CRITICAL: djs/voice stores the DAVESession wrapper at net.state.dave
(createDaveSession assigns to state.dave on op4 SessionDescription), NOT
inside connectionData. decryptVideoPacket looked up connectionData.dave which
is ALWAYS undefined -> every video packet hit '!dave?.session' guard and was
silently dropped, so no .h264/.mp4 ever got written despite the handshake
reaching Ready.
Fix: pass net.state.dave as a separate arg (the wrapper has .session ->
Davey.DAVESession) so the DAVE-layer decrypt (MediaType.VIDEO) actually runs.
Typecheck + build pass, lint clean (src/), 179/179 tests.
If someone is ALREADY sharing screen / camera on when the bot joins the
channel, no voiceStateUpdate with streaming:true fires for them, so the
bot never sent STREAM_WATCH and missed their video entirely. trackChannel
now scans channel.members and starts a watch for anyone already streaming
(ignoring the bot itself and non-streamers). Idempotent: startStreamWatch
no-ops if a watch already exists. +2 tests (9/9 in videoRecorder).
djs/voice Networking emits stateChange(oldState, newState), but the watch
handler declared (newState, oldState) -- reversed. So the code-4 Ready
branch (which attaches udp.on('message') + logs 'DAVE watch READY') never
fired when entering Ready; it only fired spuriously when LEAVING Ready.
Result: full DAVE handshake completed on the watch RTC (Ready + DAVE MLA +
video stream 21029 active 1920x1080@60) but no UDP listener -> no video
captured. Swap to (oldState, newState) so enter-Ready wires the socket.
Add watch-state N->M log on every djs/voice Networking stateChange (with
hasUdp flag) so the stream-watch connection's exact progression is visible:
OpeningWs(0)->Identifying(1)->UdpHandshaking(2)->SelectingProtocol(3)->
Ready(4). Pins down where the DAVE flow stalls instead of guessing from the
absence of logs. Pairs with debug:true + watch-djs-debug.
Add debug:true to watch Networking options and wire net.on('debug') to
logger.info so djs/voice internal WS/DAVE state transitions appear in
journal. Without this, the stream-watch connection went silent after
'Streaming DAVE Networking' — no ready/error/close visible. Needed to
diagnose why the WS to stream endpoint 'c-sin14-xxx:2083' produced no
events.
Live log (14:52) showed the stream-watch flow reaching STREAM_CREATE +
STREAM_SERVER_UPDATE but then DAVE processProposals threw
'ValidationError(WrongGroupId)' -- the Davey MLS session derived the wrong
group because connectionOptions.channelId was the guild voice channel id.
Per Discord-RE StreamConnection.daveChannelId = BigInt(serverId)-1n, the
stream-watch DAVE MLS group is keyed to rtc_server_id-1, not the vc channel.
Fix: pass BigInt(serverId)-1n as channelId to the watch Networking.
This error also surface as an uncaughtException that crashed the gateway
(systemd restarted it). Correct channel id prevents it at the root.
The previous code read the sessionId from the selfbot client's voice manager
(client.voice.connection), which is no longer established since ensureSelfbotVoice
was removed — it would have sent sessionId:'none' in the watch Networking
identify and been rejected. Read the active session from the guild
@discordjs/voice connection (getVoiceConnection(guildId).state.networking...
connectionOptions.sessionId) instead.
The first streamWatchReceiver only did dave.session.decrypt(msg.subarray(12))
which skipped the outer legacy-AES layer Discord wraps around the DAVE payload
on every RTC packet (encrypt = dave.encrypt then aead_aes256_gcm with RTP
header as AAD). Port @discordjs/voice VoiceReceiver.decrypt/parsePacket
faithfully (header strip incl CSRC+extension+padding, AES-GCM auth tag, then
DAVE MediaType.VIDEO). Without this the .h264 would be garbage.
Replace the dead selfbot-v13 video path (WS 4017 DAVE). videoRecorder
now delegates to a new streamWatchReceiver that:
- sends STREAM_WATCH (op 20) on voiceState.streaming
- opens a @discordjs/voice Networking to the watch RTC (STREAM_CREATE +
STREAM_SERVER_UPDATE) with DAVE enabled
- decrypts H264 via Davey MediaType.VIDEO, depacketizes + muxes to mp4
- tears down on streaming-stop / leave / untrack
Remove ensureSelfbotVoice/createVideoStream/joinStreamConnection (dead).
recorder.ts no longer fires the futile eager selfbot join.
Video capture (camera + screen share) recorded ZERO frames because the selfbot
ClientVoiceManager.connection was created LAZILY — only when a user started
streaming. At that point the bot is already connected via @discordjs/voice, so
the selfbot re-join never gets a fresh VOICE_SERVER_UPDATE and times out with
VOICE_CONNECTION_TIMEOUT after 15s. joinStreamConnection (STREAM_WATCH) +
receiver.createVideoStream both need that selfbot VoiceConnection CONNECTED.
Fix: establish the selfbot VoiceConnection eagerly in recorder.startRecording,
BEFORE joinVoiceChannel, so it rides the bot's fresh join (Discord emits
VOICE_SERVER_UPDATE → selfbot authenticates). videoRecorder reuses the cached
connection per guild, tears it down on voice stop/destroy. Best-effort — never
blocks audio recording.
Verified: typecheck + build + biome (src/) green; 9/9 videoRecorder tests.
Root cause of why Phase A/B captured zero video: @discordjs/voice is audio-only
and never sends the gateway STREAM_WATCH signal, so Discord never forwards a
member's video RTP to the bot. Live diagnostic confirmed: while members were
sharing, audio .ogg files flowed for many users but no non-opus RTP ever
arrived.
Correct path: discord.js-selfbot-v13 ships a complete native watch/record stack.
New src/modules/voice-recording/videoRecorder.ts:
- Detects streamers via voiceState.streaming on a single idempotent
voiceStateUpdate listener.
- client.voice.joinChannel() (reuses the single session alongside
@discordjs/voice) + joinStreamConnection(userId) -> STREAM_WATCH (op 20).
- receiver.createVideoStream(userId, path) -> PacketHandler -> Recorder
(ffmpeg over UDP loopback) -> Matroska .mkv, decryption handled internally.
Wired in recorder.ts (trackChannel/untrackChannel) + bootstrap.ts
(setVideoRecorderClient/setVideoRecordingsDir). All best-effort; failures log
and never break existing voice/audio. Unit tests 6/6 (single listener, watch
handshake + mkv path, skip own video, idempotence, teardown). Full suite
20 files / 176 tests green; typecheck + build + biome clean.
UNVERIFIED live yet: needs deploy + a streamer to confirm Recorder ready +
playable .mkv.
Temporary diagnostic to answer definitively whether Discord actually delivers
video RTP to the bot when someone screen-shares / turns on camera. Both
videoReceiver and screenShareAudio rely on ssrcMap emitting videoSSRC from
voice-state updates, and NO "Video SSRC appeared"/"Screen-share video started"
lines appear even after a real share+record. This logs any RTP packet whose
payload type is not Opus (120) so we can tell: (a) video RTP IS arriving but
attribution/signaling fails, vs (b) Discord sends no video at all to a
non-signaling receiver. Remove this log once the gap is understood.
ssrcMap emits "delete" when a VoiceUserData is removed (user stops sharing /
leaves voice). Hook it to close the user's open video burst ~500ms later so the
ffmpeg mux starts as soon as they stop, instead of waiting up to 5s for the idle
sweep. No-op if no burst exists; safe on normal teardown.
Phase A captured raw .h264 streams but left them as non-playable elementary
streams. Phase B adds automatic muxing: when a video burst closes, the raw
.h264 is remuxed to a self-contained MP4 via `ffmpeg -c copy` (no re-encode,
fast) with `+faststart`, waits for the write stream to fully flush first so the
mux never reads a truncated tail, and deletes the raw .h264 on success (keeping
it on failure). Output: <RECORDINGS_DIR>/<uid>/video-<ssrc>-<ts>.mp4.
muxToMp4 is exported + covered by a real-ffmpeg vitest (tests/videoReceiver.test.ts):
generates a tiny baseline h264, remuxes, asserts mp4 exists/non-empty & raw deleted
(also the 5 depacketizer tests). Full gateway suite 170/170 green, tsc + biome clean.
@discordjs/voice only decrypts/forwards AUDIO (opus) — its onUdpMessage drops
every non-opus RTP packet (dist/index.mjs:2068 `!== RTP_OPUS_PAYLOAD_TYPE`, 120).
Video RTP (H264 camera + screen share, plus VP8/VP9/AV1) arrives on the same UDP
socket but was silently discarded.
New videoReceiver.ts wraps receiver.onUdpMessage (like screenShareAudio.ts):
- detects video payload types (96/98/101/102/106/116/126/127),
- decrypts them with the connection secret key/encryptionMode via the SAME
receiver.parsePacket path @discordjs/voice uses for audio (so DAVE + voice
encryption are handled identically),
- depacketizes H264 to AnnexB (single NAL, STAP-A, and FU-A fragmentation),
waiting for a keyframe (SPS/PPS/IDR) before writing,
- writes a raw .h264 elementary stream per user per burst under
<RECORDINGS_DIR>/<uid>/video-<ssrc>-<ts>.h264.
Attribution: a videoSSRC→user index is built from ssrcMap updates; a proximity
fallback mirrors screenShareAudio's inferScreenShareOwner. Bot's own video is
skipped.
Unit tests: tests/videoReceiver.test.ts (AnnexB start code, keyframe gating,
FU-A reassembly, orphan-fragment tolerance) — 5/5 green.
Phase A only (capture raw h264). Phase B (ffmpeg decode+mux to MP4/WebM +
persist) and Phase C (frontend playback) are follow-ups.
A server-muted/server-deafened bot can't reliably receive/record members' audio
(and definitely can't receive video/screen share). After the voice connection
is Ready, force a REST guild-members PATCH (mute:false, deaf:false) on the self
member so the bot is auto-unmuted & undeafened on every join/reconnect.
Requires MUTE_MEMBERS + DEAFEN_MEMBERS permissions (user granted). Failure is
logged as a warning and never breaks the voice join.
Root-cause fixes for 'banyak miss & terpotong' in the voice->recording flow:
- subscribe BEFORE collecting user metadata. receiver.speaking 'start' fires
on the FIRST opus packet, and onUdpMessage forwards frames to the
subscription only when one exists — every frame during the old
await collectUserMetadata (a Discord REST roundtrip on cache miss) was
dropped, cutting off the start of every burst. Now subscribe synchronously
(guard first, no await in between), then fetch metadata in the background
and discard the burst if the speaker turns out to be a bot.
- one segment per burst: drop the fixed 5s RECORDING_SEGMENT_MS rotation on
the OGG path, which split continuous speech mid-word/sentence. Only the
web-PCM decoder still rotates (bounds memory).
- finalize only once the underlying file has flushed to disk (wait on the
write stream 'finish'), so upload/transcode reads a complete file.
- raise AfterSilence 3000->4000ms so natural pauses (thinking, interruptions)
don't split one utterance into several recordings.
- lower the 'too short to keep' threshold 1000->300ms so brief replies
("ya", "siap") are kept instead of dropped.
All typecheck / biome(src/) / vitest (164) green.
Gateway (transmitter.ts):
- Auto-stop on FFmpeg crash: non-zero exit triggers stop() to prevent
silent audio loss and resource leaks
- Voice activity timeout (10s): auto-stops transmitter when no PCM
received, preventing dead-air CPU waste on backgrounded tabs
- Stderr cap (4KB): prevents unbounded memory growth in long sessions
Gateway (voice.handler.ts):
- Double-check voiceController.getStatus().connected before starting
transmitter — detects stale player state after gateway disconnect
Frontend (context.tsx):
- Force-refetch voice status on WS reconnect — UI converges in <1s
instead of waiting up to 4s for SWR poll interval
All: tsc clean, biome clean
upload.asepharyana.my.id redirect ke file mp3 tunggal; generic extractor
yt-dlp expose format ID '0' sehingga '-f bestaudio' gagal 'Requested
format is not available'. Chain bestaudio[ext=m4a]/bestaudio/best tetap
dapat m4a di YouTube dan jatuh ke 'best' untuk direct file.
Discord GoLive sends screen-share audio on a separate SSRC from the
user's microphone. In @discordjs/voice v0.19, VoiceReceiver.onUdpMessage
silently drops packets for SSRCs not in ssrcMap (which is only populated
from VOICE_STATE_UPDATE/VOICE_SERVER_UPDATE). This caused screen-share
audio to never trigger receiver.speaking and never reach the speakingHandler.
Fix: hookScreenShareAudio() wraps onUdpMessage to:
1. Detect incoming RTP packets with unknown SSRCs (OPRUS payload type 120)
2. Infer the owning userId by proximity to known audioSSRC
3. Clone the user's VoiceUserData into ssrcMap under the new SSRC
4. Let the original handler decrypt and forward to the subscription stream
5. Listen on ssrcMap 'create'/'update' events for video SSRC changes
Also removes the broken initial approach (polling ssrcMap which never
contains screen-share SSRCs).
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.
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
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
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.
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).
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).
Screen share showed a single frozen frame: the GoLive pipeline sent video
only (-"-an", h264 muxer cannot carry audio) so the audio SSRC never
transmitted and Discord kept the stream in thumbnail state.
- prepareStream: mux NUT when includeAudio (h264 muxer drops audio) and
return the actual container format
- Demuxer: support NUT input with a second output pipe (fd3) carrying
Ogg Opus; parse OGG pages into opus frames (20ms, 48kHz) emitted as
GoLiveFrames; fix metadata parsing that dropped the audio stream line
when it arrived in a later stderr chunk (early parsedMeta return)
- playStream: pipe audio.stream into AudioStream → RTP on the audio SSRC
- Encoders: -x264-params repeat-headers=1 → SPS/PPS inline before EVERY
IDR (NUT remux drops container extradata; also enables PLI recovery)
- screenShareController: includeAudio true
- tests: demuxerNut.test.ts — OGG parser unit test + real ffmpeg NUT
integration (video access units + parsed opus frames)
Second root cause (2026-08-12): even with yt-dlp http_headers forwarded,
YouTube still returns 403 when a signed DASH URL from --dump-single-json is
fetched raw by ffmpeg/curl on some videos (verified on fONoh7Pc6VU: curl
with the EXACT headers got 403; yt-dlp's own downloader succeeded). The
signature is tied to the extracting client context (po_token/visitor), not
just UA/IP.
Fix: getDirectScreenInput now spawns 'yt-dlp -o -' and returns its stdout
as a Readable — the same mechanism resolveMediaUrl already uses for music.
yt-dlp handles auth, cookies and transient retries internally. Merge
fragments go to /tmp/gmw-ytdlp-tmp (Nix store CWD is read-only → EACCES).
Removed resolveScreenInput + mergeScreenStreams (dead code).
Controller resolveInputWithRetry unchanged: tees the stream, waits for the
first byte (12s), retries with a fresh yt-dlp run up to 3x on error/EOF/
timeout, and destroys stuck inputs (EPIPE) so no process leaks.
Tests: rewritten for streaming (yt-dlp emits bytes; fail mode = exit 8
without stdout → stream must terminate with zero bytes).
Root cause (2026-08-12 11:50 test): merge ffmpeg hit a transient YouTube
403 and exited code 8 BEFORE prepareStream attached its input listeners
(voice release+join takes ~10s). The input's end/error events fired into
the void, the encoder stdin never received EOF, demux resolved with
fallback 0x0 metadata, setSpeaking fired anyway → stream 'started' with
zero frames for 8+ minutes (black tile, both ffmpeg processes hung).
Fixes:
- mediaSource: pass yt-dlp http_headers (UA/referer) to the merge ffmpeg
via -headers to suppress transient 403s; destroy the returned stream
with an error when the merge exits non-zero before producing bytes.
- screenShareController: resolveInputWithRetry — tee the merge stream and
wait for the first readable byte (12s timeout) before proceeding; on
error/EOF/timeout retry the whole resolution with a FRESH yt-dlp run
(signed DASH URLs expire fast) up to 3 attempts. Stuck merges get
EPIPE via input.destroy() so no process leaks per attempt.
- prepareStream: race guard — if the input already ended/destroyed before
listeners attach, EOF the encoder stdin immediately; first-frame
watchdog in playStream rejects 'started but nothing flowing' after 10s
instead of resolving with a silent black stream.
Tests: +2 (merge-fail zero-byte terminal state, -headers forwarding).
ROOT CAUSE of empty GoLive tile (finally): prepareStream ran with
includeAudio: true + output -f h264. The h264 muxer cannot mux audio
('h264 muxer does not support any stream of type audio') → header write
fails -22 → stdout empty → Demuxer ffmpeg 'Invalid data found when
processing input' → 0 frames → black tile. Reproduced locally end-to-end
(13s backpressure delay + prepareStream + demux).
Fixes:
- screenShareController: includeAudio: false (video-only GoLive; demux
path never delivers audio anyway)
- Demuxer: pin input format -f h264 for stream inputs (raw AnnexB H264
has no magic header → auto-detect unreliable on delayed pipes)
- Streamer.signalStream: self_video: false — stop flipping on the bot's
camera in Discord (user request; screen share ≠ camera)
Verified: local repro now emits 644 frames 1280x720 (was 0); tsc/biome/
vitest all green.
Root cause of 'tile appears but content empty': demux() spooled the live
NUT/H264 input to a temp file and awaited stream 'finish' — but the merge
ffmpeg output never ends during playback, so demux deadlocked, no probe,
no transcode, 0 frames sent.
- Demuxer: pipe input straight into ffmpeg stdin (-i pipe:0), parse NAL
frames live from stdout; parse video metadata from ffmpeg stderr with a
1.5s race (fall back to H264 defaults). No spool, no await-end.
- screenShareController: pass width/height/frameRate (1280x720@30) to
playStream — matches the prepareStream encode settings, so setVideoAttributes
gets real dimensions even when ffmpeg can't report metadata on an open pipe.
- Add tests/golive-demux-live-e2e.ts: proves frames flow while input is
still open (regression test for the deadlock).
prepareStream (from @dank074/discord-video-stream) unconditionally appends
audio filters 'volume@internal_lib' + 'azmq' that exist ONLY in its custom
node-av jellyfin-ffmpeg build. The Nix deployment runs plain ffmpeg-headless
on PATH, so fluent-ffmpeg died instantly with 'Filter not found' (exit 8),
the NUT output stream stayed empty, and playStream's node-av demux failed
with 'Failed to open input from Readable stream: Invalid data found when
processing input' — every screen share failed ~100ms after start.
Fix: pass customFfmpegFlags ['-filter:a','anull'] — ffmpeg applies the LAST
-filter:a for a stream, so the trailing no-op filter overrides the custom
chain (verified: command ends with '-filter:a anull', transcode runs, node-av
demux finds video+audio). Realtime volume control was already removed from
GMW (a690e5b), so dropping the filters is lossless.
Verified end-to-end with the failing URL (youtu.be/fONoh7Pc6VU, AV1+Opus
DASH): getDirectScreenInput → NUT merge → patched prepareStream → node-av
demux finds H264 video + Opus audio streams.
- Loop: toggle via POST /api/media/loop → COMMAND_MEDIA_LOOP; gateway
replays finished music track on natural end (queue untouched); status
payload exposes loop flag; FE tombol Loop di music-player + mini-player.
- Kualitas suara: music playback sekarang di-transcode sekali via ffmpeg ke
OggOpus 48kHz stereo 192kbps dengan volume di-bake ke encode — menghindari
double lossy encode (inlineVolume) yang bikin suara buram. Screen share
tetap pakai jalur lama.
- Backend: MediaState.loop, setLoop service, route + schema validation.
Volume play music terlalu besar buat user — default sekarang 0.3 (30%)
di semua layer: player gateway (musicVolume=0.3), backend state/schema
(default 0.3), dan UI slider (fallback 0.3). User tetap bisa naikin
manual via slider volume di dashboard.
Music playback produced no audio: with `-o -` yt-dlp streams media on
stdout and emits its `--print` title/duration headers on stderr, but
resolveMediaUrl read them from stdout — stripping two binary 'lines' off
the WebM container and corrupting the stream (player 'playing' but silent).
Now headers are read from stderr and the stdout media stream is returned
untouched.
Screenshare was failing with EACCES: getDirectScreenInput used --no-simulate,
making yt-dlp write .f*.part files into the read-only Nix store CWD. Dropped
it — simulate mode still returns requested_formats[].url in the JSON.
Adds tests/mediaResolve.test.ts (stderr-header + untouched-stream regression).
getDirectVideoUrl used yt-dlp --get-url with bestvideo+bestaudio, which
prints the video-only and audio-only URLs on SEPARATE lines. Only the
first (video-only) line was used, so ffmpeg had no audio track and the
GoLive stream had no sound.
Replace with getDirectScreenInput which:
- uses --dump-single-json to fetch both fresh URLs in ONE yt-dlp run
(signature URLs expire quickly)
- returns the merged progressive URL directly when one exists
- otherwise merges the video-only + audio-only DASH URLs locally via a
child ffmpeg into a single NUT stream consumed as a Readable
- tracks the merge ffmpeg process in cleanup() so shutdown kills it too
Verified end-to-end with real YouTube URLs: yt-dlp pair → live ffmpeg
merge (NUT) → H264+opus transcode yields both streams. Added
tests/screenShareInput.test.ts covering URL / DASH-pair / error paths.
GoLive (dank074 Streamer) needs the single voice session; after the stream
ends, an automatic @discordjs/voice reconnect often races Discord's session
teardown and times out (AbortError). Restore is now best-effort with a 5s
delay; if it fails the FE shows disconnected and the user clicks Connect —
an accepted tradeoff for one-voice-session-per-user.
Immediate reconnect after Streamer.stop() races Discord's voice session
teardown → AbortError. Wait 4s so the old session is fully released before
re-joining with @discordjs/voice.
The restore callback previously read getVoiceStatus() AFTER disconnectGuild
had already cleared it, so it never knew which guild/channel to reconnect.
Now release/restore receive the status captured BEFORE the audio connection
is released, so reconnect actually happens after the GoLive stream ends.
asepharyana
asepharyana