# 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`

---

## 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

**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' }`

---

## 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

---

## 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 via `Promise.all`. Each `sendCallRelay` call has a 10-second timeout. This is a synchronous HTTP POST to the peer's federation endpoint — it bypasses the outbox worker entirely because call signaling is latency-sensitive and must not be queued.

### 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).

**Public URL:** `https://${DOMAIN}/livekit` — the Caddy-proxied address. Never the internal `LIVEKIT_URL`. Instances without LiveKit can still receive federated calls by forwarding the host's URL and token to the client.

**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.

---

## 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.

---

## 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
```typescript
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 (disabled in Electron)
}
```

### 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)

---

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

**Screen share audio (when enabled):**
```typescript
{
  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