- Add --autoplay-policy=no-user-gesture-required to WebView2 config
so remote participants' audio plays immediately on join (desktop app
doesn't need browser autoplay restrictions)
- Add AudioPlaybackStatusChanged handler with click-to-unlock fallback
for browsers that still block autoplay
- Replace broken VAD implementation that used setMicrophoneEnabled/
mediaStreamTrack.enabled (both fought LiveKit's track lifecycle) with
a unified GainNode audio pipeline:
rawMic → AnalyserNode (VAD) → GainNode (volume × gate) → sender
- VAD now gates by setting gain=0 instead of touching the track —
analyser always sees real audio, no stale track references
- Merge input volume and VAD into single pipeline (always active)
- Add voice settings UI: draggable sensitivity threshold on mic meter,
input/output volume sliders (0-200%), audio processing toggles
Co-Authored-By: claude-flow <ruv@ruv.net>
- Detach existing audio elements before attaching to prevent double playback on reconnects
- Remove webAudioMix to eliminate Web Audio overhead compounding with multiple participants
- Use participant.setVolume() for full 0-200% per-user volume range
- Add pli_throttle and active_loopback_prevention to LiveKit server config
- Bump version to 1.3.0
Camera fixes:
- Optimistic setLocalCamera(true) before async setCameraEnabled for instant
button highlight, with revert on failure
- Only call checkVideoMode when camera-relevant state changes, not on every
speaking poll tick (100ms)
- VideoGrid.addStream updates existing cells in place instead of
destroy+recreate to prevent black frame flashes
- VideoModeController tracks localTileAdded to avoid redundant addStream calls
Security & hardening (from prior session review):
- Settings store key allowlist prevents arbitrary key writes
- Certificate fingerprint validates SHA-256 colon-hex format
- CredentialData Debug impl redacts token and password
- LiveKit config file written with 0600 permissions
- Token TTL reduced from 24h to 4h
- Null-check on client.user before token generation
- Thread-safe getChannelID/trySendMsg helpers on Client
- Warn on default dev LiveKit credentials
- Devtools feature-gated behind cfg(feature = "devtools")
- Updater uses configure_client for self-signed cert acceptance
- Clamp voice sensitivity input to 0-100 range
- Clear lastConnectToken/Host on logout
- Fix animation frame leak in VoiceAudioTab mic meter
Speaking detection:
- Replace unreliable ActiveSpeakersChanged with per-participant
IsSpeakingChanged events (fires locally, more responsive)
- Wire speaking detection on local + remote participants
- setSpeakers updates all users including local (LiveKit is sole
authority, no local VAD)
DevTools:
- Add F12/Ctrl+Shift+I shortcut to open WebView2 DevTools
- Enable devtools feature in Tauri release builds
- Add open_devtools Rust command
CSP:
- Allow http://ipc.localhost for Tauri IPC protocol
- Allow http/ws://localhost:* for LiveKit signal connection
Connection:
- Fix LiveKit API secret minimum 32 chars
- Remove TURN TLS config that crashed LiveKit server
- Add reverse proxy at /livekit/* (kept for future use)
- Add http://localhost:* and ws://localhost:* to CSP connect-src
(WebView2 was blocking LiveKit signal connection)
- Add connection retry (3 attempts, 2s delay) for LiveKit server
startup race condition ("could not find any available nodes")
- Remove broken TURN TLS config from generated livekit.yaml
- Send direct LiveKit URL instead of proxy path (localhost is
treated as secure context in Chromium/WebView2)
- Set LiveKit server host from API config for URL resolution
- Fix SDP signaling race condition: add per-client negoMu to serialize
renegotiateParticipant / handleVoiceOffer / handleVoiceAnswer so
concurrent OnTrack goroutines don't race through rollback
- Fix handleVoiceLeave triple-fire: early return when clearVoice()
returns zeros so ICE callbacks don't re-enter and corrupt state
- Fix SQLite SQLITE_BUSY errors: add busy_timeout=5000 pragma and
SetMaxOpenConns(1) for file-based databases
- Fix deafen bypass: new remote audio elements now respect localDeafened
state so late-arriving streams are muted immediately
- Add debug-level logging for SDP negotiation, track fan-out, ICE
candidates, voice state changes, room lifecycle, and participant
add/remove
- Bump version to 1.1.1
- Add createUpdaterArtifacts: "v2Compatible" to bundle config
so CI generates .nsis.zip + .nsis.zip.sig for auto-updates
- Bump client version from 0.1.0 to 1.0.0
Voice was broken over NAT due to multiple issues across the audio pipeline:
- Fix GainNode silence: WebView2 silences remote WebRTC streams routed through
Web Audio createMediaStreamSource→GainNode→createMediaStreamDestination.
Replaced with direct HTMLAudioElement playback for remote audio.
- Fix NAT traversal: Add Google public STUN server (stun.l.google.com:19302)
so remote clients can discover their public IP for ICE connectivity.
- Fix signaling race: Catch createOffer InvalidStateError when server
renegotiation offer arrives before client's initial offer is sent.
- Fix device switch: Use replaceTrack() instead of removeTrack+addTrack
to avoid SDP renegotiation. Safe rollback on failure (stop old last).
- Fix speaking flicker: setSpeakers skips local user (VAD is sole authority).
- Fix VAD sample rate: Force 48kHz AudioContext instead of system default
(192kHz) which spread FFT bins too wide for voice frequency detection.
- Fix CSP for WASM: Add wasm-unsafe-eval to script-src for RNNoise.
- Fix clearAuth leak: leaveVoice() called before resetVoiceStore().
- Fix ICE rate limit: Separate limit for ICE candidates (50/s vs 20/s).
- Fix stale ICE errors: Silently drop voice_ice with no PeerConnection.
- Fix audio play() race: Deferred to queueMicrotask after DOM attachment.
Debugging infrastructure:
- Logs tab: Copy All button, Voice Diagnostics panel with live session
state, Probe Audio Levels (measures actual signal at 3 pipeline points),
Test Direct Playback button, Copy Diagnostics button.
- Client logging: WebRTC (PeerConnection lifecycle, ICE candidates with
type/address, track events, negotiation), VAD (start/threshold/destroy),
Audio (device acquisition with settings, device changes), noise suppression
(WASM load timing, worklet vs fallback path), voiceSession (remote stream
parsing failures, deafen state, audio element playback events).
- Server logging: SFU init config, voice room mode transitions/track
lifecycle/close, RTP forwarding with packet counts and first-packet
detection, 5s no-packet warning, track fan-out counts, subscriber
transceiver state, ICE candidate details, voice credentials issued.
- Logger: Error objects now serialize .message and .stack instead of {}.
Per-user volume right-click now works on voice user rows in sidebar.
RNNoise ML noise suppression with AudioWorklet + ScriptProcessor fallback.
Tests: 7 new test cases (replaceTrack, setSpeakers skip-local, clearAuth).
- Add tauri-plugin-updater and tauri-plugin-process for in-app updates
- Rust commands (check_client_update, download_and_install_update) build
updater with dynamic endpoint at runtime for self-hosted compatibility
- Server endpoint GET /api/v1/client-update/{target}/{version} translates
GitHub Releases into Tauri updater JSON format with .sig content
- UpdateNotifier banner component with install/dismiss controls
- CI workflow produces signed .nsis.zip + .sig updater artifacts
- Self-signed TLS support via dangerousAcceptInvalidCerts config
Server:
- Add POST /api/v1/uploads and GET /api/v1/files/{id} endpoints
- Add CreateAttachment DB method for file upload records
- Allow empty message content when attachments are present
- CORS headers on file serving for WebView2 compatibility
Client — File uploads & attachments:
- Clipboard paste (Ctrl+V) and attach button (+) for file uploads
- Preview bar above input with thumbnail, spinner, and remove button
- Images fetched via Tauri HTTP plugin as base64 data URIs (bypasses
self-signed cert rejection in WebView2)
- Three-layer image cache: memory → IndexedDB → network
- In-flight deduplication prevents duplicate concurrent fetches
- Image lightbox with click-to-zoom, scroll wheel zoom, pan, keyboard shortcuts
Client — URL previews & embeds:
- URLs in messages rendered as clickable links
- YouTube embeds with thumbnail, play button, video title via oEmbed API
- Generic link previews with OG metadata (title, description, image)
- Fetched via Tauri HTTP plugin with Facebook crawler User-Agent
- YouTube title cache and OG metadata cache prevent re-fetch on re-render
- Links open in default browser via tauri-plugin-opener
Client — Voice & audio fixes:
- Mute uses replaceTrack(null) for reliable RTP-level muting in WebView2
- Deafen also mutes mic; undeafen/unmute unmutes both
- Muted users show crossed mic icon, deafened show crossed mic + headphone
- Re-apply mute state after input device switch
Client — UX improvements:
- Disable browser context menu globally (only custom menus show)
- Emoji search now matches by keyword names (smile, heart, fire, etc.)
- Emoji picker closes on click outside
- User bar status text moved below username
- Messages sorted chronologically (oldest first, newest at bottom)
- Scroll to bottom on initial load with deferred retries for layout shifts
- Image attachments constrained to 400x350px with click-to-lightbox
jevb