- docker-compose: use external volume/network with backspace naming
- deploy.sh: add --remote/-r flag for off-network deploys, update remote path
- Dockerfile: update DB_PATH to backspace.db
Add a server-side instance_settings table (single-row, CHECK(id=1))
that stores admin-configurable streaming bounds: bitrate min/max/step,
allowed resolutions, and allowed framerates.
Backend:
- New instance_settings schema + migrations (is_admin on users, default
settings row, first-registered-user promoted to admin)
- GET/PATCH /api/settings/streaming endpoints with admin-only writes
and full input validation including cross-field checks
Frontend:
- settingsStore fetches limits on WebSocket ready, exposes isAdmin flag
- ScreenShareSettingsPopover reads bounds from store instead of
hardcoded constants, auto-clamps stale localStorage values
- buildScreenShareOptions() clamps bitrate to server limits at build
time as enforcement backstop
- ServerSettings modal gains a "Streaming" tab (admin-only) with
bitrate range sliders, resolution/framerate toggles, and save/reset
When a viewer joined while a screen share was active, autoSubscribe:true
caused a subscribe-then-unsubscribe dance for screen share tracks,
triggering cascading renegotiations, MaxListeners warnings, and
negotiation timeouts. Now the SFU starts with no subscriptions and we
explicitly subscribe only to non-screen-share tracks (audio, camera).
Screen shares remain controlled by the watch/unwatch UI flow.
Camera was stuck at 180p because our custom <video> rendering bypassed
LiveKit's adaptive stream observer. Replaced manual srcObject binding
with track.attach()/detach() in VoiceUser, StreamTile, and PictureInPicture
so the SFU receives viewport dimensions and forwards the correct
H.264 simulcast layer.
Screen share VP9 SVC with L3T3 spatial layers failed because hardware
VP9 encoders (NVENC, QSV, VCE) don't support spatial scalability —
Chrome silently degrades to L1T1. Reverted to VP9 single-layer
(simulcast: false, no scalabilityMode). Also targets encodings[length-1]
in applyOverdrive() for correct simulcast layer targeting.
- Add customBitrateKbps to ScreenShareConfig (null = auto matrix lookup)
- Slider in Stream Settings popover: 500 kbps–20 Mbps, step 500 kbps
- "Reset to Auto" clears override back to preset-derived bitrate
- Live updates via existing applyOverdrive() pipeline on active streams
- Persist version 5 → 6 with migration
- Add MarkdownRenderer with syntax-highlighted code blocks (prism-react-renderer),
GFM tables, and Discord-style theming
- Add MentionBadge that resolves user IDs to display names with role colors
- Add MentionPopover autocomplete triggered by @ in MessageInput
- Fix mention:// URL sanitization — allowlist the scheme in urlTransform so
react-markdown v9 passes it through to the component override
- Highlight messages that mention the current user (amber border)
- Bump rate limit from 60 to 200 req/min
Remove the legacy TEXT role column ('owner'/'admin'/'member') from
server_members and make the bitwise RBAC member_roles junction table
the sole authority for role assignments. Owner detection now uses
servers.ownerId exclusively.
- Remove MemberRole type and role field from shared types
- Remove role from Drizzle schema, raw SQL CREATE TABLE, and seed data
- Rewrite PATCH /members/:uid to accept { roleIds: string[] }
- Fix GET /members to populate roles array (was TODO)
- Replace member.role === 'owner' guard with isServerOwner()
- Remove getMemberRole() helper and legacy bridge code
- MemberSidebar groups by highest-positioned role instead of legacy string
- ServerSettings replaces admin/member dropdown with role checkboxes
- Message.tsx derives color from roles[] with owner fallback via ownerId
- Existing DBs keep vestigial column (Drizzle ignores it); new DBs omit it
Wire the bitwise permission engine end-to-end:
- Add sendToChannel() to ConnectionManager, filtering WS recipients by VIEW_CHANNEL
- Convert 6 channel-scoped events (messages, typing, reactions) from sendToServer to sendToChannel
- Add broadcastOverrideChange() to push channel_updated/channel_deleted per-user on override mutations
- Bridge legacy server_members.role TEXT to member_roles junction table on PATCH
- Add pushReadyPayload() to force re-sync frontend store after role changes
- Filter channels by VIEW_CHANNEL in GET /api/servers/:id to prevent initial load data leak
- Pre-compute viewers before CASCADE delete on channel_deleted
- Fix frontend channel event handlers to upsert/cleanup channelToServerMap and channelPermissions
- Add ChannelSettingsModal with Private Channel toggle and gear icon in ChannelSidebar
Replace string-based role checks (role === 'admin') with a bitwise BigInt
permission system. Adds computePermissions() resolution engine following
Discord's model: @everyone base → role union → admin shortcut → channel
overrides (role deny/allow → member deny/allow). Ready payload now filters
channels by VIEW_CHANNEL and attaches per-user myPermissions to each
server and channel. Includes channel_overrides table, @everyone role
auto-creation, migration for existing servers, and override CRUD API.
Wrap all multi-write DB operations in atomic transactions (server/channel
creation, message+attachment linking, DM creation, friend acceptance,
cascading deletes) to prevent partial-write corruption.
Batch N+1 queries in WS ready payload into O(1) bulk fetches with
chunked inArray() to respect SQLite's variable limit.
Fix chat history regression where background WS messages bypassed
channel load by switching the guard from messages.has() to hasMore.has().
Add LRU channel eviction (20 cached, evict to 15) and per-channel
message cap (200) to bound client memory growth.
Shorten WS heartbeat from 30s to 15s for aggressive proxy/NAT
environments. Clear all user-scoped stores on logout to prevent
cross-session data leaks.
Extract LiveKit internal accessors into shared livekitInternals utility.
Replace rigid SCREEN_QUALITY_MAP (6 hardcoded VideoPreset strings) with a
builder function that computes bitrate, degradation preference, and content
hint from three independent axes (height, fps, content mode). Camera is
decoupled onto a fixed 720p30 preset so screen share changes no longer
affect camera quality. New ScreenShareSettingsPopover replaces the old
VideoQualityPopover with pill-style selectors. Store migrated to v5 with
backwards-compatible migration from videoQuality string.
Extract stats polling into useTrackStats hook with per-sender/receiver
stats via RTCRtpSender.getStats(), delta-based FPS (fixes 25fps overwrite
bug with camera+screenshare), and track-to-source matching via LiveKit
TrackPublication identity. ConnectionInfoPopover becomes a pure display
component with Network/Audio/Video sections. Remove soft-launch diagnostic
loop from useLiveKit.
DM uploads silently failed because the 5-point chain (types, frontend,
POST, GET, WS broadcast) was never wired for attachments. Added
buildDmMessageWithUser/getDmMessageWithUser helpers mirroring the server
channel pattern, and plumbed attachmentIds + replyToId through all DM
code paths.
Unlock group DMs by removing the 2-member assumption across the stack.
No schema migration needed — dm_members junction table already supports
N members. Includes dedup bug fix, add/leave member endpoints, late-join
call support, group-aware sidebar rendering, and AddDmMemberModal.
Replace dual voiceStates + activeCalls maps with a single VoiceRoom
system that tracks both server channels and DM calls uniformly.
Fixes four bugs:
- voice_status silently dropped for DM call participants
- DM calls not cleaned up on WebSocket disconnect
- DM call state missing from ready payload on reconnect
- No spatial tracking of DM call participants
VoiceControls was gated on server-only `currentVoiceChannelId`, now also
triggers on `activeDmCall`. Added VoiceChatPanel to DM call view matching
the server voice channel pattern.
Closure guards on all 15 handlers already isolate React state from stale
rooms — the sledgehammer removeAllListeners() was stripping the SDK's own
internal listeners (PCManager, SignalClient), causing "Tried to add a track
for a participant that's not present" errors on rapid channel switches.
LiveKit's Room.disconnect() tears down WebRTC but leaves .on() handlers
attached. Every connect() registers ~15 event handlers, which accumulate
on orphaned Room instances during rapid channel switches or HMR, causing
MaxListenersExceededWarning. Added destroyRoom() helper that calls
removeAllListeners() before disconnect() at all four teardown sites.
Merge connectDm() into connect() with isDm flag, eliminating ~145 lines
of duplicated LiveKit room setup. DM calls now inherit all event handlers
(SpeakingDetector cleanup, deafen broadcasts, metadata changes). Replace
monolithic DmCallView with shared VoiceGrid + VoiceControlBar components,
making DM calls group-DM-ready with full feature parity.
Camera and LIVE badges in the channel sidebar were only visible to users
who had joined the same LiveKit room. Widen the voice_status WS event
from {isMuted, isDeafened} to {isMuted, isDeafened, isCameraOn, isScreenSharing}
so all server members see camera/screenshare indicators without joining voice.
30s ping/pong keepalive prevents proxy/NAT from killing idle connections.
Reconnect sound now only plays if downtime exceeds 3s, suppressing phantom
audio from brief network blips.
AudioManager.setScreenShareActive(true) was firing before the browser's screen
picker, killing the mic stream. The picker suspends getUserMedia while its secure
overlay is open, trapping the mic in a dead state for 5-30s. Now the AEC rebuild
fires after the track is acquired — mic stays alive during the picker.
Root cause: degradationPreference 'maintain-resolution' forced FPS drops under CPU
pressure during fast-motion gaming. Changed to 'maintain-framerate' so the encoder
drops resolution temporarily instead of stuttering. Added contentHint='motion' to
optimize for temporal prediction (more P-frames, fewer I-frames).
Screen sharing was publishing at h360 (640x360) and never ramping to target
resolution due to stale closure in setTimeout, wrong initial quality anchor,
and 5 competing code paths with inconsistent bitrates.
- Create utils/screenShare.ts as single source of truth for all screen share ops
- Publish at target resolution from the start (not h360 → ramp)
- Read store at call time in timers (eliminates stale closure bug)
- Use maintain-resolution for screen content, maintain-framerate for camera
- Fix OS-level "Stop sharing" not resetting store or restoring AEC
- Enable dynacast for SFU quality signaling
- Reconcile QUALITY_MAP to canonical bitrates across all 7 files
LiveKit's server-side VAD has a high, non-configurable threshold that misses
conversational speech. Replace it with a SpeakingDetector singleton that uses
Web Audio AnalyserNodes to read actual RMS levels per participant (50ms poll,
0.008 threshold, 250ms hysteresis hold).
Eliminate double-state architecture (useState → useEffect bridge → store)
that lost speaking events due to React 18 batching. ActiveSpeakersChanged
now writes speakingParticipantIds directly to voiceStore; 200ms poll
catches missed SDK events. Each VoiceUser subscribes to its own identity
via fine-grained selector for minimal re-renders. Also adds connection
quality indicator and ConnectionInfoPopover.
tsc was emitting compiled .js files directly into packages/web/src/
alongside the .tsx source files because noEmit was not set. These
artifacts were never used — Vite compiles from .tsx source directly.
- Add noEmit: true to packages/web/tsconfig.json (tsc = type-check only)
- Delete all 62 orphaned .js files from src/ (-6,129 lines)
- Add packages/web/src/**/*.js to .gitignore as safeguard
- RNNoise (AI Noise Suppression) now enabled by default for all users
- Remove redundant browser Noise Suppression toggle from VoiceControls
and UserSettings — AudioManager handles it automatically as fallback
- Add AI Noise Suppression toggle to UserSettings panel
- Rename toggleRnnoise → setRnnoiseEnabled for clearer API
- Store migration v3→v4: enable RNNoise for existing users
- Keep Echo Cancellation and Auto Gain Control (orthogonal features)
RNNoise worklet outputs mono (1ch) which played only in the left ear.
Replace unreliable automatic up-mixing (channelCount/channelCountMode on
inputGain) with a ChannelMergerNode that duplicates the mono signal to
both L and R channels — guaranteed stereo by the Web Audio spec.
Inject @sapphi-red/web-noise-suppressor into AudioManager's input pipeline
as a toggleable AudioWorkletNode. The worklet is loaded lazily on first
enable, then kept alive — toggling bypasses by rewiring the graph without
destroying the WASM instance. Browser NS is forced off when RNNoise is
active to avoid double-processing. InputGain forced to stereo up-mix to
prevent mono-left-only output from the worklet. Also wires Phase 2 output
device routing (setSinkId) into AppLayout/ChannelSidebar.
- syncMic now uses setMicrophoneEnabled(false/true) to mute/unmute the mic
track in-place instead of tearing down and re-publishing via unpublishTrack().
This eliminates WebRTC renegotiation on every mute cycle and preserves the
Web Audio pipeline for future AudioWorklet injection (RNNoise WASM).
- Unify DmCallView mute/deafen with the primary syncMic path — removed direct
setMicrophoneEnabled calls, added missing auto-mute/unmute on deafen toggle.
- Strip redundant applyConstraints from VoiceControls noise suppression toggle
that was hardcoding echoCancellation:true/autoGainControl:true and fighting
AudioManager's constraint pipeline.
- Redesign DM close as a visibility flag (closed column) instead of row deletion,
so message broadcasts still reach users who closed a DM conversation
- Add broadcastDmMessage() helper that auto-resurfaces closed DMs when new messages arrive
- Wire Remove Friend button in DM welcome header with onClick handler and friend check
- Fix deploy.sh to cd to its own directory so rsync always runs from project root
- Add WS events: dm_channel_created, dm_channel_closed, friend_removed,
channel_created/updated/deleted, server_updated
- Fix first-ever DM: broadcast dm_channel_created to recipient
- Add DELETE /api/dm/:id for closing DMs with re-open support
- Wire Close DM button in sidebar
- Fix dm_message_created for unknown channels (safety net)
- Broadcast friend_removed on friend deletion
- Sound system: track realtimeMessageEvents separately from API loads,
play notification for messages in all channels, not just current
- Optimistic updates: send/edit/delete messages appear instantly with
rollback on failure, temp message deduplication on WS echo
- Channel CRUD broadcasts to all server members
- Server update broadcast on PATCH
- Server join registers user in connectionManager immediately
- Reconnect: only reload current channel, preserve other channel caches
- Activity panel, right panel, member list toggle components
Chrome's AEC uses getDisplayMedia audio as a reference signal and
aggressively ducks the microphone even when headphones are used.
This adds a screenShareActive flag to AudioManager that forces
echoCancellation off during screen share, plus Chromium-specific
goog* constraints as belt-and-suspenders.
Disable AGC by default to prevent Chrome from crushing mic sensitivity
when stream audio is playing. Add user-facing toggles for echo
cancellation, noise suppression, and auto gain control. Decouple voice
and stream audio by routing through ctx.destination instead of shared
compressor. Track mic stream generation to re-publish when settings change.
Add MutationObserver to neutralize LiveKit's re-attached <audio> elements,
mark our keep-alive elements with data-opencord, detach tracks on unsubscribe,
remove dangerous blanket .play(), and soften compressor to transparent limiter.
Show local user's own screen share in their stream tile by bypassing
the isSubscribed check for local participants and auto-watching/unwatching
local streams on publish/unpublish. Eliminate audio double-playback by
detaching LiveKit's auto-attached audio elements so GlobalAudioRenderer
is the sole playback path, making attenuation and volume controls effective.
Fix GainNode leak in useAudioTrackPlayer cleanup paths.
Streams no longer auto-focus into large view when they start — tiles
stay in the equal-size grid until manually clicked. Audio playback is
moved out of VoiceUser/StreamTile into a new GlobalAudioRenderer
component rendered in AppLayout so it survives channel navigation.
Streams now appear as separate tiles in the voice grid alongside the
user's camera/avatar tile, matching Discord's model. Each stream tile
has independent volume, mute, watch/unwatch controls, quality badges,
and stream attenuation that ducks audio when someone speaks.
Pass audio: true to setScreenShareEnabled so the browser offers the
"Share audio" checkbox. Track ScreenShareAudio from remote participants
and play it through a dedicated audio element with the same volume
pipeline (including boost >100%).
Jannis Braun