = 0; i <= cpb_cnt_minus1; i++) {
const bit_rate_value_minus1 = readUE();
writeUE(bit_rate_value_minus1);
const cpb_size_value_minus1 = readUE();
writeUE(cpb_size_value_minus1);
const cbr_flag = readBit(1);
writeBit(cbr_flag, 1);
}
const initial_cpb_removal_delay_length_minus1 = readBit(5);
writeBit(initial_cpb_removal_delay_length_minus1, 5);
const cpb_removal_delay_length_minus1 = readBit(5);
writeBit(cpb_removal_delay_length_minus1, 5);
const dpb_output_delay_length_minus1 = readBit(5);
writeBit(dpb_output_delay_length_minus1, 5);
const time_offset_length = readBit(5);
writeBit(time_offset_length, 5);
}
if (nal_hrd_parameters_present_flag || vcl_hrd_parameters_present_flag) {
const low_delay_hrd_flag = readBit(1);
writeBit(low_delay_hrd_flag, 1);
}
const pic_struct_present_flag = readBit(1);
writeBit(pic_struct_present_flag, 1);
const bitstream_restriction_flag = readBit(1);
writeBit(1, 1);
if (!bitstream_restriction_flag) {
addBitstreamRestriction();
} else {
const motion_vectors_over_pic_boundaries_flag = readBit(1);
writeBit(motion_vectors_over_pic_boundaries_flag, 1);
const max_bytes_per_pic_denom = readUE();
writeUE(max_bytes_per_pic_denom);
const max_bits_per_mb_denom = readUE();
writeUE(max_bits_per_mb_denom);
const log2_max_mv_length_horizontal = readUE();
writeUE(log2_max_mv_length_horizontal);
const log2_max_mv_length_vertical = readUE();
writeUE(log2_max_mv_length_vertical);
readUE(); // _num_reorder_frames
writeUE(0);
readUE(); // _max_dec_frame_buffering
writeUE(max_num_ref_frames);
}
}
writeBit(1, 1); // rbsp_stop_one_bit
writer.flush();
return writer.toBuffer();
}
@@ -1,45 +0,0 @@
/**
* StreamConnection — GoLive stream connection (screen share).
* Ported from @dank074/discord-video-stream StreamConnection.js.
*/
import { BaseMediaConnection } from "./BaseMediaConnection.js";
import { VoiceOpCodes } from "./VoiceOpCodes.js";
export class StreamConnection extends BaseMediaConnection {
_streamKey: string | null = null;
_serverId: string | null = null;
setSpeaking(speaking: boolean): void {
if (!this.webRtcParams) throw new Error("WebRTC connection not ready");
this.sendOpcode(VoiceOpCodes.SPEAKING, {
delay: 0,
speaking: speaking ? 2 : 0,
ssrc: this.webRtcParams.audioSsrc,
});
}
get daveChannelId(): string {
if (this._serverId === null) {
throw new Error("Server ID not set (this shouldn't happen)");
}
const channelId = BigInt(this._serverId) - 1n;
return channelId.toString();
}
get serverId(): string | null {
return this._serverId;
}
set serverId(id: string | null) {
this._serverId = id;
}
get streamKey(): string | null {
return this._streamKey;
}
set streamKey(value: string | null) {
this._streamKey = value;
}
}
@@ -1,280 +0,0 @@
/**
* Streamer — gateway-level GoLive controller. Ported from
* @dank074/discord-video-stream Streamer.js.
*
* Drives the Discord gateway (VOICE_STATE_UPDATE, STREAM_CREATE, ...) and
* hands back a VoiceConnection / StreamConnection once the media server
* session is ready.
*/
import { EventEmitter } from "node:events";
import { GatewayOpCodes } from "./GatewayOpCodes.js";
import type { NativePeerConnection } from "./native.js";
import { StreamConnection } from "./StreamConnection.js";
import { generateStreamKey, parseStreamKey } from "./utils.js";
import { VoiceConnection } from "./VoiceConnection.js";
import type { WebRtcConnWrapper } from "./WebRtcWrapper.js";
/** Minimal surface of a discord.js-selfbot-v13 client used by Streamer. */
export interface StreamerClientLike {
user: { id: string; username?: string } | null;
token: string | null;
on(
event: "raw",
listener: (packet: { t: string; d: unknown }) => void,
): unknown;
ws: {
broadcast(data: { op: number; d: unknown }): void;
};
guilds?: {
// biome-ignore lint/suspicious/noExplicitAny: discord.js-selfbot client shape is dynamic
fetch(id: string): Promise<any>;
};
}
/** Minimal channel shape accepted by joinVoiceChannel. */
export interface VoiceChannelLike {
id: string;
type: string;
guildId?: string | null;
}
export class Streamer {
_voiceConnection: VoiceConnection | null = null;
_client: StreamerClientLike;
_gatewayEmitter = new EventEmitter();
constructor(client: StreamerClientLike) {
this._client = client;
// listen for gateway dispatch events
this.client.on("raw", (packet) => {
this._gatewayEmitter.emit(packet.t, packet.d);
});
}
get client(): StreamerClientLike {
return this._client;
}
get opts(): Record<string, unknown> {
return {};
}
get voiceConnection(): VoiceConnection | null {
return this._voiceConnection;
}
sendOpcode(code: number, data: unknown): void {
this.client.ws.broadcast({ op: code, d: data });
}
joinVoiceChannel(channel: VoiceChannelLike): Promise<WebRtcConnWrapper> {
let guildId: string | null = null;
if (
channel.type === "GUILD_STAGE_VOICE" ||
channel.type === "GUILD_VOICE"
) {
guildId = channel.guildId ?? null;
}
return this.joinVoice(guildId, channel.id);
}
/**
* Joins a voice channel and resolves with the WebRtcConnWrapper when the
* media session is ready.
*/
joinVoice(
guild_id: string | null,
channel_id: string,
): Promise<WebRtcConnWrapper> {
return new Promise((resolve, reject) => {
if (!this.client.user) {
reject(new Error("Client not logged in"));
return;
}
const user_id = this.client.user.id;
const voiceConn = new VoiceConnection(
this,
guild_id,
user_id,
channel_id,
(conn) => {
resolve(conn);
},
);
this._voiceConnection = voiceConn;
this._gatewayEmitter.on(
"VOICE_STATE_UPDATE",
(d: { user_id: string; session_id: string }) => {
if (user_id !== d.user_id) return;
voiceConn.setSession(d.session_id);
},
);
this._gatewayEmitter.on(
"VOICE_SERVER_UPDATE",
(d: {
guild_id: string | null;
channel_id?: string;
endpoint: string;
token: string;
}) => {
if (guild_id !== d.guild_id) return;
// channel_id is not set for guild voice calls
if (d.channel_id && channel_id !== d.channel_id) return;
voiceConn.setTokens(d.endpoint, d.token);
},
);
this.signalVideo(false);
});
}
/** Create a GoLive stream (screen share) on top of the voice connection. */
createStream(): Promise<WebRtcConnWrapper> {
return new Promise((resolve, reject) => {
if (!this.client.user) {
reject(new Error("Client not logged in"));
return;
}
if (!this.voiceConnection) {
reject(
new Error("cannot start stream without first joining voice channel"),
);
return;
}
this.signalStream();
const {
guildId: clientGuildId,
channelId: clientChannelId,
session_id,
} = this.voiceConnection;
const clientUserId = this.client.user.id;
if (!session_id) throw new Error("Session doesn't exist yet");
const streamConn = new StreamConnection(
this,
clientGuildId,
clientUserId,
clientChannelId,
(conn) => {
resolve(conn);
},
);
this.voiceConnection.streamConnection = streamConn;
this._gatewayEmitter.on(
"STREAM_CREATE",
(d: { stream_key: string; rtc_server_id: string }) => {
const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
if (
clientGuildId !== guildId ||
clientChannelId !== channelId ||
clientUserId !== userId
) {
return;
}
streamConn.serverId = d.rtc_server_id;
streamConn.streamKey = d.stream_key;
streamConn.setSession(session_id);
},
);
this._gatewayEmitter.on(
"STREAM_SERVER_UPDATE",
(d: { stream_key: string; endpoint: string; token: string }) => {
const { channelId, guildId, userId } = parseStreamKey(d.stream_key);
if (
clientGuildId !== guildId ||
clientChannelId !== channelId ||
clientUserId !== userId
) {
return;
}
streamConn.setTokens(d.endpoint, d.token);
},
);
});
}
async setStreamPreview(image: Buffer): Promise<void> {
if (!this.client.token) throw new Error("Please login :)");
if (!this.voiceConnection?.streamConnection?.guildId) return;
const data = `data:image/jpeg;base64,${image.toString("base64")}`;
const { guildId } = this.voiceConnection.streamConnection;
if (!this.client.guilds) return;
const server = await this.client.guilds.fetch(guildId);
// biome-ignore lint/suspicious/noExplicitAny: discord.js-selfbot dynamic
(server as any).members.me?.voice?.postPreview(data);
}
stopStream(): void {
const stream = this.voiceConnection?.streamConnection;
if (!stream) return;
stream.stop();
this.signalStopStream();
this.voiceConnection.streamConnection = null;
this._gatewayEmitter.removeAllListeners("STREAM_CREATE");
this._gatewayEmitter.removeAllListeners("STREAM_SERVER_UPDATE");
}
leaveVoice(): void {
this.voiceConnection?.stop();
this.signalLeaveVoice();
this._voiceConnection = null;
this._gatewayEmitter.removeAllListeners("VOICE_STATE_UPDATE");
this._gatewayEmitter.removeAllListeners("VOICE_SERVER_UPDATE");
}
signalVideo(video_enabled: boolean): void {
if (!this.voiceConnection) return;
const { guildId: guild_id, channelId: channel_id } = this.voiceConnection;
this.sendOpcode(GatewayOpCodes.VOICE_STATE_UPDATE, {
guild_id: guild_id,
channel_id,
self_mute: false,
self_deaf: true,
self_video: video_enabled,
});
}
signalStream(): void {
if (!this.voiceConnection) return;
const {
type,
guildId: guild_id,
channelId: channel_id,
botId: user_id,
} = this.voiceConnection;
this.sendOpcode(GatewayOpCodes.STREAM_CREATE, {
type,
guild_id,
channel_id,
preferred_region: null,
});
this.sendOpcode(GatewayOpCodes.STREAM_SET_PAUSED, {
stream_key: generateStreamKey(type, guild_id, channel_id, user_id),
paused: false,
});
}
signalStopStream(): void {
if (!this.voiceConnection) return;
const {
type,
guildId: guild_id,
channelId: channel_id,
botId: user_id,
} = this.voiceConnection;
this.sendOpcode(GatewayOpCodes.STREAM_DELETE, {
stream_key: generateStreamKey(type, guild_id, channel_id, user_id),
});
}
signalLeaveVoice(): void {
this.sendOpcode(GatewayOpCodes.VOICE_STATE_UPDATE, {
guild_id: null,
channel_id: null,
self_mute: true,
self_deaf: false,
self_video: false,
});
}
}
export type { NativePeerConnection };
@@ -1,20 +0,0 @@
/**
* VideoStream — feeds encoded H264 frames into the WebRTC connection.
* Ported from @dank074/discord-video-stream VideoStream.js.
*/
import { BaseMediaStream } from "./BaseMediaStream.js";
import type { WebRtcConnWrapper } from "./WebRtcWrapper.js";
export class VideoStream extends BaseMediaStream {
_conn: WebRtcConnWrapper;
constructor(conn: WebRtcConnWrapper, noSleep = false) {
super("video", noSleep);
this._conn = conn;
}
async _sendFrame(frame: Buffer, frametime: number): Promise<void> {
this._conn.sendVideoFrame(frame, frametime);
}
}
@@ -1,25 +0,0 @@
/**
* VoiceConnection — guild/DM voice channel GoLive connection.
* Ported from @dank074/discord-video-stream VoiceConnection.js.
*/
import { BaseMediaConnection } from "./BaseMediaConnection.js";
import type { StreamConnection } from "./StreamConnection.js";
export class VoiceConnection extends BaseMediaConnection {
streamConnection: StreamConnection | null = null;
get daveChannelId(): string {
return this.channelId;
}
get serverId(): string | null {
// for guild vc it is the guild id, for dm voice it is the channel id
return this.guildId ?? this.channelId;
}
stop(): void {
super.stop();
this.streamConnection?.stop();
}
}
@@ -1,38 +0,0 @@
/** Discord voice WebSocket opcodes — ported from @dank074/discord-video-stream. */
export enum VoiceOpCodes {
IDENTIFY = 0,
SELECT_PROTOCOL = 1,
READY = 2,
HEARTBEAT = 3,
SELECT_PROTOCOL_ACK = 4,
SPEAKING = 5,
HEARTBEAT_ACK = 6,
RESUME = 7,
HELLO = 8,
RESUMED = 9,
CLIENTS_CONNECT = 11,
VIDEO = 12,
CLIENT_DISCONNECT = 13,
SESSION_UPDATE = 14,
MEDIA_SINK_WANTS = 15,
VOICE_BACKEND_VERSION = 16,
CHANNEL_OPTIONS_UPDATE = 17,
FLAGS = 18,
SPEED_TEST = 19,
PLATFORM = 20,
DAVE_PREPARE_TRANSITION = 21,
DAVE_EXECUTE_TRANSITION = 22,
DAVE_TRANSITION_READY = 23,
DAVE_PREPARE_EPOCH = 24,
MLS_INVALID_COMMIT_WELCOME = 31,
}
/** Binary voice WebSocket opcodes (DAVE / MLS). */
export enum VoiceOpCodesBinary {
MLS_EXTERNAL_SENDER = 25,
MLS_KEY_PACKAGE = 26,
MLS_PROPOSALS = 27,
MLS_COMMIT_WELCOME = 28,
MLS_ANNOUNCE_COMMIT_TRANSITION = 29,
MLS_WELCOME = 30,
}
@@ -1,205 +0,0 @@
/**
* WebRTC connection wrapper for GoLive — ported from
* @dank074/discord-video-stream WebRtcWrapper.js, with the media stack
* (packetizers, RTCP SR/NACK, pacing) provided by the libdatachannel-min
* binding instead of node-datachannel's JS-exposed media classes.
*/
import {
H264Helpers,
H264NalUnitTypes,
splitNalu,
startCode3,
} from "./AnnexBHelper.js";
import { CodecPayloadType } from "./CodecPayloadType.js";
import type { NativePeerConnection, NativeTrack } from "./native.js";
import { loadNative } from "./native.js";
import { rewriteSPSVUI } from "./SPSVUIRewriter.js";
import { normalizeVideoCodec } from "./utils.js";
export type WebRtcVideoCodec = "H264" | "H265" | "VP8" | "VP9" | "AV1";
export interface WebRtcParams {
address: string;
port: number;
audioSsrc: number;
videoSsrc: number;
rtxSsrc: number;
supportedEncryptionModes: string[];
}
/** Minimal surface of the media connection that WebRtcWrapper drives. */
export interface VideoAttribute {
fps: number;
width: number;
height: number;
}
export interface MediaConnectionLike {
daveReady: boolean;
daveSession: {
encryptOpus(frame: Buffer): Buffer;
encrypt(mediaType: number, codec: number, frame: Buffer): Buffer;
} | null;
webRtcParams: WebRtcParams | null;
setSpeaking(speaking: boolean): void;
setVideoAttributes(enabled: boolean, attr?: VideoAttribute): void;
}
/** Media types used by DAVE encryption (from @dank074). */
export enum DaveMediaType {
AUDIO = 0,
VIDEO = 1,
}
/** DAVE codec ids (from @dank074). */
export enum DaveCodec {
UNKNOWN = 0,
VP8 = 2,
VP9 = 3,
H264 = 4,
H265 = 5,
AV1 = 6,
}
export class WebRtcConnWrapper {
private _mediaConn: MediaConnectionLike;
private _webRtcConn: NativePeerConnection | null = null;
private _audioTrack: NativeTrack | null = null;
private _videoTrack: NativeTrack | null = null;
private _videoCodec: WebRtcVideoCodec | null = null;
/** Assigned by BaseMediaConnection to send the gathered SDP to Discord. */
onLocalDescription: ((sdp: string) => void) | null = null;
constructor(mediaConn: MediaConnectionLike) {
this._mediaConn = mediaConn;
}
initWebRtc(): NativePeerConnection {
const native = loadNative();
this._webRtcConn = new native.PeerConnection({
iceServers: ["stun:stun.l.google.com:19302"],
});
// Track mids must match @dank074: "0" audio, "1" video.
this._audioTrack = this._webRtcConn.addTrack("0", "audio");
this._videoTrack = this._webRtcConn.addTrack("1", "video");
return this._webRtcConn;
}
close(): void {
this._webRtcConn?.close();
this._webRtcConn = null;
}
get webRtcConn(): NativePeerConnection | null {
return this._webRtcConn;
}
get ready(): boolean {
return this._webRtcConn?.state() === "connected";
}
get mediaConnection(): MediaConnectionLike {
return this._mediaConn;
}
sendAudioFrame(frame: Buffer, frametime: number): void {
if (!this.ready || !this._audioTrack) return;
const clockRate = CodecPayloadType.opus.clockRate;
if (this.mediaConnection.daveReady && this.mediaConnection.daveSession) {
frame = this.mediaConnection.daveSession.encryptOpus(frame);
}
this._audioTrack.sendFrame(frame);
this._audioTrack.addTimestamp(Math.round((frametime * clockRate) / 1000));
}
sendVideoFrame(frame: Buffer, frametime: number): void {
if (!this.ready || !this._videoTrack) return;
const clockRate = CodecPayloadType[this._videoCodec ?? "H264"].clockRate;
if (this._videoCodec === "H264") {
let spsRewritten = false;
const nalus = splitNalu(frame).map((el) => {
if (H264Helpers.getUnitType(el) === H264NalUnitTypes.SPS) {
spsRewritten = true;
return rewriteSPSVUI(el);
}
return el;
});
if (spsRewritten)
frame = Buffer.concat(nalus.flatMap((el) => [startCode3, el]));
}
if (this.mediaConnection.daveReady && this.mediaConnection.daveSession) {
let daveCodec = DaveCodec.UNKNOWN;
switch (this._videoCodec) {
case "H264":
daveCodec = DaveCodec.H264;
break;
case "H265":
daveCodec = DaveCodec.H265;
break;
case "VP8":
daveCodec = DaveCodec.VP8;
break;
case "VP9":
daveCodec = DaveCodec.VP9;
break;
case "AV1":
daveCodec = DaveCodec.AV1;
break;
default:
break;
}
frame = this.mediaConnection.daveSession.encrypt(
DaveMediaType.VIDEO,
daveCodec,
frame,
);
}
this._videoTrack.sendFrame(frame);
this._videoTrack.addTimestamp(Math.round((frametime * clockRate) / 1000));
}
setPacketizer(videoCodec: string): void {
if (!this.mediaConnection.webRtcParams) {
throw new Error("WebRTC connection not ready");
}
const { audioSsrc, videoSsrc } = this.mediaConnection.webRtcParams;
this._videoCodec = normalizeVideoCodec(videoCodec);
// Audio packetizer: opus 120 @ 48kHz, playout delay ext id 5 (like @dank074)
this._audioTrack?.setPacketizer(
"audio",
audioSsrc,
CodecPayloadType.opus.payload_type,
CodecPayloadType.opus.clockRate,
5,
0,
1,
);
// Video packetizer: H264/H265/AV1 with their payload types
const codecEntry = CodecPayloadType[this._videoCodec];
if (!codecEntry) {
throw new Error(`Packetizer not implemented for ${this._videoCodec}`);
}
const nativeKind =
this._videoCodec === "H264"
? "h264"
: this._videoCodec === "H265"
? "h265"
: this._videoCodec === "AV1"
? "av1"
: (() => {
throw new Error(
`Packetizer not implemented for ${this._videoCodec}`,
);
})();
this._videoTrack?.setPacketizer(
nativeKind,
videoSsrc,
codecEntry.payload_type,
codecEntry.clockRate,
5,
0,
10,
);
}
}
@@ -1,19 +0,0 @@
/**
* goLive public API — re-exports the ported @dank074 modules.
* Drop-in replacement for `@dank074/discord-video-stream` in
* screenShareController.ts.
*/
export { AudioStream } from "./AudioStream.js";
export { BaseMediaConnection } from "./BaseMediaConnection.js";
export { BaseMediaStream } from "./BaseMediaStream.js";
export { CodecPayloadType } from "./CodecPayloadType.js";
export { demux } from "./Demuxer.js";
export { Encoders } from "./Encoders.js";
export { playStream, prepareStream } from "./prepareStream.js";
export { StreamConnection } from "./StreamConnection.js";
export { Streamer } from "./Streamer.js";
export { normalizeVideoCodec } from "./utils.js";
export { VideoStream } from "./VideoStream.js";
export { VoiceConnection } from "./VoiceConnection.js";
export { WebRtcConnWrapper } from "./WebRtcWrapper.js";
@@ -1,116 +0,0 @@
/**
* Loader + typings for the minimal libdatachannel N-API binding
* (native/libdatachannel-min). The binding exposes ONLY what GoLive needs:
* PeerConnection, DataChannel, Track (raw RTP + media packetizer chain).
*
* The .node file is built by node-gyp against libdatachannel 0.24.0 (built
* from source — nixpkgs 0.24.1 is glibc-incompatible with this host). It is
* NOT shipped via npm; the Nix flake builds it as part of the gateway.
*/
export interface NativeTrack {
/** Send a RAW RTP/RTCP packet (no media handler installed). */
send(buffer: Uint8Array): void;
/** Send an ENCODED frame; the packetizer chain turns it into RTP. */
sendFrame(buffer: Uint8Array): void;
/** Advance the packetizer RTP timestamp by delta (clock-rate units). */
addTimestamp(delta: number): void;
/** Install the media-handler chain (packetizer → RTCP SR → NACK → pacing). */
setPacketizer(
kind: "audio" | "h264" | "h265" | "av1",
ssrc: number,
payloadType: number,
clockRate: number,
playoutDelayId: number,
playoutDelayMin: number,
playoutDelayMax: number,
): void;
isOpen(): boolean;
close(): void;
}
export interface NativePeerConnection {
/** mid must be "0" (audio) or "1" (video) — matches @dank074's track defs. */
addTrack(mid: string, kind: "audio" | "video"): NativeTrack;
/** Resolves with the full SDP (incl. candidates) after gathering completes. */
createOffer(): Promise<string>;
/** Resolves with the auto-generated answer SDP. */
createAnswer(offerSdp: string): Promise<string>;
setRemoteDescription(sdp: string, type: "offer" | "answer"): void;
state(): string;
close(): void;
onStateChange(cb: (state: string) => void): void;
}
export interface NativeBinding {
PeerConnection: new (config: {
iceServers: string[];
}) => NativePeerConnection;
DataChannel: unknown;
Track: unknown;
}
let cached: NativeBinding | null = null;
/** Load the native binding. Throws only if the .node is truly missing —
* callers (screen share) guard with `isNativeAvailable()`. */
export function loadNative(): NativeBinding {
if (cached) return cached;
// Resolve relative to this file: src/goLive/ → native/libdatachannel-min/
const candidates = [
new URL(
"../../native/libdatachannel-min/build/Release/datachannel_min.node",
import.meta.url,
),
new URL(
"../../../native/libdatachannel-min/build/Release/datachannel_min.node",
import.meta.url,
),
];
let lastErr: unknown;
for (const url of candidates) {
try {
// @ts-expect-error — .node modules are not typed; dynamic require via file URL
const mod = process.dlopen ? null : null;
void mod;
const nativePath = url.pathname;
// eslint-disable-next-line @typescript-eslint/no-require-imports
const req = createRequire(import.meta.url);
const binding = req(nativePath) as NativeBinding;
if (typeof binding.PeerConnection === "function") {
cached = binding;
return binding;
}
} catch (e) {
lastErr = e;
}
}
// Fallback: plain relative require (tsx / jest environments)
try {
const req = createRequire(import.meta.url);
const binding = req(
"../../native/libdatachannel-min/build/Release/datachannel_min.node",
) as NativeBinding;
if (typeof binding.PeerConnection === "function") {
cached = binding;
return binding;
}
} catch (e) {
lastErr = e;
}
throw new Error(
`libdatachannel-min native binding not built (${String(lastErr)}). Run: cd native/libdatachannel-min && npx node-gyp rebuild`,
);
}
import { createRequire } from "node:module";
/** True when the native binding is built — screen share stays disabled otherwise. */
export function isNativeAvailable(): boolean {
try {
loadNative();
return true;
} catch {
return false;
}
}
@@ -1,325 +0,0 @@
/**
* prepareStream & playStream — ported from @dank074/discord-video-stream
* newApi.js (Encoders/prepareStream/playStream), but uses `child_process.spawn`
* + ffmpeg CLI args directly instead of fluent-ffmpeg + node-av.
*
* Replaces the @dank074 video pipeline entirely:
* input (URL or Readable) → ffmpeg spawn → H264 AnnexB frames
* → Demuxer stream → VideoStream/AudioStream → WebRtcConnWrapper
*/
import { type ChildProcess, spawn } from "node:child_process";
import { existsSync, readdirSync } from "node:fs";
import { join } from "node:path";
import { PassThrough, type Readable } from "node:stream";
import { demux } from "./Demuxer.js";
import { type EncoderSettings, Encoders } from "./Encoders.js";
import { VideoStream } from "./VideoStream.js";
import type { WebRtcConnWrapper } from "./WebRtcWrapper.js";
export interface PrepareStreamResult {
command: ChildProcess;
output: PassThrough;
encoder: () => Record<string, EncoderSettings>;
options: Record<string, unknown>;
videoCodec: string;
width: number;
height: number;
frameRate?: number;
includeAudio: boolean;
}
function isFiniteNonZero(n: unknown): n is number {
return typeof n === "number" && !!n && Number.isFinite(n);
}
const DEFAULT_HEADERS = {
"User-Agent":
"Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/107.0.0.0 Safari/537.36",
Connection: "keep-alive",
};
/** Resolve ffmpeg binary (env override → PATH → Nix store ffmpeg-headless). */
function resolveFfmpeg(): string {
if (process.env.FFMPEG_PATH && existsSync(process.env.FFMPEG_PATH)) {
return process.env.FFMPEG_PATH;
}
const store = "/nix/store";
if (existsSync(store)) {
const entries = readdirSync(store);
for (const entry of entries) {
if (!entry.includes("ffmpeg-headless-")) continue;
const candidate = join(store, entry, "bin", "ffmpeg");
if (existsSync(candidate)) return candidate;
}
}
return "ffmpeg";
}
const FFMPEG_BIN = resolveFfmpeg();
/**
* prepareStream — build an ffmpeg command (as spawn args + PassThrough output)
* that transcodes the input into a pipe we can demux. Mirrors @dank074's
* prepareStream but produces a raw spawn instead of a fluent-ffmpeg command.
*/
export function prepareStream(
input: string | Readable,
options: Record<string, unknown> = {},
): PrepareStreamResult {
const mergedOptions = {
noTranscoding: false,
width: isFiniteNonZero(options.width)
? Math.round(options.width as number)
: -2,
height: isFiniteNonZero(options.height)
? Math.round(options.height as number)
: -2,
frameRate:
isFiniteNonZero(options.frameRate) && (options.frameRate as number) > 0
? options.frameRate
: undefined,
videoCodec: (options.videoCodec as string) ?? "H264",
bitrateVideo:
isFiniteNonZero(options.bitrateVideo) &&
(options.bitrateVideo as number) > 0
? Math.round(options.bitrateVideo as number)
: 5000,
bitrateVideoMax:
isFiniteNonZero(options.bitrateVideoMax) &&
(options.bitrateVideoMax as number) > 0
? Math.round(options.bitrateVideoMax as number)
: 7000,
bitrateAudio:
isFiniteNonZero(options.bitrateAudio) &&
(options.bitrateAudio as number) > 0
? Math.round(options.bitrateAudio as number)
: 128,
includeAudio: options.includeAudio ?? true,
encoder:
(options.encoder as () => Record<string, EncoderSettings>) ??
Encoders.software(),
customHeaders: {
...DEFAULT_HEADERS,
...(options.customHeaders as Record<string, string> | undefined),
},
customInputOptions: (options.customInputOptions as string[]) ?? [],
customFfmpegFlags: (options.customFfmpegFlags as string[]) ?? [],
minimizeLatency: options.minimizeLatency ?? false,
};
const output = new PassThrough();
const args: string[] = [
"-hide_banner",
"-loglevel",
"error",
...(typeof input === "string" ? ["-i", input] : ["-i", "pipe:0"]),
...mergedOptions.customInputOptions,
];
if (mergedOptions.minimizeLatency) {
args.push("-fflags", "nobuffer", "-analyzeduration", "0");
}
if (typeof input === "string" && input.startsWith("http")) {
const headerStr = Object.entries(mergedOptions.customHeaders)
.map(([k, v]) => `${k}: ${v}`)
.join("\r\n");
args.push(
"-headers",
headerStr,
"-reconnect",
"1",
"-reconnect_at_eof",
"1",
"-reconnect_streamed",
"1",
"-reconnect_delay_max",
"4294",
);
}
// Video
args.push("-map", "0:v:0");
if (mergedOptions.noTranscoding) {
args.push("-c:v", "copy");
} else {
args.push(`-vf`, `scale=${mergedOptions.width}:${mergedOptions.height}`);
if (mergedOptions.frameRate)
args.push("-r", String(mergedOptions.frameRate));
const enc = mergedOptions.encoder()[mergedOptions.videoCodec];
if (!enc)
throw new Error(
`Encoder settings not specified for ${mergedOptions.videoCodec}`,
);
// Encoder options are declared as single strings like "-forced-idr 1";
// spawn needs each flag and value as separate argv entries.
const encOptions = enc.options.flatMap((opt) =>
opt.split(/\s+/).filter(Boolean),
);
args.push(
"-b:v",
`${mergedOptions.bitrateVideo}k`,
"-maxrate:v",
`${mergedOptions.bitrateVideoMax}k`,
"-bufsize:v",
`${Math.round(mergedOptions.bitrateVideo / 2)}k`,
"-bf",
"0",
"-pix_fmt",
"yuv420p",
"-force_key_frames",
"expr:gte(t,n_forced*1)",
"-c:v",
enc.name,
...encOptions,
...(enc.globalOptions ?? []).flatMap((opt) =>
opt.split(/\s+/).filter(Boolean),
),
);
}
// Audio
if (mergedOptions.includeAudio) {
args.push("-map", "0:a:0?");
args.push(
"-c:a",
"libopus",
"-b:a",
`${mergedOptions.bitrateAudio}k`,
"-ar",
"48000",
"-ac",
"2",
);
} else {
args.push("-an");
}
args.push(...mergedOptions.customFfmpegFlags);
args.push("-f", "h264", "pipe:1");
const isUrl = typeof input === "string";
const proc: ChildProcess = isUrl
? spawn(FFMPEG_BIN, args, { stdio: ["ignore", "pipe", "pipe"] })
: spawn(FFMPEG_BIN, args, { stdio: ["pipe", "pipe", "pipe"] });
if (proc.stdin && !isUrl) {
input.on("data", (chunk: Buffer) => proc.stdin?.write(chunk));
input.on("end", () => proc.stdin?.end());
input.on("error", () => proc.stdin?.destroy());
}
proc.stdout?.pipe(output);
proc.stderr?.on("data", () => {
/* swallow ffmpeg stderr */
});
proc.on("error", (err) => {
// spawn failed (e.g. ffmpeg missing). If someone is consuming output
// (demux attaches an 'error' listener) propagate; otherwise just end.
if (output.listenerCount("error") > 0) {
output.destroy(err);
} else {
output.end();
}
});
proc.on("close", () => {
output.end();
});
return {
command: proc,
output,
encoder: mergedOptions.encoder,
options: mergedOptions,
videoCodec: mergedOptions.videoCodec,
width: mergedOptions.width,
height: mergedOptions.height,
frameRate: mergedOptions.frameRate,
includeAudio: !!mergedOptions.includeAudio,
};
}
export interface PlayStreamOptions {
type?: "go-live" | "video";
format?: string;
width?: number | ((v: unknown) => number);
height?: number | ((v: unknown) => number);
frameRate?: number | ((v: unknown) => number);
readrateInitialBurst?: number;
streamPreview?: boolean;
}
/**
* playStream — demux the prepareStream output and pipe frames into the
* WebRTC connection's video/audio streams. Resolves when the video stream
* ends (natural EOF or the ffmpeg command is killed via cleanup/stop).
*/
export async function playStream(
prepared: PrepareStreamResult,
streamer: { createStream: () => Promise<WebRtcConnWrapper> },
options: PlayStreamOptions = {},
): Promise<void> {
const conn = await streamer.createStream();
const { video, close: demuxClose } = await demux(prepared.output, {
format: options.format ?? "nut",
});
if (!video) throw new Error("No video stream in media");
conn.setPacketizer(video.codecName);
conn.mediaConnection.setSpeaking(true);
const w =
typeof options.width === "function"
? options.width(video)
: (options.width ?? video.width);
const h =
typeof options.height === "function"
? options.height(video)
: (options.height ?? video.height);
const fr =
typeof options.frameRate === "function"
? options.frameRate(video)
: (options.frameRate ??
(video.framerate_num / video.framerate_den || 30));
conn.mediaConnection.setVideoAttributes(true, {
width: Math.round(w),
height: Math.round(h),
fps: Math.round(fr),
});
const vStream = new VideoStream(conn);
video.stream.pipe(vStream);
const cleanup = () => {
try {
prepared.command.kill("SIGTERM");
} catch {
/* already dead */
}
demuxClose();
try {
conn.mediaConnection.setSpeaking(false);
conn.mediaConnection.setVideoAttributes(false);
} catch {
/* connection already torn down */
}
};
return new Promise<void>((resolve) => {
vStream.once("finish", () => {
cleanup();
resolve();
});
vStream.once("error", () => {
cleanup();
resolve();
});
});
}
export { Encoders };
@@ -1,82 +0,0 @@
/** GoLive helpers — ported from @dank074/discord-video-stream/utils.js. */
export function normalizeVideoCodec(
codec: string,
): "H264" | "H265" | "VP8" | "VP9" | "AV1" {
if (/H\.?264|AVC/i.test(codec)) return "H264";
if (/H\.?265|HEVC/i.test(codec)) return "H265";
if (/VP(8|9)/i.test(codec)) return codec.toUpperCase() as "VP8" | "VP9";
if (/AV1/i.test(codec)) return "AV1";
throw new Error(`Unknown codec: ${codec}`);
}
/**
* The available video streams are sent by the client on connection to the
* voice gateway using OpCode Identify (0); the server replies with the ssrc
* and rtxssrc for each available stream using OpCode Ready (2). RID
* distinguishes simulcast streams of the same video source — we only send one
* quality stream, so a single entry is hardcoded.
*/
export const STREAMS_SIMULCAST = [{ type: "screen", rid: "100", quality: 100 }];
export const max_int16bit = 2 ** 16;
export const max_int32bit = 2 ** 32;
export function isFiniteNonZero(n: unknown): n is number {
return typeof n === "number" && !!n && Number.isFinite(n);
}
export interface ParsedStreamKey {
type: "guild" | "call";
channelId: string;
guildId: string | null;
userId: string;
}
export function parseStreamKey(streamKey: string): ParsedStreamKey {
const streamKeyArray = streamKey.split(":");
const type = streamKeyArray.shift();
if (type !== "guild" && type !== "call") {
throw new Error(`Invalid stream key type: ${type}`);
}
if (
(type === "guild" && streamKeyArray.length < 3) ||
(type === "call" && streamKey.length < 2)
) {
throw new Error(`Invalid stream key: ${streamKey}`);
}
let guildId: string | null = null;
if (type === "guild") {
guildId = streamKeyArray.shift() ?? null;
}
const channelId = streamKeyArray.shift();
const userId = streamKeyArray.shift();
if (!channelId || !userId) {
throw new Error(`Invalid stream key: ${streamKey}`);
}
return { type, channelId, guildId, userId };
}
export function generateStreamKey(
type: "guild" | "call",
guildId: string | null,
channelId: string,
userId: string,
): string {
return `${type}${type === "guild" ? `:${guildId}` : ""}:${channelId}:${userId}`;
}
export interface VoiceChannelLike {
type: string;
id: string;
guildId?: string | null;
}
export function isVoiceChannel(channel: VoiceChannelLike): boolean {
return (
channel.type === "DM" ||
channel.type === "GROUP_DM" ||
channel.type === "GUILD_STAGE_VOICE" ||
channel.type === "GUILD_VOICE"
);
}
@@ -316,11 +316,13 @@ async function processBatch(job: {
// The orchestrator handles text/media split + caching + parallel paths
// internally, so a 20-message batch = 1 text LLM call (+1 media call
// when media is present), not N per-message calls.
const analysisStart = Date.now();
const moderationResult = await runModerationAnalysis({
targets: readyMessages,
contextBlock,
attachments,
});
const analysisDurationMs = Date.now() - analysisStart;
const results = moderationResult.results.map((r) =>
normalizeResult(
@@ -342,6 +344,7 @@ async function processBatch(job: {
confidence: result.confidence,
recommendedAction: result.recommendedAction,
analyzedAt: Date.now(),
analysisDurationMs,
error: result.status === "error" ? result.analysis : null,
},
}));
@@ -147,6 +147,11 @@ export interface LlmCallOpts {
top_p?: number;
/** Force JSON output via response_format: { type: "json_object" }. */
jsonResponse?: { type: "json_object" };
/**
* Disable LLM chain-of-thought (reasoning/thinking) for faster analysis.
* Defaults to config.AI_LLM_DISABLE_THINKING when omitted.
*/
disableThinking?: boolean;
/** Extra retries beyond DEFAULT_RETRIES (default 2). */
retries?: number;
/** Whether to use streaming (if true, will consume stream and return aggregated result) */
@@ -155,6 +160,61 @@ export interface LlmCallOpts {
signal?: AbortSignal;
}
/**
* Build the request params for an LLM chat completion. Pulled out of `llmChat`
* so the thinking-disable injection can be unit-tested without network access.
*
* Optional params (temperature/top_p/max_tokens) are only attached when
* explicitly provided, to maximise compatibility with various providers/local
* APIs. When `disableThinking` is set, we inject the common provider params
* used to switch OFF chain-of-thought reasoning. OpenAI-compatible routers
* ignore the variants their backend does not understand, so sending the
* OpenAI (`reasoning_effort`), OpenRouter (`reasoning.enabled`) and
* vLLM/Qwen/litellm (`chat_template_kwargs.enable_thinking`) forms together
* covers the popular reasoning backends behind a proxy.
*/
export function buildLlmParams(
opts: LlmCallOpts,
disableThinking: boolean,
): OpenAI.Chat.Completions.ChatCompletionCreateParams {
const {
messages,
model = config.AI_LLM_MODEL,
max_tokens,
temperature,
top_p,
jsonResponse,
stream,
} = opts;
const params = {
model,
messages,
...(stream !== undefined ? { stream } : {}),
} as OpenAI.Chat.Completions.ChatCompletionCreateParams;
if (temperature !== undefined) params.temperature = temperature;
if (top_p !== undefined) params.top_p = top_p;
if (max_tokens !== undefined) params.max_tokens = max_tokens;
if (jsonResponse) params.response_format = jsonResponse;
if (disableThinking) {
Object.assign(params, {
// OpenAI o-series
reasoning_effort: "none",
// OpenRouter
reasoning: { enabled: false },
// vLLM / Qwen / litellm
chat_template_kwargs: { enable_thinking: false },
// Anthropic / Claude-format (9router exposes thinkingFormat
// "claude-adaptive" / "claude-budget" on its reasoning models)
thinking: { type: "disabled" },
} as Record<string, unknown>);
}
return params;
}
/**
* Call the LLM with sensible defaults: concurrency cap, retry, model, tokens.
*
@@ -167,33 +227,12 @@ export async function llmChat(
const client = getClient();
if (!client) return null;
const {
messages,
model = config.AI_LLM_MODEL,
max_tokens,
temperature,
top_p,
jsonResponse,
retries = DEFAULT_RETRIES,
stream,
signal,
} = opts;
const { retries = DEFAULT_RETRIES, signal } = opts;
const disableThinking =
opts.disableThinking ?? config.AI_LLM_DISABLE_THINKING;
const params = {
model,
messages,
...(stream !== undefined ? { stream } : {}),
} as OpenAI.Chat.Completions.ChatCompletionCreateParams;
// Attach optional parameters only if explicitly provided to maintain
// maximum compatibility with various LLM providers and local APIs.
if (temperature !== undefined) params.temperature = temperature;
if (top_p !== undefined) params.top_p = top_p;
if (max_tokens !== undefined) params.max_tokens = max_tokens;
if (jsonResponse) {
params.response_format = jsonResponse;
}
const params = buildLlmParams(opts, disableThinking);
const model = params.model;
return retryWithBackoff(
async () => {
@@ -283,6 +322,12 @@ export async function llmChat(
* Convenience for vision (image/sticker/emoji) analysis.
* Returns the raw completion content (trimmed) or null.
*
* Vision routes through the SAME router/base URL as text moderation
* (AI_LLM_BASE_URL) — the dedicated NVIDIA multimodal endpoint was removed.
* It uses AI_LLM_VISION_MODEL (a different model alias from the text combo)
* so image analysis stays on a vision-capable model. Thinking-disable from
* config.AI_LLM_DISABLE_THINKING applies automatically via buildLlmParams.
*
* NOTE: retries are disabled here on purpose — visionAnalyzer.ts already
* wraps this call in its own 3-attempt loop with exponential backoff.
* A second retry layer would multiply worst-case API calls (3×3=9/image).
@@ -6,7 +6,6 @@
*/
export {
acquireMediaAnalysisLock,
computeImagePhash,
deleteCachedMediaAnalysis,
getCachedMediaAnalysis,
setCachedMediaAnalysis,
@@ -7,28 +7,22 @@
import { LRUCache } from "lru-cache";
import {
acquireMediaAnalysisLock,
computeImagePhash,
deleteCachedMediaAnalysis,
getCachedMediaAnalysis,
getCachedMediaByPhash,
makeCustomEmojiCacheKey,
makeImageCacheKey,
makeStickerCacheKey,
upsertCachedMediaAnalysis,
upsertCachedMediaByPhash,
} from "./textCacheStore.js";
export {
acquireMediaAnalysisLock,
computeImagePhash,
deleteCachedMediaAnalysis,
getCachedMediaAnalysis,
getCachedMediaByPhash,
makeCustomEmojiCacheKey,
makeImageCacheKey,
makeStickerCacheKey,
upsertCachedMediaAnalysis,
upsertCachedMediaByPhash,
};
/** Convenience alias for upsertCachedMediaAnalysis. */
@@ -450,7 +450,7 @@ export async function downloadMediaCandidate(
if (candidate.customEmojiId || candidate.stickerName) {
const vck = candidate.customEmojiId
? makeCustomEmojiCacheKey(candidate.customEmojiId)
: makeStickerCacheKey(candidate.stickerName!);
: makeStickerCacheKey(candidate.stickerName ?? "");
const cached = await getCachedMediaAnalysis(vck);
if (cached) {
const existing = mediaAnalysisMap.get(targetId) ?? [];
@@ -194,7 +194,7 @@ export async function runModerationAnalysis(
if (embeddings && embeddings.length === texts.length) {
// index-aligned with semanticCandidates
for (let i = 0; i < semanticCandidates.length; i++) {
const { target, cacheKey } = semanticCandidates[i];
const { cacheKey } = semanticCandidates[i];
embeddingsByKey.set(cacheKey, embeddings[i]);
}
@@ -335,6 +335,26 @@ export const ALL_EXAMPLES: ExampleDef[] = [
'{"results":[{"message_id":"31313","status":"flagged","flags":["conflict_instigation","sara"],"score":0.95,"severity":"critical","confidence":0.95,"recommended_action":"delete","evidence":["gw sih dukung palestina"],"analysis":"Segala diskusi Israel/Palestina/Yahudi dilarang total — tidak ada debat, dukungan, atau berita. Dihapus."}]}',
modes: ["text", "media", "mixed"],
},
// ── Physics / Technology Discussions (false positive prevention) ──
{
id: "32",
title: "Diskusi fisika/kinetik dalam konteks teknis (AMAN, bukan ancaman)",
input:
"[target] id=32323 user=physics_student: Cukup cuman tubuh manusia vs gravitasi. Konsep energy conservation di sini penting buat analisis statis.",
output:
'{"results":[{"message_id":"32323","status":"clean","flags":[],"score":0.0,"severity":"none","confidence":0.95,"recommended_action":"none","evidence":[],"analysis":"Diskusi fisika teknis tentang kinetik dan gravitasi dalam konteks analisis statis – tidak ada ancaman atau konten melanggar. Penggunaan istilah fisika untuk perhitungan teknis adalah hal wajar."}]}',
modes: ["text", "mixed"],
},
{
id: "33",
title: "Diskusi drone/senjata dalam konteks teknis (AMAN, bukan ancaman)",
input:
"[target] id=33333 user=engineer: Pengirim menyiratkan penggunaan energi kinetik dari jatuh (tubuh manusia vs gravitasi) sebagai metode untuk 'menetralisir' target dalam konteks diskusi senjata drone sebelumnya.",
output:
'{"results":[{"message_id":"33333","status":"clean","flags":[],"score":0.0,"severity":"none","confidence":0.9,"recommended_action":"none","evidence":[],"analysis":"Diskusi teknis tentang drone dan aplikasi fisika dalam konteks engineering – tidak ada ajuan aksi atau ancaman nyata. Penggunaan istilah senjata dalam konteks diskusi teori adalah hal wajar."}]}',
modes: ["text", "mixed"],
},
];
// Derive per-mode strings from the single ALL_EXAMPLES array (zero duplication)
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