- Loop: toggle via POST /api/media/loop → COMMAND_MEDIA_LOOP; gateway
replays finished music track on natural end (queue untouched); status
payload exposes loop flag; FE tombol Loop di music-player + mini-player.
- Kualitas suara: music playback sekarang di-transcode sekali via ffmpeg ke
OggOpus 48kHz stereo 192kbps dengan volume di-bake ke encode — menghindari
double lossy encode (inlineVolume) yang bikin suara buram. Screen share
tetap pakai jalur lama.
- Backend: MediaState.loop, setLoop service, route + schema validation.
Volume sudah di-set default 0.3 di gateway (suara kecil saat play), dan
user bisa naikin sendiri di Discord (command media:volume) — jadi kontrol
volume lewat dashboard tak perlu. Hapus:
- BE: POST /api/media/volume route, mediaVolumeSchema, setVolume service
- FE: useMediaVolume hook, mediaApi.volume, slider volume di music-player
+ mini-player, field volume/setVolume di MediaPlayerProvider
Pertahankan COMMAND_MEDIA_VOLUME di gateway (masih dipakai command DC)
dan mic volume (terpisah, tetap di voice page).
Volume play music terlalu besar buat user — default sekarang 0.3 (30%)
di semua layer: player gateway (musicVolume=0.3), backend state/schema
(default 0.3), dan UI slider (fallback 0.3). User tetap bisa naikin
manual via slider volume di dashboard.
Rombak total alur data frontend: dari static-export CSR (tiap browser
fetch sendiri + akumulasi state voice per-tab) jadi server-side rendering.
Frontend (Next.js):
- next.config: output export -> standalone; halaman jadi server components
- server data layer baru src/lib/api/server.ts (GMW_BACKEND_URL, no window)
- dashboard/media/messages/moderation/recordings/voice page -> RSC yang
fetch backend di render-time, seed ke client view (SWR fallbackData)
- hook-hook utama terima initialData -> first paint data server, revalidate
SWR setelahnya, tanpa spinner-blank-load
- messages: guild/channel/tab/selected dibaca dari URL di server, page awal
di-fetch server-side
Shared realtime state (voice) server-authoritative:
- backend src/modules/voice/live-speaker.ts: agregat voice_active_user dari
gateway jadi snapshot authoritatif (single source of truth semua browser)
- GET /api/voice/status kini include activeSpeakers
- WS initial states kirim voice_state snapshot saat connect (late join
langsung dapat state yang sama, bukan daftar kosong)
- useSpeakers seed dari server snapshot + voice_state full-replace +
voice_active_user delta upsert
Deploy:
- flake.nix: frontend package build SSR standalone (server.js wrapper,
GMW_FRONTEND_PORT=4017); proxy nginx template proxy / -> Next server,
/api + /ws tetap ke backend :4001
Music playback produced no audio: with `-o -` yt-dlp streams media on
stdout and emits its `--print` title/duration headers on stderr, but
resolveMediaUrl read them from stdout — stripping two binary 'lines' off
the WebM container and corrupting the stream (player 'playing' but silent).
Now headers are read from stderr and the stdout media stream is returned
untouched.
Screenshare was failing with EACCES: getDirectScreenInput used --no-simulate,
making yt-dlp write .f*.part files into the read-only Nix store CWD. Dropped
it — simulate mode still returns requested_formats[].url in the JSON.
Adds tests/mediaResolve.test.ts (stderr-header + untouched-stream regression).
shutdown() awaited httpServer.close(), which waits for ALL open
connections. A lingering WS/keep-alive socket left the process zombie
forever after an uncaughtException (e.g. pg 'Connection terminated
unexpectedly' to imrnes) — no exit, so systemd Restart=always could
never revive it; /api/guilds returned 502 until manual restart.
Add 10s force-exit timer in shutdown(); clear it on clean completion.
getDirectVideoUrl used yt-dlp --get-url with bestvideo+bestaudio, which
prints the video-only and audio-only URLs on SEPARATE lines. Only the
first (video-only) line was used, so ffmpeg had no audio track and the
GoLive stream had no sound.
Replace with getDirectScreenInput which:
- uses --dump-single-json to fetch both fresh URLs in ONE yt-dlp run
(signature URLs expire quickly)
- returns the merged progressive URL directly when one exists
- otherwise merges the video-only + audio-only DASH URLs locally via a
child ffmpeg into a single NUT stream consumed as a Readable
- tracks the merge ffmpeg process in cleanup() so shutdown kills it too
Verified end-to-end with real YouTube URLs: yt-dlp pair → live ffmpeg
merge (NUT) → H264+opus transcode yields both streams. Added
tests/screenShareInput.test.ts covering URL / DASH-pair / error paths.
Gateway publishes {playing, activeMode, musicVolume, current, queue} but the
backend MediaState interface + normalizeMediaState dropped activeMode, so the
FE's 'Screen share active' / 'Music playing' badge never rendered. Carry it
through so the FE↔BE media contract stays in sync.
GoLive (dank074 Streamer) needs the single voice session; after the stream
ends, an automatic @discordjs/voice reconnect often races Discord's session
teardown and times out (AbortError). Restore is now best-effort with a 5s
delay; if it fails the FE shows disconnected and the user clicks Connect —
an accepted tradeoff for one-voice-session-per-user.
Immediate reconnect after Streamer.stop() races Discord's voice session
teardown → AbortError. Wait 4s so the old session is fully released before
re-joining with @discordjs/voice.
The restore callback previously read getVoiceStatus() AFTER disconnectGuild
had already cleared it, so it never knew which guild/channel to reconnect.
Now release/restore receive the status captured BEFORE the audio connection
is released, so reconnect actually happens after the GoLive stream ends.
The dank074 Streamer creates its own WebRTC voice connection, but Discord
allows only ONE voice session per user. When VoiceController (audio) was
already connected, the Streamer join hung forever (never got
VOICE_SERVER_UPDATE). Now:
1. ScreenShareController takes releaseVoice/restoreVoice callbacks.
2. Before joining, it disconnects the @discordjs audio connection via
VoiceController.disconnectGuild.
3. Streamer joins + streams GoLive.
4. After the stream ends, restoreVoice reconnects the audio connection so
mic/listen keep working.
5. media.handler wires these via a new setVoiceController accessor from
commandHandler; VoiceController is the single source of truth.
Also adds caller-bound timeouts & safe .catch() everywhere so a stream
failure can never become an unhandledRejection again.
Root cause: ScreenShareController created @dank074 Streamer but never called
streamer.joinVoiceChannel() — playStream threw 'Bot is not connected to a
voice channel', and since the code only used .finally() (no .catch), the
rejection became an unhandledRejection that took down the whole gateway
(graceful shutdown triggered, systemd restarted).
Fixes:
1. Resolve active channel + streamer.joinVoiceChannel(channel) before
prepareStream/playStream (dank074 needs its OWN WebRTC voice connection).
2. .catch() on the playStream done promise — log + kill ffmpeg instead of
crashing the process.
3. .catch() on playback.done in media.handler too.
4. stop() now kills ffmpeg AND stops the streamer's voice connection.
Audit voice (kirim/terima/music/screenshare) menemukan 3 masalah:
1. media:queue SILENT no-op — backend publish {source,mode} tapi gateway
handler baca payload.url → selalu 'received without a URL'. Backend
sekarang kirim {url,mode}, gateway terima url ATAU source (robust).
2. Voice connect gagal diam-diam saat user pilih channel tanpa permission
(joinable=false, contoh Music 32/64/128/256k). Backend+gateway sekarang
expose joinable; FE disable channel 'no akses' + empty state.
3. FE tidak kasih feedback saat connect gagal — tambah toast.error dengan
pesan dari backend.
Verified live: @discordjs/voice connect ke Lofi Radio joinable sukses
(VOICE READY, DAVE session OK) — pipeline voice sebenarnya sehat, masalah
utama UX. media:queue fix akan di-verify setelah deploy.
- chatbot-container: remove chatOpen layer — the minimized bubble now
expands straight into the chat panel (single click, no extra button)
- Remove the redundant 'Tanya soal server...' quick-prompt button and
the PanelLeft collapse toggle (one less state to fight)
- Bubble fixed h-[460px], chat panel flex-fills remaining space
- chat-panel: suggestion chips on empty state (suasana server, channel
paling ramai, total pesan, pesan bermasalah) so first-time users can
start with a single click
- Input: roomier padding, more descriptive placeholder, bigger send
button, autoComplete off
- Drop chatOpen/setChatOpen from context (dead state)
Backend:
- New chatbot.tools.ts: 4 tools (get_server_stats, get_top_channels,
get_recent_activity, get_top_flagged) with real DB executors
- chatbot.service: agentic loop — stream:true, parse SSE, execute
tool_calls, feed results back, up to 4 rounds
- controller: resolve userId from X-User-Id header (no-login device
uuid) with auth middleware precedence; history/clear scoped per user
Frontend:
- use-chatbot-user: mint UUID in localStorage, send as X-User-Id
- chatbotApi.send/getHistory/clearHistory accept userId header
- client.ts: apiRequest supports custom headers per call
- provider: history load + send + clear keyed to device user id
- Delete chatbot-canvas.tsx (placeholder face canvas) + its export
- Remove canvas area from chatbot container; chat panel gets the freed
space (248px → 300px) and bubble height drops 440px → 400px
- Chatbot bubble: 220px → 320px wide, 440px tall; proper header with
drag handle; quick-prompt row when chat is closed
- ChatPanel: show ALL history (not last 8), Indonesian placeholder/empty
state/error copy (backend speaks Indonesian), timestamps (id-ID),
clear-history button, typing indicator bubbles, Enter-to-send
- Send active guildId as context so backend answers reference the real
server (serverInsights path in chatbot.service)
- GuildId sync from layout → ChatbotProvider via ChatbotGuildSync
- chatbotApi.send(message, guildId) → POST /api/chat {message, context}
matching BE zod schema (guildId optional)
The Redis media:status key was only rewritten after a command received via
Redis. When the last track ended naturally (AudioPlayer Idle -> advanceQueue
with an empty queue), currentTrackItem was cleared but the status key was not
persisted — so the backend's cached status and the frontend's 10s polling
stayed stuck showing the finished track as 'playing' forever.
Wire a media-status sink (commandHandler provides the real redisPub to
MediaHandler) and re-publish status after auto-advance, so natural track end
updates the UI.
VoiceActivityTimeline was never given a data prop — the Activity tab always
showed an empty Recharts bar chart while the connection tab already had live
speaker state. Replace the dead chart with a live speaker/activity list fed
from the same WebSocket data, so the tab reflects real state instead of
misleading empty bars.
The voice_recordings table has no duration column, but the backend aliased
duration_bytes = size_bytes (file size in bytes) and RecordingCard divided it
by 60 as if it were seconds — a 3MB MP3 rendered as a nonsensical '55924:3'
fake timestamp. Drop the fabricated field and show real file size instead.
Also render edited_content fallback in the analysis search results for
consistency with message cards/detail.
message_updated broadcasts only {id, edited_content, edited_at} (+ reset
ai_* fields), but the frontend replaced the whole cached record, wiping
username/content/channel_id/created_at -> blank cards and the
'the channel_id of undefined' crash on /messages. Merge partials over the
existing record (list + detail), make list-patching channel-filter aware,
show edited content/badge, and fix the message_updated WS type.
Also broadcast type:'edited' + ai reset in message_updated so the live UI
matches the DB update.
Backend:
- GET /api/dashboard/activity?days=1..90 — daily buckets (messages,
flagged, active_users) + hourly distribution last 24h
- clamped days param, reuses existing indexes (idx_messages_created,
ai_status_created)
Frontend:
- ActivityChart: area chart messages+flagged per day, 7/14/30d range
- HourlyActivityChart: 24h bars with peak highlight
- ModerationDonut: clean/flagged/warned/error breakdown with live
summary line (server X% clean)
- Dashboard layout: activity 2/3 + donut 1/3, hourly + top channels
Verified: endpoint returns real data from prod DB (784/1897/8 msgs
per day), tsc clean backend+frontend.
packages/shared dihapus (5802d02), modul pindah ke src/shared/. Update
import placeholder.test.ts (gateway + backend) ke path lokal + aktifkan
tests/ di backend vitest config dengan alias @. Sebelumnya vitest run
gateway selalu gagal; sekarang 60+36 tests pass.
- recording-card: kartu aktif di-highlight (ring primary + glow pulse),
badge 'Now Playing'/'Loading'/'Paused', tombol play berubah jadi Pause
saat playing dan spinner saat loading, waveform equalizer beranimasi
(animate-eq, delay per bar) saat playing / pulse saat loading.
- recording-player: jadi now-playing panel — tombol play/pause + spinner
loading, progress bar + waktu (current/duration), status 'loading…',
audio element pindah ke sini + event onPlay/onPause/onWaiting/onCanPlay/
onPlaying/onError naik ke page.
- recordings/page: state isPlaying/isLoadingAudio + audioRef, togglePlay
(klik card lain = ganti track, klik card sama = pause/resume).
- globals.css: keyframes eq-bounce + card-glow.
Verified: FE tsc0, next build 10/10 static pages.
Voice page (connection tab):
- ListenControl baru: toggle Listen (Headphones) — mulai PcmPlayer dari
user gesture, subscribe onPcm WS, volume slider, bar level per-user
REAL dari PCM (bukan random).
- lib/audio/pcm-player.ts (baru): ScriptProcessorNode mixer — ring buffer
2s per user (hash FNV-1a sama dengan gateway), upsampling 24k→48k
linear, mix semua user ke mono, gain volume, cleanup ring diam 5s.
- useVoiceListen + hashUserId di hooks; auto-stop saat disconnect.
Recordings:
- recording-player: reset src+load+play() eksplisit (bukan autoPlay doang),
tampilkan filename + error state 'playback failed' kalau file rusak.
- recording-card: tombol Download fetch blob (CORS tele open) → objectURL
→ force download dengan nama asli; fallback buka tab baru kalau fetch
gagal; spinner saat mendownload.
Verified: FE tsc 0, next build 10/10 static pages.
Audit lanjutan: 6x 'LLM API request failed: Request was aborted' per jam.
Root cause: 9router/omniroute SELALU balas SSE (data: chunks) walau request
tanpa stream:true — SDK OpenAI non-stream menunggu FULL body sebelum parse,
jadi batch moderasi besar yang upstream-nya lambat kena timeout 30-60s dan
di-abort. llmClient sudah punya agregasi streaming (chunks → ChatCompletion).
Fix: stream:true di llmCaller (moderasi batch/individual), llmVision,
cultureLearner, userProfileLearner. Verified: SDK stream test 806ms vs
sebelumnya abort. Caller lain (recovery worker dll) lewat llmCaller sama.
Audit log produksi (sejak deploy13:38) menemukan 3 isu:
1. mediaDownloader.ts spawn /usr/bin/ffprobe + /usr/bin/ffmpeg (path keras) —
ENOENT di Nix karena binary cuma di ffmpeg-headless closure. Pakai
PATH-resolved ('ffprobe'/'ffmpeg') seperti voice-recording module
(ffmpegProcess.ts/transmitter.ts) — 5 media warning hilang.
2. individualFallbackProcessor log error 'Success' di level50 tiap fallback
BERHASIL (logModerationError dengan new Error('Success')) — ganti
logger.info dengan verdict yang sama; error log cuma untuk error asli.
3. moderationResponseParser: strip frasa penutup generik ('Tidak ada
indikasi pelanggaran.') yang masih sering dikeluarkan LLM walau prompt
melarang (277/1486 analisis mengandung frasa, termasuk hari ini).
sanitizeGenericCleanCloser hanya mencocok frasa di AKHIR, teks substantif
tetap utuh. Unit test: 6/6 pass.
Migration 0011 (add voice_recordings.transcription) when=1781388000000
lebih kecil dari 0010 (1781390000000) yang sudah ter-apply — drizzle
skip diam-diam (folderMillis <= max(created_at)), kolom transcription
tidak pernah dibuat. Recording OGG sukses tapi INSERT voice_recordings
gagal 42703 di produksi.
Fix: when=1785600000000 (> max applied 1785551832190) + apply manual
ALTER TABLE + insert row __drizzle_migrations dengan hash file yang
sama (c368acb0...) supaya gateway restart berikutnya skip (idempotent).
pnpm 11.17 mengabaikan field pnpm.onlyBuiltDependencies di package.json.
Native deps voice (@discordjs/opus, @lng2004/node-datachannel, zeromq, dll)
tidak pernah kebangun di Nix store karena flake pnpmInstall pakai
--ignore-scripts dan pnpm rebuild tanpa approval. Hasil: receiver/rekaman/
GoLive diam-diam tanpa decoder/encoder native.
pnpm approve-builds --all menulis allowBuilds:true per package di
pnpm-workspace.yaml (harus tracked — flake source cuma ikut file git).
node-crc tetap gagal build (MSRV cargo:: check) tapi tidak pernah
di-import di source — harmless.
asepharyana