Server:
- Extract image width/height on upload via image.DecodeConfig (header-only)
- Add width/height columns to attachments table (migration 007)
- Include dimensions in AttachmentInfo JSON (optional, backward-compatible)
Client:
- Reserve exact space for images using server-provided dimensions
- Fallback min-height: 200px for external/old images, cleared on load
- Remove premeasureAll() — was caching wrong heights for unloaded images
- Smart per-type height estimates: 32px dividers, 42/72px text, +220px
per image attachment, +320px for YouTube embeds
- Batch ResizeObserver corrections to single RAF with anchor-based scroll
preservation (topmost visible item stays in place)
- Fenwick tree for O(log n) offset lookups (replaces O(n) linear scans)
- CSS contain: layout style on .msg-image to isolate layout shift
Server:
- Fix YAML injection in LiveKit config generation (quote values)
- Revert token TTL to 4h (no server-side JWT revocation)
- Derive LiveKit publish permissions from user role (prevent SFU bypass)
- Add CAS guard for webhook/voice_leave race condition
- Add voice_leave broadcast to rollbackVoiceJoin (prevent ghost state)
- Limit webhook body to 64KB (prevent memory abuse)
- Add rate limit to voice_token_refresh handler (1/60s)
- Add LiveKit health check endpoint (GET /api/v1/livekit/health, 503 on degraded)
- Add voice_token_refresh WS handler for client-initiated token refresh
- Consolidate voice quality constants (single source of truth)
- Fix video limit TOCTOU race (count from DB instead of LiveKit API)
- Raise default voice_max_video from 10 to 25 (Discord parity)
- Add CountActiveCameras DB query
- Non-blocking broadcast send, circuit breaker, exponential backoff
- Close send channel before context cancel in serve.go
- Guard voice mute/deafen for active channel
- Delete orphaned message on attachment link failure
- Redact query string from proxy logs (prevent token leak)
- Use instance-level HTTP client for health checks (no redirect following)
- Set cmd.WaitDelay to prevent goroutine leak on Windows
- Log buildJSON marshal errors
Client:
- Refactor livekitSession.ts from singleton module to LiveKitSession class
- Share single AudioContext for all analysers (was 1 per participant)
- Extract createRoom() helper (DRY)
- Add token refresh timer (3.5h interval, re-arms on failure)
- Skip setSpeakers if unchanged (sort in-place, no allocations)
- Distinguish user-initiated leave from connection error in retry
- Add YouTube videoId validation (prevent iframe src injection)
- Add try/finally to disableCamera
- Wrap store subscription callbacks in try/catch
- Track and cancel initial scroll RAF on cleanup
- Add 5s timeout + encodeURIComponent to YouTube oEmbed fetch
- Clean raw mic stream on RNNoise suppressor failure
- Full voice cleanup on logout via cleanupAll()
Tests:
- Add 7 new server tests (webhook parsing, voice guards, quality fallback)
- Fix 2 pre-existing test failures (mute/deafen invalid payload)
Architecture:
- Add subscribeSelector to store.ts for selective state subscriptions
- Create reconcileList utility for DOM list patching without rebuild
- Create shared createContextMenu utility (dedup 3 files)
- Convert all 20 subscribe() calls to subscribeSelector across 11 files
- Split renderers.ts (1131L) into 7 focused files by concern
- Split ConnectPage.ts (838L) into ServerPanel + LoginForm + shell
- Fix ineffective (s) => s selector in ChannelSidebar
GIF visibility:
- Add media-visibility.ts with IntersectionObserver + canvas snapshots
- GIFs auto-pause after 10s, play/pause button overlay on hover
- Freeze GIFs on scroll-away, window blur, and minimize
- Wire into media.ts, attachments.ts, embeds.ts renderers
Tests: 46 new tests (1073 total), all passing
Net: -1166 lines across client codebase
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
- 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
- Add reverse proxy at /livekit/* that forwards to LiveKit server
- Server sends relative URL "/livekit" in voice_token; client
resolves to wss://server:port/livekit using known server host
- Fix API secret minimum length (32 chars required by LiveKit)
- Pass TLS config to LiveKit process manager for TURN certs
- livekit_webhook.go: HTTP handler for LiveKit participant_joined/left
events with JWT verification; syncs stale voice state on crash recovery
- Mount webhook at POST /api/v1/livekit/webhook in router
- Speaker detection handled client-side via LiveKit SDK events
- Replace VoiceConfig STUN/TURN/MediaPort fields with LiveKit
API key, secret, URL, and binary path
- Add livekit_process.go: companion process manager with
auto-restart and config generation
- Add livekit.go: token generation, room service client,
participant management, video track counting
- Add LiveKit server SDK dependency (server-sdk-go/v2)
- Update config tests for new VoiceConfig fields
- 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
Audio and video tracks from the same user were both using stream ID "user-{id}",
causing the client's duplicate stream check to silently drop the video track.
Changed to "user-{id}-audio" and "user-{id}-video" so each gets its own stream.
- setupOnTrack now handles both audio and video tracks via kind branch
instead of returning early on non-audio tracks
- RTP forwarding works identically for both kinds; speaker detection
only runs for audio tracks
- handleVoiceCamera enforces MaxVideo limit before DB update, rejecting
with VIDEO_LIMIT error when the cap is reached
- Added TestHandleVoiceCamera_MaxVideoEnforced test
- CRIT-2: Prevent HTTP header injection in Content-Disposition via mime.FormatMediaType
- HIGH-1: Call hub.GracefulStop() on server shutdown to clean up WebSocket/voice state
- HIGH-2: Remove double send on serveErr channel that caused goroutine leak
- HIGH-3: Handle GetUserByID error after registration to prevent nil panic
- HIGH-4: Return nil,nil on sql.ErrNoRows in GetAttachmentByID (matching codebase convention)
- HIGH-5: Add voiceDone channel to bound RTP goroutine lifetime on voice teardown
- Fix handleVoiceICE to return VOICE_ERROR when no PC (consistent with other voice handlers)
- Update tests to match corrected behavior
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).
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
When switching voice channels, the old PC's OnICEConnectionStateChange(closed)
fires asynchronously after the new PC is set. Both server and client had the
same race: the stale callback saw "voice is active" and called handleVoiceLeave,
closing the new session.
Server: setupICEMonitor now compares the PC reference before acting on events
Client: joinVoice now calls leaveVoice(false) to clean up old session first
Add comprehensive tests for db, storage, updater, and ws packages
covering edge cases, error paths, and voice handler functions.
New test files for attachment queries and WebSocket coverage boost.
- Exclude voiceSession.ts from coverage (browser API dependency, same
pattern as audio.ts/vad.ts/webrtc.ts)
- Remove stale TestHub_Register_CleansUpOldVoiceState test that tested
old duplicate-login behavior removed in b53c729
- Server: handle ICEConnectionStateClosed in setupICEMonitor to clean up
phantom participants when client PC is destroyed
- Server: skip TURN config when turn_secret is empty (suppresses noisy
"password is empty" errors)
- Client: voiceSession.leaveVoice() now sends voice_leave to server by
default, fixing the case where WebRTC failure triggers local cleanup
but server never learns the user left
- Client: explicit leave paths (UI button, logout, beforeunload) pass
sendWs=false to avoid double-sending voice_leave
Rewrites setupOnTrack to create TrackLocalStaticRTP for fan-out,
store tracks on VoiceRoom, add them to other participants' PCs,
and forward RTP while parsing audio levels. Updates handleVoiceJoin
to subscribe new joiners to existing tracks and handleVoiceLeave
to remove departing users' tracks from all subscribers.
Adds Hub.renegotiateParticipant to ws/voice_handlers.go. The function
creates a new SDP offer for a client's PeerConnection and sends it as
voice_offer, implementing the impolite side of Perfect Negotiation by
skipping renegotiation when the PC is in have-remote-offer state.
Add two new server-to-client WebSocket message builders following the
existing buildVoiceAnswer pattern. Expose them via export_test.go
wrappers and cover with ws_test package tests (TDD: RED → GREEN).
Introduces VoiceTrack (Remote/Local track pair + per-subscriber RTPSender map)
and five thread-safe methods on VoiceRoom (SetTrack, RemoveTrack, GetTracks,
TrackUserIDs, GetTrack) needed for SFU audio forwarding in Phase 3.
Close() now also resets the tracks map. Five new tests cover all paths (TDD).
Server now checks IsUserConnected before accepting a new WebSocket and
returns an auth_error with a clear message instead of silently replacing
the old session. Client dispatcher surfaces the error via transient UI
state so the ConnectPage can display it.
- Add cert mismatch modal for TOFU certificate pinning
- Separate voice channels from text in sidebar with user lists
- Implement scrollToMessage and jump-to-pinned-message in overlay
- Add server profiles with credential auto-fill on connect page
- Fix credential auto-fill race condition on rapid profile clicks
- Add channel_focus event for channel-scoped message delivery
- Fix member list case-insensitive role filtering
- Server normalizes role names to lowercase for protocol consistency
- Remove redundant permission-denied log in handleChannelFocus
- Voice store: bulk set states from ready payload, leave cleanup
- WebSocket reconnect and structured logging improvements
- Add tests for cert modal, overlay managers, voice sidebar,
message list scroll, quick switcher, and profile management
- Fix VoiceWidget disconnect not sending voice_leave to server
- Add voice cleanup on logout (send voice_leave before ws.disconnect)
- Add beforeunload handler for best-effort voice_leave on app close
- Broadcast voice_state/voice_leave to ALL clients (not just channel
members) so every sidebar updates when users join/leave voice
- Fix CleanupVoiceForChannel using BroadcastToChannel instead of
BroadcastToAll
- Remove user list from VoiceWidget (users only shown in sidebar)
- Guard VoiceWidget disconnect against double-click
- Add 6 tests for voice disconnect behavior
- #16: Fix golangci-lint issues (unchecked Close(), unused funcs, naming)
- #17: Add KeybindsTab and LogsTab unit tests (19 tests, 81%+ coverage)
- #18: Add rate limiting to chat_edit and chat_delete handlers
- #19: Fix cert mismatch handling via event listener instead of string match
- #20: Validate SHA-256 colon-hex fingerprint format in Rust ws_proxy
- #21: Optimize session+ban check with single JOIN query
- #22: Sort channels by position when redirecting after deletion
- #23: Rename admin test files for clarity
Also fixes ban expiry regression (H-1 from code review) by using
auth.IsEffectivelyBanned() to properly respect temporary ban expiry.
Fix 70+ errcheck violations by adding explicit error discards (`_ =`)
for unchecked return values across test helpers and deferred Close()
calls. Remove unused `senderID` field from broadcastMsg and unused
`defaultCleanupMaxWindow` const. Apply De Morgan's law, remove empty
branch, and simplify redundant type declaration per staticcheck.
- Fix double-close panic in Hub.Stop/GracefulStop using sync.Once (#3)
- Bump golangci-lint action to v9 with v2.11.3 for Go 1.25 support (#4)
- Add input validation guards to SearchMessages (#5)
- Handle promise rejections in InviteManager with error toasts (#6)
- Add missing reply_to and edited_at columns to admin test schema (#7)
- Add ClientCount to HubBroadcaster interface and wire into stats endpoint (#8)
Server:
- Add message search and pinned messages support
- Add admin hub integration and live connection stats
- Update admin test mocks for hub interface
Client:
- Add data-testid attributes to components for E2E testing
- Add window management capabilities (position, size, maximize)
- Add prod E2E test config and script
- Fix CSS imports (use vite bundling instead of HTML link tags)
- Add inline styles to InviteManager overlay for reliability
- Update CHATSERVER.md references from WPF to Tauri
Docs:
- Update quick-start guide
Server fixes:
- Move ATTACH_FILES permission check before CreateMessage to prevent
orphaned messages on permission denial
- Fix hardcoded /api/files/ URL to /api/v1/files/ per spec
- Add error logging for GetAttachmentsByMessageIDs failure
- Set 1MB WebSocket read limit to match client-side limit
- Extract requireChannelPerm helper, replacing 8 repeated patterns
Client fixes:
- Wire onUnauthorized callback to clear auth on 401 responses
- Store auth token in authStore before WS connect
- Reset WS state to disconnected when Tauri APIs unavailable
- Add connectivity guard and 200ms send debounce on message send
- Add toast container to MainPage with error feedback on 5 API failures
- Clear voice currentChannelId on server-driven voice_leave for current user
- Apply stored theme/font/compact preferences at app startup
- Fix infinite scroll throttle to use store subscription instead of fixed timer
Tests:
- Add TestChatSend_AttachmentsDeniedNoMessageCreated
- Add attachments table to handler test schema
- REST authorization: 6 tests verifying READ_MESSAGES enforcement
on GET /channels, GET /channels/{id}/messages, and GET /search
with channel override deny and admin bypass
- WS authorization: 4 tests verifying channel_focus and chat_send
permission checks with deny overrides and admin bypass
- Contract tests: 3 tests asserting response shapes match API.md
(message fields, user object, attachments, reactions with me flag,
search result fields)
Closes test gaps identified in CODE_REVIEW.md.
- Add attachment_queries.go with GetAttachmentByID, LinkAttachmentsToMessage,
and GetAttachmentsByMessageIDs
- Wire attachment linking in handleChatSend with ATTACH_FILES permission check
- Include linked attachments in chat_message WS broadcast payload
- Wire attachment batch-fetch into GetMessagesForAPI for REST responses
- Add attachments table to channel handler test schema
- Critical #1: Add READ_MESSAGES permission checks to channel_focus,
GET /channels, GET /messages, and GET /search
- Critical #2: Send type "auth_error" instead of "error" with AUTH_ERROR
code, preventing infinite client reconnect loops
- Critical #3: Replace role_id (number) with role (string name) in
member_join, auth_ok, and ready payloads via JOIN on roles table
- Critical #4: Always include attachments field (empty array) in
chat_message broadcasts to prevent client crash
- High #2: Add /api/v1/health endpoint alongside /health
- Medium #1: Handle ping WS messages with pong response
- Redesign ConnectPage with modern dark theme, profile cards with delete buttons, login/register toggle
- Add DPAPI-encrypted password saving with "Remember my password" checkbox
- Fix permission bit constants to match SCHEMA.md (Member role 0x663)
- Add migration 004 to fix existing Member role permissions
- Add comprehensive audit logging across all server packages (auth, admin, ws, setup)
- Add member_join WebSocket broadcast so new users appear in members list in real-time
- Add host URL normalization (strip scheme prefix) for reverse proxy compatibility
- Add REST API client, ChatService orchestrator, WebSocket service with reconnection
- Add model types (WsEnvelope payloads, API responses), converters, tests
jevb