backspace/docs/systems/voice.md

Voice, Video & Calls System

Source files:

• Server: routes/livekit.ts, ws/handler.ts, ws/events.ts
• Client: hooks/useLiveKit.ts, stores/voiceStore.ts, utils/voice.ts, utils/voiceActions.ts, utils/screenShare.ts
• Shared: packages/shared/src/constants.ts (bitrate matrix, resolutions)
• Audio: audio/AudioManager.ts, audio/SpeakingDetector.ts

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Voice Channel Join Flow

1. Client sends voice_join { channelId } via WS
2. Server checks CONNECT permission, enforces one-room-per-user
3. Server loads voice restrictions from DB (space mute/deafen)
4. Server broadcasts voice_state_update { action: 'join' } to space
5. Client calls POST /api/livekit/token { channelId } → gets JWT + LiveKit URL
6. Client connects to LiveKit room with token

Voice presence bootstrap on mid-session space join

A client learns who is sitting in a space's voice channels from the WS ready payload at connect time. Joining a space without reloading therefore needs the same bootstrap for the new space, or its voice channels render empty until a refresh. The server pushes a scoped space_voice_state snapshot from ConnectionManager.addUserSpace (the single join chokepoint), built by buildSpaceVoiceState(spaceId, userId) — the same VIEW_CHANNEL-filtered helper that feeds ready. The client applies it via utils/voiceStateSync.applySpaceVoiceState. See docs/systems/websocket.md → "Mid-session space join" for the full rationale (single ordered channel, no snapshot-vs-stream race).

Microphone pre-arm (iOS user-gesture discipline)

utils/voice.joinVoiceChannel fires AudioContext.resume() and AudioManager.setInputDevice(inputDeviceId) (which ends in getUserMedia({audio:…})) synchronously inside the click handler, before the connectFn(channelId) call. iOS Safari only surfaces the microphone permission prompt when getUserMedia is invoked from inside an active user-gesture; the original flow only acquired the mic in useLiveKit's syncMic effect, which fires AFTER room.connect() resolves (token fetch + WS handshake) — many awaits past the gesture window. iOS PWA standalone is especially strict and would silently never surface the prompt; the user would see "Waiting for others to join…" indefinitely until they locked/unlocked the device (which iOS treats as a fresh activation).

Pre-arm is fire-and-forget: the mic acquisition runs in parallel with the LiveKit handshake, and AudioManager.inputSwitchChain's serialization guarantees useLiveKit.syncMic's subsequent call short-circuits on the already-acquired currentStream (no double prompt, no second getUserMedia).

Listener mode (micPermissionDenied)

When the user denies the prompt (or has previously denied at the OS level), the pre-arm's setInputDevice rejects with NotAllowedError. The voiceStore.micPermissionDenied flag is set to true and the LiveKit connect proceeds anyway — the user appears in the voice channel as a connected participant who can hear others but has no microphone publication. useLiveKit.syncMic checks the flag at the top of its body and skips the publish branch entirely.

The flag clears via:

• requestMicPermission() (in utils/voice.ts) — must be called from a user-gesture handler (button click). Clears AudioManager.inputDenialError cache, calls setInputDevice from a fresh activation. On success, sets micPermissionDenied=false and useLiveKit.syncMic re-fires (dep on micPermissionDenied) to publish the freshly acquired track.
• voiceStore.leaveVoice() / handleForceDisconnect() / resetSession() / reset() — flag resets so the next join attempts a fresh prompt.

UI affordances:

• Mobile (MobileVoiceFullScreen): banner below the header reads "Microphone access denied — You're listening only". A right-aligned "Allow microphone" button calls requestMicPermission().
• Desktop (VoiceControlBar): (Future) — same listener-mode state needs a parity affordance. Desktop is unaffected by the iOS gesture-window bug in practice (browsers there prompt on getUserMedia regardless of activation state), but if a desktop user denies the prompt, the same flow applies.

AudioManager.inputDenialError caches the most recent NotAllowedError. Subsequent setInputDevice calls re-throw the cached error rather than firing a second getUserMedia — iOS otherwise would queue a second prompt that has lost its activation, leading to a silent hang. Cleared by AudioManager.clearInputDenial() (called from joinVoiceChannel's pre-arm and from requestMicPermission).

Token grants (space channels):

• SPEAK → can publish MICROPHONE + CAMERA
• STREAM → can publish SCREEN_SHARE + SCREEN_SHARE_AUDIO
• Missing permission → grant excludes those sources

Token grants (DM calls): Always full (canSpeak=true, canStream=true)

Identity format: {userId}:{username}, TTL: 1 hour, Room: {channelId} or dm-{dmChannelId}

Multi-tab: Each user has one voiceWs binding. New tab → old socket gets voice_disconnected { reason: 'displaced' }

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DM Call State Machine

States: ringing → active → destroyed

Event	Action	State
dm_call_start	Room created, caller bound, 60s timeout starts	ringing
dm_call_incoming	Broadcast to DM members (excludes caller)	ringing
dm_call_accept	First accept: ringing→active. Late joins welcome (group DM)	active
dm_call_reject	Room destroyed, caller unbound	—
dm_call_end	All participants unbound, room destroyed	—
Timeout (60s)	Auto-cleanup if still ringing, broadcast dm_call_ended	—

Edge cases:

• Starting new call cancels any other ringing calls by same caller
• Socket close during ringing → auto-cleanup
• Participants drop to 0 in active state → room destroyed

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Federated DM Calls

DM calls work across federated instances. The caller's instance hosts the LiveKit room; remote clients connect to it directly. Call signaling is relayed to ALL active federation peers via synchronous HTTP POST (not the outbox worker). This ensures calls ring on every instance where a participant is connected, even if the DM is local-only on the caller's instance.

Universal Relay

All dm_call_* signaling events (start, accept, reject, end) are relayed to every active federation peer in parallel. Each sendCallRelay call has a 10-second HTTP timeout. This bypasses the outbox worker — call signaling is latency-sensitive.

Auto-peering at send time. If the target origin has no active peer record, sendCallRelay races an ensurePeered handshake against a 3 s deadline (CALL_PEERING_TIMEOUT_MS). On success the relay POSTs normally; on timeout it returns peer_transient_failure without aborting the background handshake, so a subsequent attempt typically succeeds. Typing (sendTypingRelay) passes peeringTimeoutMs: 0 — the POST is skipped for non-active peers and a warm-up ensurePeered runs in the background.

Call relay failure surface. Every dm_call_{start,accept,reject,end} relay is failure-aware. On failure the originating server emits a dm_call_undeliverable event with a phase discriminator identifying which action failed. Client copy is phase-specific; state rollback depends on the phase.

phase	terminal	Emitted when	Client action
start	true	No plausible recipient after targeted-peer fan-out; ring room destroyed.	Clear outgoingCall, disconnect LK, warning toast.
start	false	Some targeted peers failed but reachable recipients remain; ring continues.	Keep state; info toast.
accept	true	Acceptor's B→host relay failed; optimistic state is rolled back on B.	Clear activeDmCall + incomingCall, disconnect LK, warning toast.
accept	false	Host → peer fan-out of accept failed; local host call continues.	No state change; info toast.
reject	false	Rejector's relay to host failed OR host's fan-out after a local reject failed; state already cleared.	No state change; info toast.
end	false	Ender's relay to host failed OR host's fan-out after a local end failed; state already cleared.	No state change; info toast.
host_unreachable	true	A FederatedCallEntry's federatedCallHost peer transitions out of active, OR the 30s sentinel detects a non-active host for an existing entry.	Clear activeDmCall + incomingCall, disconnect LK, warning toast ("Call ended — {label} became unreachable.").
no_recipient	true	Remote returned 200 but had no reachable recipient (Path A: all members offline; Path B: zero participant matches). Caller fast-fails within the relay round-trip; ring room destroyed.	Clear outgoingCall, disconnect LK, warning toast ("{peerLabel} couldn't ring anyone."). Folds into multi-failure info copy when not the sole failure.

Accept-rollback semantics. handleDmCallAccept Path 2 transitions the FederatedCallEntry to active and broadcasts dm_call_accepted optimistically so the acceptor's UI flips immediately. If the B→host relay fails, the server clears the entry, fans dm_call_undeliverable { phase: 'accept', terminal: true } out to all ringed users on B (via sendToFederatedCallUsers), and the client tears its call state back down.

Reject / end are optimistic. Local state is cleared before the relay is awaited because the user's intent is to terminate. If the relay fails, the originator receives an informational dm_call_undeliverable { terminal: false } so they know remote peers may briefly display stale state; no local rollback.

Ring-timeout fan-out. When the host's 60 s ringing timeout fires without an accept, dm_call_end is fanned out to all remote peers so stranded Path-A/B ringees on other instances exit their ring state instead of lingering. Registered via connectionManager.setRingTimeoutFanoutHook from the WS events module.

Remaining edge. When a non-host participant ends an active call and the relay to the host fails, the host's activeDmCall marker lingers until manual end — LK ParticipantDisconnected tears down the voice UI but does not clear the DM-call marker on the host side. This is the caller-side mirror of the remote-participant problem and is not covered by the Remote-Participant Host Unreachable Eviction mechanism above (which only reasons about FederatedCallEntry state). Tracked separately.

Remote-Participant Host Unreachable Eviction

When a FederatedCallEntry's federatedCallHost becomes unreachable (peer status transitions to unreachable, needs_attention, rejected, or revoked), the entry owner evicts the stranded state and notifies its local ringed users with dm_call_undeliverable { phase: 'host_unreachable', terminal: true }. Two signals drive the eviction:

1. Fast path (onPeerDeactivated hook): every peer-status transition out of active invokes ConnectionManager.evictFederatedCallsForHost(peerOrigin, ...). Call sites are listed in the onPeerDeactivated docstring (audit via grep onPeerDeactivated().
2. Backstop (30s sentinel): runFederatedCallSentinelTick in federationWorker.ts iterates active entries, looks up each distinct federatedCallHost's current peer status, and evicts non-active matches.

Typical eviction latency is ~90s (time for outbox traffic to fail the unreachable threshold + one sentinel tick). Worst case on an idle instance with no outbox traffic is ~15.5min (health-check cadence + sentinel).

Covers the ringing and active states on the remote-participant side. The caller-side mirror — host's own activeDmCall lingering when its LK room empties silently — is a separate, documented out-of-scope edge.

Dual-Path Processing

When a peer instance receives a call relay, it uses one of two delivery paths:

Path	Condition	Delivery
A	DM exists on the receiving instance	Look up dm_members for the local dmChannelId and deliver to connected members
B	DM does not exist on the receiving instance	Match participants by homeUserId + homeInstance identity against connected WebSocket users

Path B enables calls to ring for federated users even when no local DM channel has been created yet (e.g., first contact via a federated call).

FederatedCallEntry

The in-memory call state (FederatedCallEntry) is keyed by federatedId (not dmChannelId):

• dmChannelId is nullable — null for Path B scenarios where no local DM channel exists
• ringedUserIds tracks all users who were notified of the incoming call, used for end-call cleanup
• callerId, callerHomeUserId, callerHomeInstance identify the caller across instances

Late-Bind dmChannelId

When findOrCreateDmChannel creates a local DM channel during an active federated call (e.g., the first message arrives while a call is ringing), it binds the dmChannelId on the existing FederatedCallEntry. This transitions the call from Path B to Path A delivery without interrupting the call.

Token Generation & Room Identity

Token generation: generateFederatedCallToken(federatedId, homeUserId, displayName) in routes/livekit.ts issues 5-minute tokens scoped to the federatedId room (not the local dmChannelId). Grants full DM permissions (mic, camera, screen share, subscribe, data channel).

LiveKit URL: The relay sends config.livekit.url (e.g., wss://nova.ddns.net/livekit). Must be wss://, not https:// — the LiveKit SDK requires a WebSocket URL.

Token endpoint: POST /api/livekit/token uses federatedId as the room name when the DM channel has a federatedId set, ensuring both instances join the same LiveKit room.

Identity format:

• Federated calls: ${homeUserId}:${displayName} — stable across all instances
• Local calls: ${userId}:${username} — unchanged

Client identity resolution: For federated calls, the client splits the LiveKit participant identity on : and matches homeUserId against the DM member list (which stores homeUserId for all members). This resolves the correct display name and avatar regardless of which instance the participant is on.

Client-Side Call Routing

callOrigin: Set to the WS origin that delivered the dm_call_incoming event (the home instance), NOT the call host URL. Accept/reject/end route through this WS. The home instance's server finds the FederatedCallEntry and relays to the host via S2S HTTP. This is reliable regardless of whether the client has a multi-instance WS to the host.

handleAccept: Sets activeDmCall and clears incomingCall directly in the click handler — does not wait for the server's dm_call_accepted response (races with connectFn's async AudioContext resume).

Passive ready handler: On page refresh/restart, the ready payload includes active calls but the client does NOT auto-connect to LiveKit. Users must re-accept. This prevents identity slot wars when the same user has multiple sessions.

A DM call has no currentVoiceChannelId (space↔DM are mutually exclusive). Entering a space channel clears activeDmCall (setCurrentVoiceChannel); entering a DM call must clear currentVoiceChannelId. The latter is done by clearSpaceVoiceForDmCall() (utils/voice.ts), invoked synchronously at the top of connect() when isDm. Without it, VoiceChannel renders the occupant list for currentVoiceChannelId from the live LiveKit participants, so a lingering space currentVoiceChannelId maps the DM call's participants onto the old space channel — the caller/acceptor appears to still be sitting in it. The server already drops the user from the space room (dm_call_start / dm_call_accept → leaveCurrentRoom → broadcastRoomLeave), so this is a client-state fix; it also optimistically removes self from the old channel's voiceUsers for an immediate sidebar update. Regression test: utils/clearSpaceVoiceForDmCall.test.ts.

Caller connect guard. In dm_call_accepted, the caller connects to the DM room gated on wasOutgoingCall (only the initiating session ever sets outgoingCall) — not on !isLiveKitConnected. A caller already sitting in a space voice channel is LiveKit-connected; gating on that would skip the DM connect and strand them in the space channel. connect() de-dupes an already-connected same room, so wasOutgoingCall alone is sufficient.

DM-call teardown never disconnects a space connection (teardownDmCall). The dm_call_ended / dm_call_rejected / terminal dm_call_undeliverable handlers all route through teardownDmCall() (useWebSocket.ts), which clears the call UI/federation state and tears down LiveKit only when currentVoiceChannelId is null. disconnectFn() tears down whatever room is active, and a space channel and a DM call are mutually exclusive (setCurrentVoiceChannel clears activeDmCall). The load-bearing case: when the last participant in a DM call joins a space voice channel, their post-connect voice_join empties the server-side DM room, so broadcastRoomLeave (events.ts) broadcasts dm_call_ended back to every DM member — including them. Without the guard, that echo would disconnectFn() the space room they just connected to, stranding the UI on "Connecting…" until a manual rejoin. The first participant to leave is unaffected (room still occupied → no dm_call_ended). Regression test: hooks/teardownDmCall.test.ts.

SoundController Federation Awareness

The SoundController uses isSelf(id) which checks against BOTH currentUser.id (local snowflake) and currentUser.homeUserId (federated home ID). In federated calls, updateParticipants resolves identity to the local snowflake when activeDmCall is set, but reverts to raw homeUserId when it's cleared during disconnect. Both formats must be recognized as "self" to prevent phantom join/leave sounds.

Disconnect teardown: roomRef is set to null before calling destroyRoom(). This prevents ParticipantDisconnected events (fired during teardown) from triggering updateParticipants, which would cause user_leave sounds for departing participants alongside the disconnect sound.

Sound effects. The full system-sound inventory and trigger map lives in docs/systems/sounds.md. This includes the stream_watch data-channel protocol used for viewer detection (mirroring the existing deafen data-channel ping receiver in handleDataReceived).

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Voice Moderation

Three independent muting mechanisms:

1. User Self-Mute/Deafen

• Client toggles in voiceStore
• Broadcasts via voice_status WS event
• If also space-muted, remains effectively muted

2. Space Mute/Deafen (moderator, persisted)

• Requires MUTE_MEMBERS / DEAFEN_MEMBERS permission
• Stored in voice_restrictions table (survives reconnect)
• In-memory: spaceMutedUsers / spaceDeafenedUsers sets ("spaceId:userId" keys)
• On voice_join: restrictions loaded from DB into memory
• Broadcasts voice_space_muted / voice_space_deafened to all space members

3. Permission Mute (automatic, ephemeral)

• Triggered when user loses SPEAK permission (role update)
• checkVoicePermissions(spaceId) re-evaluates all users in space voice
• NOT persisted — derived from role permissions on demand
• Broadcasts voice_permission_muted

Effective state: effectiveMuted = isMuted || spaceMuted || permissionMuted

Move & Disconnect

• voice_move: Requires MOVE_MEMBERS. Same space only. Preserves voice status.
• voice_disconnect: Requires DISCONNECT_MEMBERS. Full teardown.

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Screen Sharing

Resolution & Framerate Options

Standard resolutions: 540, 720, 1080, 1440, 2160 (+ 'native')
Standard framerates: 30, 45, 60, 75, 90, 120
Width map: 540→960, 720→1280, 1080→1920, 1440→2560, 2160→3840

VP9 Bitrate Matrix (kbps)

       30    45    60    75    90    120
540:  1500  2000  2500  2800  3200  4000
720:  3000  3500  4000  4500  5000  6000
1080: 6000  7000  8000  9000  10000 12000
1440: 10000 12000 14000 16000 18000 22000
2160: 20000 24000 28000 32000 38000 45000

Config Object

ScreenShareConfig {
  height: number | 'native',       // Resolution or capture at display res
  fps: number,                     // 30-120
  mode: 'gaming' | 'text',         // Affects bitrate & content hint
  customBitrateKbps: number | null, // Admin override (if allowed)
  shareAudio: boolean               // System audio loopback (see Platform Support below)
}

Build Pipeline (buildScreenShareOptions())

1. Resolve bitrate from matrix (custom > override > default > native estimate)
2. Clamp to instance limits (minBitrateKbps, maxBitrateKbps)
3. Compute min bitrate = 25% of max
4. Codec: VP9 (default) or H.264 (hardware overdrive)
5. VP8 simulcast backup at reduced framerate/bitrate
6. Content hint: 'detail' (text) or 'motion' (gaming)

Native Mode

• Captures at display's full resolution
• Snaps to nearest known tier for bitrate lookup
• Scales proportionally: baseKbps * (capturedPixels / knownPixels) * (fps / knownFps)

Hardware Overdrive

• Forces H.264 hardware encoder via SDP profile override
• Applied 2s after stream starts (after WebRTC negotiation), re-applied at 5s
• 4s: detects if using software fallback, warns user

Instance-Level Limits (admin-configured)

• allowedResolutions, allowedFramerates (CSV in instance_settings)
• maxResolution, maxFramerate, maxBitrateKbps, minBitrateKbps
• allowCustomBitrate toggle
• bitrateMatrixOverrides (JSON sparse overrides)

System Audio Loopback (shareAudio)

The "Share system audio" toggle in ScreenSharePicker adds an audio track to the screen-share publication. In the browser it maps to getDisplayMedia({ audio: true }). In Electron, the setDisplayMediaRequestHandler callback (packages/desktop/src/main.ts) returns audio: 'loopback' to opt into Chromium's system-audio loopback path.

Platform	Mechanism	Notes
Browser (Chrome/Edge)	getDisplayMedia({ audio: true })	Tab/window/system audio per the user's pick
Electron / Windows	Chromium native loopback	Works out of the box
Electron / macOS 13+	CoreAudio Tap (Catap)	Requires NSAudioCaptureUsageDescription (set by electron-builder.yml#mac.extendInfo)
Electron / Linux	PulseAudio loopback	Requires the PulseaudioLoopbackForScreenShare Chromium feature flag — enabled at startup in main.ts for Linux. Works on PulseAudio and on PipeWire systems with the pipewire-pulse compat layer. PipeWire-only systems without pulse compat will fail.

Failure handling. When loopback is not supported, Chromium rejects the entire getDisplayMedia request — the source-picker selection has already been consumed, so silently retrying without audio would re-prompt the picker. startScreenShare (utils/screenShare.ts) instead surfaces a warning toast directing the user to disable "Share system audio" if their system does not support loopback. We do not auto-mutate the user's shareAudio preference.

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Mobile Voice Rendering

Mobile (MobileVoiceFullScreen) renders the same VoiceGrid component as desktop. There is no mobile-specific tile component — the rendering, attach/detach, adaptive-stream subscription, focused-publisher layout, and context menus all come from the shared VoiceGrid / VoiceUser / StreamTile pipeline. The only mobile-specific addition is auto-focus on the first live screen-share publication (so phone users don't have to discover tap-to-focus).

See docs/systems/mobile-ui.md → "MobileVoiceFullScreen" for the auto-focus state machine, control-bar wiring, and layout sizing.

Why the shared component path matters.

• Local camera preview: VoiceUser attaches the local participant's videoTrack to a <video muted>. Mobile gets the self-preview "for free".
• Remote cameras: Track.attach(videoEl) registers the element with LiveKit's RemoteVideoTrack adaptive-stream observer, so the SFU automatically picks the appropriate simulcast layer based on the painted tile size on the phone. No mobile-specific bitrate clamp is needed.
• Screen-share: StreamTile lazily subscribes via setStreamSubscription only after the user taps "Watch Stream" (or auto-focus does so on mobile, which currently still requires the user to tap the in-tile "Watch Stream" CTA — auto-focus only sets the focused publisher; it does not auto-subscribe to bandwidth-heavy screen-share tracks).
• Mute / deafen / speaking-ring overlays, watch/unwatch controls, local mute, volume sliders — identical between mobile and desktop.

Screen-share button wiring on mobile. MobileVoiceFullScreen's screen-share button calls handleScreenShareAction() from utils/voiceActions, not voiceStore.toggleScreenShare. The store action only flips the isScreenSharing boolean and never calls getDisplayMedia. The canonical handleScreenShareAction is shared with desktop's VoiceControlBar and the keybind manager; it calls startScreenShare(room) / stopScreenShare(room) and broadcasts voice status to peers. iOS Safari does not support getDisplayMedia (the call rejects); this is a platform limitation. Android Chrome supports it and works.

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Voice Fullscreen

The fullscreen toggle in VoiceControlBar flips the voiceFullscreen flag in uiStore; an effect in MainContent.tsx enters/exits the browser's Fullscreen API on voiceContainerRef. A second effect listens to fullscreenchange and reflects the actual document fullscreen element back into the store, so pressing Esc or system-level fullscreen-exit keeps state in sync. voiceChatOpen && !voiceFullscreen hides the side chat panel while fullscreen is active.

Cross-browser API fallback. iOS Safari (and iPadOS pre-16.4) does not implement the standard Element.requestFullscreen() on generic elements, so the enter-fullscreen effect probes for the API in this order:

1. el.requestFullscreen() — standard
2. el.webkitRequestFullscreen() — older WebKit (some iPads, older Safari)
3. Silent fall-through — pure iPhone Safari has neither API on a <div> (only HTMLVideoElement.webkitEnterFullscreen() works, which we cannot use for the multi-tile voice container)

When neither native API is available the effect returns without throwing; the voiceFullscreen flag still applies h-screen to voiceContainerRef, which acts as the in-page maximize fallback (chat panel hides, header fades, control bar stays). The exit path mirrors this with document.exitFullscreen() → document.webkitExitFullscreen() → no-op. Both paths are wrapped in try/catch so a Promise rejection (e.g. user cancels via Esc mid-transition) does not surface as an unhandled error. The fullscreenchange listener is registered for both fullscreenchange and webkitfullscreenchange. Before this fallback, calling the missing API directly threw TypeError: requestFullscreen is not a function on iPhone Safari, which surfaced as a full-screen error overlay when an iPhone user crossed the 768 px desktop breakpoint in landscape mode.

Overlay portals: While fullscreen is active the browser's Fullscreen API renders only descendants of voiceContainerRef. Every overlay reachable during a call (context menus on StreamTile/VoiceUser/VoiceChannel, tooltips on the control bar, ConnectionInfoPopover, ScreenShareSettingsPopover, ConfirmDialog invoked from voice context-menu actions, and ScreenSharePicker) portals through usePortalContainer() so it lands inside the fullscreen element. Adding new overlays that can be opened from inside the call must follow the same contract — see docs/systems/design-system.md Surface Material Tiers.

---

Audio Processing

Feature	Default	User Control	Notes
Echo Cancellation	on	yes	Stays on during screen share (Chrome AEC handles it)
Noise Suppression	overridden	—	Managed by RNNoise state
Auto Gain Control	on	yes
RNNoise (ML)	on	yes	When enabled: browser NS forced off

Audio constraints applied to mic track:

{
  echoCancellation: userSetting,     // stays on during screen share
  noiseSuppression: rnnoiseEnabled ? false : userSetting,
  autoGainControl: userSetting,
}

Screen share audio (when enabled):

{
  restrictOwnAudio: true,    // Chrome 141+: exclude own tab audio
  echoCancellation: false,
  noiseSuppression: false,
  autoGainControl: false,
  channelCount: 2             // Stereo
}

Persistence: voiceStore with Zustand localStorage. Keys: echoCancellation, autoGainControl, rnnoiseEnabled, screenShareConfig.

Camera preset: 1280x720, 2Mbps, 30fps, H.264

---

Audio Device Selection (Microphone & Speakers)

Users pick mic and speaker devices in two surfaces:

1. User Settings → Voice & Video (AudioInputSection.tsx, AudioOutputSection.tsx) — full picker with input volume, live level meter, output volume, and a "Play test sound" button.
2. Bottom-left UserArea quick popups (ChannelSidebar.tsx UserAreaPanel) — opened by the caret buttons next to mute (input picker) and deafen (output picker). Same picker UX, more compact.

Both surfaces are backed by the shared useAudioDevices() hook. The store fields inputDeviceId and outputDeviceId (both string, default 'default') are persisted in voiceStore.

useAudioDevices() hook (canonical enumeration)

• Mirrors VideoSection.tsx's permission/enumeration/devicechange pattern.
• Mount-time probe: navigator.permissions.query({ name: 'microphone' }). Never auto-fires getUserMedia — that requires an explicit user gesture via the returned requestPermission().
• States: unknown → granted | prompt | denied. Lists are populated only in granted.
• Refreshes both inputs and outputs on every devicechange event.
• Output devices are gated behind microphone permission (no separate output permission exists in browsers).
• Returns inputLabels / outputLabels maps with disambiguation suffixes for duplicate names (e.g. "USB Audio (1)", "USB Audio (2)").

Output routing — AudioContext.setSinkId

All audio (remote voice, screen-share audio, sound effects) flows through AudioManager's master bus → AudioContext.destination. Output device switching is therefore done via AudioContext.setSinkId(deviceId), NOT via LiveKit's switchActiveDevice('audiooutput') (which targets <audio> elements that are killed by AppLayout's MutationObserver). Safari < 17 lacks setSinkId on AudioContext — AudioOutputSection detects this (once a real context exists) and falls back to OS default with an explanatory note.

Mobile platforms with no per-element output routing (iOS Safari): AudioOutputSection feature-detects 'setSinkId' in HTMLMediaElement.prototype at module load (cached). When false, the entire section is hidden — no header, no fallback copy. iOS users adjust audio routing via OS controls (Bluetooth menu, Control Center) and do not expect per-app output selection. Android Chrome ≥ 110 supports setSinkId and renders the picker normally. The detection runs before any hooks via an outer wrapper (AudioOutputSection → early-return-null → AudioOutputSectionInner) so the inner component's hook order remains stable.

Touch-close on device pickers

The Audio Input, Audio Output, and Video device dropdowns (AudioInputSection.tsx, AudioOutputSection.tsx, VideoSection.tsx) all listen for both mousedown AND touchstart ({ passive: true }) when implementing click-outside-to-close. iOS Safari does not synthesize mousedown reliably from a single tap; without the touchstart listener, mobile users would have to tap twice to dismiss an open popover.

Input pipeline — republish, never switchActiveDevice

Input device changes flow through AudioManager.setInputDevice(deviceId) (serialized chain). The useLiveKit syncMic effect detects the bumped stream generation and unpublishes/republishes via getFreshTrack(). This asymmetry vs. the camera (which uses room.switchActiveDevice('videoinput', …)) is intentional and documented under "Architectural asymmetry" below — the published mic track is the output of a Web Audio graph (RNNoise, gain, AEC), not a raw getUserMedia track.

Hot-plug seamlessness

The global devicechange handler in AppLayout.tsx does four things on every event:

1. Prune persisted IDs that no longer exist (pruneStaleDevices).
2. Re-acquire the live mic stream when inputDeviceId === 'default' AND AudioManager.hasActiveStream(). Chromium does NOT migrate an existing getUserMedia track to the new OS-default — calling setInputDevice('default') triggers a fresh getUserMedia which picks up the new default; syncMic then republishes.
3. Re-apply setSinkId('') when outputDeviceId === 'default', for the analogous reason.
4. Toast on a new audioinput group appearing (debounced 1s, deduped by groupId for 30s). Removals do not toast — the user already knows they unplugged it. Toast is informational ("AirPods Pro detected — choose it in Voice settings to switch") — never auto-switches; auto-switch would be a privacy/UX regression for users who deliberately keep a non-default device selected.

Mic-track-loss recovery

The published mic track is a clone of AudioManager's MediaStreamAudioDestinationNode output (see getFreshTrack()), and a destination-node track does not end on upstream loss — it just outputs silence. So the published track's onended is the wrong signal. Instead, AudioManager installs onended on every track of the upstream getUserMedia stream and exposes a subscription API:

• AudioManager.onInputTrackEnded(cb) — subscribers receive a 'unplug' | 'revoke' | 'unknown' reason hint and probe getUserMedia themselves to classify.
• Deliberate replacements (setInputDevice, setRnnoiseEnabled, setVoiceProcessing re-init) detach the per-track listener BEFORE calling .stop() and null currentStream immediately, so subscribers are never notified for non-loss events. A surviving listener (e.g. attached by a future external caller) bails via the currentStream !== capturedStream identity check.

useLiveKit subscribes to this signal whenever a room is connected, captures subscriberRoom = roomRef.current, and on emission:

1. Bail if the room has been replaced.
2. Probe getUserMedia({audio:{deviceId}}) to classify:
   • Probe succeeds → setInputDevice(deviceId) to re-acquire AND call republishMicrophone(subscriberRoom, lastMicGenRef) directly (the syncMic dep array does not include streamGeneration, so we cannot rely on it to re-fire).
   • NotAllowedError → "Microphone permission was revoked" (warning toast).
   • NotFoundError with non-default device → set store to 'default' and toast "Microphone disconnected — switched to system default". The store change triggers syncMic, which re-acquires + republishes via the shared helper.
   • NotFoundError on default → "Microphone disconnected".
   • Other → "Microphone could not be restored".

republishMicrophone is a module-level helper extracted from syncMic so both the normal device-change path and the recovery path share the staleness-check / unpublish / getFreshTrack / publish flow.

Privacy gate — never auto-fire getUserMedia

The useAudioDevices hook only calls getUserMedia from the explicit requestPermission() action. The previous ChannelSidebar.UserAreaPanel.loadDevices implementation fired getUserMedia({audio:true}) on every panel open as long as no AudioContext existed — which flashed the mic indicator even when permission had been previously granted in another session. That probe has been removed.

Resolved-default hint

When inputDeviceId === 'default' and a stream is active, AudioInputSection shows a Currently using: <label> subline by reading AudioManager.getCurrentInputDeviceId() and looking up the label in inputLabels. This makes the "default → which device?" indirection visible to the user.

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Camera Device Selection

Users pick a camera in User Settings → Voice & Video → Video. Selection is persisted in voiceStore.cameraDeviceId (string | null; null = "let LiveKit/browser auto-pick on next fresh enable").

Store field

• cameraDeviceId: string | null — persisted via partialize. No persist-version bump was needed when it was added: the existing merge { ...currentState, ...persistedState } hydrates absent keys from initialState automatically.
• Sister action: pruneStaleDevices() — sweeps mic, speaker, and camera persisted IDs, resetting any that aren't present in enumerateDevices(). Skips a kind whose enumerated set has no non-empty deviceIds (Firefox/Safari pre-permission obscures IDs and we cannot distinguish stale from obscured). Called from AppLayout mount and on every devicechange event.

Camera enable path (canonical)

utils/voiceActions.handleCameraAction() is the sole camera-toggle path — voice-bar button, mobile button, and keybind all call it. It applies CAMERA_PRESET (720p30 H.264) and injects cameraDeviceId into VideoCaptureOptions.deviceId when non-null.

Hot-swap mid-call

useLiveKit's syncCamera effect watches cameraDeviceId, isCameraOn, isConnected. When the published track's actual getSettings().deviceId differs from the store target (and target is non-null), it calls room.switchActiveDevice('videoinput', targetId) — an in-place source swap, no re-publish. Failure path: try to restore the previous deviceId in store (if its track is still live), else disable the camera entirely; toast "Could not switch camera". The null ("Auto") target is intentionally a no-op: no force-switch of an already-live publication.

isConnected here is strictly state === ConnectionState.Connected (useLiveKit.ts). Do not broaden it to include Reconnecting without revisiting the effect — switchActiveDevice against a reconnecting room would fail.

Track-end detection

On RoomEvent.LocalTrackPublished for the camera, the underlying MediaStreamTrack gets an onended listener. When it fires:

1. If consumeIntentionalCameraOff() returns true (user clicked the camera off; flag was set in handleCameraAction's disable branch), bail — no probe, no toast.
2. Else re-probe getUserMedia({video:{deviceId}}) to distinguish causes: NotAllowedError → "Camera permission was revoked" (macOS Privacy revoke); NotFoundError → "Camera disconnected" (unplug); other → "Camera unavailable".
3. Tear down camera state via the unified path: markIntentionalCameraOff() → setCameraEnabled(false) → isCameraOn = false → broadcastVoiceStatus() → toast.

The _intentionalCameraOff module-level flag in voiceActions.ts is the gate. Producers: handleCameraAction disable branch, syncCamera rollback, the track-end handler's own teardown. Consumer: the track-end handler.

Two-mode preview (in VideoSection.tsx)

Mode	Triggered when	Source
In-call	An LK camera publication exists	Attach the LK MediaStreamTrack to the preview <video>
Pre-call	No room or no publication	Open a getUserMedia({ video: { deviceId } }) stream for the selected device

Mode is reactive on isCameraOn changes. Pre-call streams stop on tab hide (visibilitychange), modal close, panel switch, and component unmount; in-call attaches detach the same way but the LK track keeps running.

Privacy: dormant-by-default. The pre-call mode never auto-starts. On section mount, navigator.permissions.query({ name: 'camera' as PermissionName }) reports the permission state without firing the camera. The preview tile is dormant (placeholder + "Click to test camera" overlay) until the user explicitly clicks it, or until the prompt-state CTA button triggers getUserMedia (which both grants permission and opens preview in one step). Rationale: macOS holds the camera LED on for ~2s after release, so any incidental getUserMedia call (probe, transient mount) flashes the LED — a privacy/UX defect. The only entry points to getUserMedia are explicit user gestures: dormant-tile click, prompt CTA, "Try again" in the denied banner, and dropdown change while preview is already running.

Mobile pre-join preview (in MobileVoiceJoinSheet.tsx)

The mobile bottom-sheet voice-join flow exposes the same dormant-by-default camera preview pattern as VideoSection's pre-call mode. When the user taps a voice channel on MobileSpacesScreen, the join sheet opens with a 16:9 preview tile. The tile starts dormant ("Tap to preview camera") — never auto-fires getUserMedia. Tapping the tile, the prompt-state CTA, or "Try again" after a denial calls getUserMedia({ video: { deviceId: ... } }) with the user's persisted cameraDeviceId from voiceStore.

Lifecycle is hard-bound to the sheet:

• Arm: explicit user tap inside the sheet (any of the entry-point buttons).
• Disarm: sheet close (any path: backdrop tap, close button, channel switch, Join Voice tap which transitions to the in-call flow). The single source of truth for "camera off when sheet closes" is the cleanup effect on the component's unmount — the parent (MobileSpacesScreen) removes the sheet, the cleanup runs stopPreview(), and tracks are stopped + srcObject cleared.
• Tab-hide: matches VideoSection — release on visibilitychange === 'hidden', no auto-resume; user must re-tap.
• Camera switch: when multiple cameras are present, a picker overlay in the bottom-left of the tile lets the user swap. The picker is gated on permState === 'granted' && cameraDevices.length > 1 so it doesn't appear for single-camera devices. Switching cameras while preview is running re-opens getUserMedia for the new deviceId; cameraDeviceId is shared with voiceStore so the selection persists into the call.
  • Picker popup is portaled to document.body. The trigger button sits inside the aspect-video overflow-hidden preview tile, but the dropdown list is rendered as a position: fixed element via createPortal so it can extend above the tile. Position is captured from the trigger's getBoundingClientRect() (re-captured on resize / capturing scroll) and pinned via bottom = window.innerHeight - rect.top + 4 so the popup expands upward. The list has max-height: min(50vh, 320px), overflow-y: auto, and -webkit-overflow-scrolling: touch so every entry stays reachable on a long device list. Click-outside dismissal listens for both mousedown and touchstart, and excludes both the anchor and the portaled popup (the popup is not a DOM descendant of the anchor since it lives in document.body).

The <video> element is set up identically to VideoSection for iOS Safari compatibility: autoPlay playsInline muted attributes on the element, srcObject set after the await getUserMedia, and a defensive videoEl.play().catch(() => {}). iOS Safari requires autoPlay because the user-gesture context expires across the await — play() alone fails silently.

Architectural asymmetry: mic republishes, camera switches

Mic publishes the output of a Web Audio graph (RNNoise, gain, AEC) — LocalParticipant.switchActiveDevice cannot operate on it because the published track is a MediaStreamAudioDestinationNode.stream's track, not a raw mic track. Mic device changes therefore unpublish/republish via AudioManager.getFreshTrack(). Camera publishes the raw getUserMedia track and uses switchActiveDevice for in-place swaps. Do not unify.

Federation

No federation work. The LK room is at the host instance; remote peers connect to it directly. switchActiveDevice is room-internal and works regardless of where the room lives.