@camstack/addon-pipeline 0.1.20 → 0.2.0

package/dist/stream-broker/index.mjs.map

@@ -1 +0,0 @@
-{"version":3,"file":"index.mjs","sources":["../../src/stream-broker/stream-broker/decoder-session-proxy.ts","../../src/stream-broker/stream-broker/frame-handle-plane.ts","../../src/stream-broker/stream-broker/audio-chunk-plane.ts","../../src/stream-broker/rtsp/audio/aac-config.ts","../../src/stream-broker/stream-broker/audio-codec-session.ts","../../src/stream-broker/stream-broker/stream-pipe-server.ts","../../src/stream-broker/stream-broker/encoded-ring-buffer.ts","../../src/stream-broker/rtsp/rtsp-types.ts","../../src/stream-broker/rtsp/rtp-packetizer.ts","../../src/stream-broker/rtsp/annexb-deframer.ts","../../src/stream-broker/rtsp/sdp-builder.ts","../../src/stream-broker/rtsp/egress-audio-sdp.ts","../../src/stream-broker/rtsp/rtsp-restreamer.ts","../../src/stream-broker/rtsp/sdp-parser.ts","../../src/stream-broker/rtsp/rtsp-client.ts","../../src/stream-broker/rtsp/rfc4571-reader.ts","../../src/stream-broker/rtmp/flv-parser.ts","../../src/stream-broker/rtmp/rtmp-client.ts","../../src/stream-broker/rtmp/rtmp-reader.ts","../../src/stream-broker/rtsp/rtp-depacketizer.ts","../../src/stream-broker/rtsp/audio-rtp-decoder.ts","../../src/stream-broker/stream-broker/g711-mulaw.ts","../../src/stream-broker/stream-broker/placeholder-frames.ts","../../src/stream-broker/stream-broker/stream-broker.ts","../../src/stream-broker/stream-broker/audio-codec-proxy.ts","../../src/stream-broker/rtsp/rtsp-session.ts","../../src/stream-broker/rtsp/rtsp-listen-server.ts","../../src/stream-broker/rtsp/rtsp-restream-provider.ts","../../src/stream-broker/stream-broker/ffmpeg-args-common.ts","../../src/stream-broker/stream-broker/ffmpeg-invocation.ts","../../src/stream-broker/rtsp/udp-audio-rtp-receiver.ts","../../src/stream-broker/stream-broker/transcode-egress.ts","../../src/stream-broker/stream-broker/transcode-pipeline.ts","../../src/stream-broker/stream-broker/profile-restreamers.ts","../../src/stream-broker/stream-broker/stream-broker-manager.ts","../../src/stream-broker/encode-profile-resolver.ts","../../src/stream-broker/webrtc/webrtc-fallback-profile.ts","../../src/stream-broker/webrtc/nal-utils.ts","../../src/stream-broker/webrtc/h264-utils.ts","../../src/stream-broker/webrtc/h265-utils.ts","../../src/stream-broker/webrtc/mdns-resolve.ts","../../src/stream-broker/webrtc/jitter-buffer.ts","../../src/stream-broker/webrtc/h265-repacketizer.ts","../../src/stream-broker/webrtc/h264-repacketizer.ts","../../src/stream-broker/webrtc/session.ts","../../src/stream-broker/webrtc/loss-tracker.ts","../../src/stream-broker/webrtc/adaptive-controller.ts","../../src/stream-broker/webrtc/broker-webrtc-server.ts","../../src/stream-broker/device-scoped-providers.ts","../../src/stream-broker/addon.ts","../../src/stream-broker/stream-broker/frame-dropper.ts","../../src/stream-broker/stream-broker/ffmpeg-decoder-session.ts","../../src/stream-broker/stream-broker/ffmpeg-decoder-provider.ts","../../src/stream-broker/frame-transport/codecs/length-prefix-codec.ts","../../src/stream-broker/frame-transport/codecs/decoded-frame-codec.ts","../../src/stream-broker/frame-transport/transports/tcp-transport.ts","../../src/stream-broker/frame-transport/transports/uds-transport.ts","../../src/stream-broker/frame-transport/transports/inprocess-transport.ts","../../src/stream-broker/frame-transport/frame-server.ts","../../src/stream-broker/frame-transport/frame-client.ts","../../src/stream-broker/frame-transport/auth.ts"],"sourcesContent":["import type { EncodedPacket, DecodedFrame, DecoderSessionConfig, DecoderStats, FrameHandle } from '@camstack/types'\n\nexport interface DecoderCapApi {\n  supportsCodec(input: { readonly codec: string }): Promise<boolean>\n  getInfo(): Promise<{ readonly id: string; readonly name: string; readonly isPullMode?: boolean; readonly priority?: number }>\n  createSession(config: DecoderSessionConfig): Promise<{ readonly sessionId: string; readonly nodeId: string }>\n  pushPacket(input: { readonly sessionId: string; readonly packet: EncodedPacket }): Promise<void>\n  pullFrames(input: { readonly sessionId: string; readonly maxCount: number }): Promise<readonly DecodedFrame[]>\n  /**\n   * Drain the zero-pixel `FrameHandle[]` of a `frameSink: 'shm'` session\n   * (Phase 5 / D9). Returns `[]` for a `'callback'` session — a session is\n   * one mode or the other.\n   */\n  pullHandles(input: { readonly sessionId: string; readonly maxCount: number }): Promise<readonly FrameHandle[]>\n  destroySession(input: { readonly sessionId: string }): Promise<void>\n  openStream(input: { readonly sessionId: string; readonly url: string }): Promise<void>\n  updateConfig(input: { readonly sessionId: string; readonly config: Partial<DecoderSessionConfig> }): Promise<void>\n  getStats(input: { readonly sessionId: string }): Promise<DecoderStats>\n}\n\nexport class DecoderSessionProxy {\n  private polling = false\n\n  constructor(\n    private readonly api: DecoderCapApi,\n    readonly sessionId: string,\n  ) {}\n\n  async pushPacket(packet: EncodedPacket): Promise<void> {\n    await this.api.pushPacket({ sessionId: this.sessionId, packet })\n  }\n\n  async openStream(url: string): Promise<void> {\n    await this.api.openStream({ sessionId: this.sessionId, url })\n  }\n\n  async startPolling(onFrame: (frame: DecodedFrame) => void): Promise<void> {\n    this.polling = true\n    while (this.polling) {\n      const frames = await this.api.pullFrames({ sessionId: this.sessionId, maxCount: 4 })\n      for (const frame of frames) onFrame(frame)\n      if (frames.length === 0) await new Promise((r) => setTimeout(r, 1))\n    }\n  }\n\n  /**\n   * Poll a `frameSink: 'shm'` session for `FrameHandle`s (Phase 5 / D9).\n   * Mirrors `startPolling` but drains `pullHandles` — the decoder has\n   * already written the pixels into a shared-memory ring, so what crosses\n   * the cap boundary is the tiny serialisable handle. Runs until\n   * `stopPolling`, `destroy`, or the session is destroyed externally.\n   *\n   * When the decoder reports the session is gone (`unknown sessionId` /\n   * `Service … not found`) the loop exits gracefully — those errors are\n   * expected during decoder restart/shutdown and must not be re-thrown to\n   * the caller's `.catch()` handler, which would otherwise spin forever.\n   */\n  async startHandlePolling(onHandle: (handle: FrameHandle) => void): Promise<void> {\n    this.polling = true\n    while (this.polling) {\n      try {\n        const handles = await this.api.pullHandles({ sessionId: this.sessionId, maxCount: 4 })\n        for (const handle of handles) onHandle(handle)\n        if (handles.length === 0) await new Promise((r) => setTimeout(r, 1))\n      } catch (err) {\n        // Session destroyed externally — stop polling gracefully so the\n        // caller's .catch() doesn't spin forever.\n        this.polling = false\n        const msg = err instanceof Error ? err.message : String(err)\n        // Re-throw only for unexpected errors; session-gone errors are\n        // normal during decoder shutdown or restart.\n        if (!msg.includes('unknown sessionId') && !msg.includes('Service') && !msg.includes('not found')) {\n          throw err\n        }\n        return\n      }\n    }\n  }\n\n  stopPolling(): void {\n    this.polling = false\n  }\n\n  async updateConfig(config: Partial<DecoderSessionConfig>): Promise<void> {\n    await this.api.updateConfig({ sessionId: this.sessionId, config })\n  }\n\n  async getStats(): Promise<DecoderStats> {\n    return this.api.getStats({ sessionId: this.sessionId })\n  }\n\n  async destroy(): Promise<void> {\n    this.stopPolling()\n    await this.api.destroySession({ sessionId: this.sessionId })\n  }\n}\n","/**\n * `FrameHandlePlane` — the broker's shared-memory frame-handle surface\n * (Phase 5 / D9).\n *\n * This is the handle-based replacement for `StreamBroker.onDecodedFrame`'s\n * live-object callback path. Instead of fanning pixel `DecodedFrame`s out to\n * in-process callbacks (which cannot cross a process boundary), the broker\n * publishes zero-pixel, serialisable `FrameHandle`s a consumer drains over\n * tRPC and reads back from a shared-memory ring with a `FrameRingReader`.\n *\n * ## Ring-per-format model\n *\n * A `frameSink: 'shm'` decoder session produces exactly ONE pixel format\n * (the decoder's scaler is configured for it). The broker therefore keeps\n * one decoder session — hence one shm ring — per `(brokerId, format)` that a\n * consumer actually subscribes to:\n *\n *   - motion subscribes `gray`  → one `gray` decoder session + ring\n *   - detection subscribes `rgb` → one `rgb` decoder session + ring\n *\n * Only the formats consumers ask for are materialised. There is NO\n * broker-side `sharp` conversion any more — each consumer reads the exact\n * format it requested. A session is reference-counted by its subscriber set\n * and torn down when the last subscriber of that format leaves.\n *\n * ## fps throttling\n *\n * Throttling is implicit: the ring is latest-wins, so a slow consumer that\n * polls `pullFrameHandles` infrequently simply receives the newest handles\n * and the intervening frames are dropped at the ring (a recycled slot the\n * consumer never read). The optional `maxFps` from `subscribeFrames` is NOT\n * enforced here — it is echoed back to the consumer as a reader-side polling\n * cadence hint.\n *\n * ## Packet feed\n *\n * The plane's decoder sessions are push-mode by default: the broker forwards\n * every video `EncodedPacket` to `pushPacket`. For a pull-mode decoder the\n * broker supplies a `pullModeUrl` and the plane calls `openStream` once the\n * session exists.\n */\nimport { randomUUID } from 'node:crypto'\n\nimport { RingBuffer, errMsg } from '@camstack/types'\nimport type {\n  EncodedPacket,\n  FrameHandle,\n  FrameHandleFormat,\n  IScopedLogger,\n} from '@camstack/types'\n\nimport type { DecoderCapApi } from './decoder-session-proxy.js'\nimport { DecoderSessionProxy } from './decoder-session-proxy.js'\n\n/** Per-subscription handle-queue capacity. A handle is tiny; 64 absorbs a\n *  burst between two `pullFrameHandles` polls without unbounded growth. */\nconst HANDLE_QUEUE_CAPACITY = 64\n\n/** Default reader-side cadence hint when a subscriber omits `maxFps`. */\nconst DEFAULT_HINT_FPS = 5\n\n/** Rolling window for the aggregate `decodeFps` rate, ms. Mirrors the\n *  broker's 2s `inputFps` window so the two stats are comparable. */\nconst DECODE_FPS_WINDOW_MS = 2000\n\n/** A consumer's frame-handle subscription. */\ninterface HandleSubscription {\n  readonly id: string\n  readonly format: FrameHandleFormat\n  /** Reader-side cadence hint, frames/s — echoed to the consumer, not enforced. */\n  readonly maxFps: number\n  /** Short caller-identity tag for diagnostics. */\n  readonly tag: string\n  /** Ms epoch the subscription was opened. */\n  readonly subscribedAt: number\n  /** Latest-wins handle queue drained by `pullFrameHandles`. */\n  readonly queue: RingBuffer<FrameHandle>\n  framesDelivered: number\n}\n\n/** A diagnostic row for one frame-handle subscription (mirrors `BrokerDecodedClient`). */\nexport interface FrameHandleSubscriberInfo {\n  readonly tag: string\n  readonly subscribedAt: number\n  readonly maxFps: number\n  readonly framesDelivered: number\n}\n\n/**\n * One decoder session producing a single pixel format, shared by every\n * subscription of that format. Reference-counted by `subscriberIds`.\n */\ninterface FormatSession {\n  readonly format: FrameHandleFormat\n  /** The decoder session, or `null` before it has been created. */\n  proxy: DecoderSessionProxy | null\n  /** Moleculer nodeID of the decoder provider hosting `proxy`. */\n  nodeId: string | null\n  /** Subscription ids currently reading this format. */\n  readonly subscriberIds: Set<string>\n}\n\n/** What the broker tells the plane about the stream it is decoding. */\nexport interface FrameHandlePlaneStreamInfo {\n  /** Video codec — `h264` / `h265` / `hevc`. */\n  readonly codec: string\n  /** Numeric device id for decoder log scoping, when known. */\n  readonly numericDeviceId?: number\n  /** Decoder session log tag (`broker:<deviceId>/<profile>`). */\n  readonly tag: string\n  /**\n   * Local pipe URL for a pull-mode decoder. `null` for push-mode decoders\n   * (the broker feeds packets through `pushPacket`).\n   */\n  readonly pullModeUrl: string | null\n}\n\n/** Dependencies a `FrameHandlePlane` needs from its owning broker. */\nexport interface FrameHandlePlaneOptions {\n  readonly decoderApi: DecoderCapApi\n  readonly logger: IScopedLogger | undefined\n  /**\n   * Resolve the current stream descriptor at session-creation time. Called\n   * lazily so the plane always decodes with the freshest detected codec.\n   * Returns `null` when the broker has no stream to decode yet.\n   */\n  readonly resolveStreamInfo: () => FrameHandlePlaneStreamInfo | null\n}\n\n/**\n * Owns the broker's shared-memory frame-handle subscriptions and the\n * per-format `frameSink: 'shm'` decoder sessions that feed them.\n */\nexport class FrameHandlePlane {\n  private readonly decoderApi: DecoderCapApi\n  private readonly logger: IScopedLogger | undefined\n  private readonly resolveStreamInfo: () => FrameHandlePlaneStreamInfo | null\n\n  private readonly subscriptions = new Map<string, HandleSubscription>()\n  private readonly sessions = new Map<FrameHandleFormat, FormatSession>()\n  private disposed = false\n\n  /** Re-entrancy guard for `armPendingSessions` — collapses the per-packet\n   *  `pushPacket` calls into a single in-flight deferred-arm pass. */\n  private armInFlight = false\n\n  /** Aggregate decoded-frame rate state — frames fanned out in the current\n   *  rolling window, the window's start epoch, and the last computed fps. */\n  private framesInWindow = 0\n  private windowStartMs = Date.now()\n  private decodedFps = 0\n\n  constructor(options: FrameHandlePlaneOptions) {\n    this.decoderApi = options.decoderApi\n    this.logger = options.logger\n    this.resolveStreamInfo = options.resolveStreamInfo\n  }\n\n  /** Number of active handle subscriptions — surfaced for diagnostics. */\n  get subscriberCount(): number {\n    return this.subscriptions.size\n  }\n\n  /**\n   * Moleculer nodeIDs of the decoder providers currently hosting this plane's\n   * `frameSink: 'shm'` sessions. Used by the broker-manager's per-agent\n   * hwaccel-change handler to decide which brokers need a decoder rotation.\n   */\n  decoderNodeIds(): readonly string[] {\n    const ids: string[] = []\n    for (const session of this.sessions.values()) {\n      if (session.nodeId) ids.push(session.nodeId)\n    }\n    return ids\n  }\n\n  /**\n   * Rotate every shm decoder session hosted on `agentNodeId` — destroy and\n   * rebuild it so the new session re-pulls the latest per-agent decoder\n   * preferences (hwaccel backend, …). Subscriptions are preserved: the same\n   * `subscriptionId`s read from the rebuilt session's fresh ring. A no-op for\n   * sessions on other nodes. Returns the number of sessions rotated.\n   */\n  async rotate(agentNodeId: string, reason: string): Promise<number> {\n    if (this.disposed) return 0\n    let rotated = 0\n    for (const session of this.sessions.values()) {\n      const sessionAgent = session.nodeId && session.nodeId.includes('/')\n        ? session.nodeId.split('/')[0]\n        : session.nodeId\n      if (sessionAgent !== agentNodeId) continue\n      this.logger?.info('frame-handle plane: rotating shm decoder session', {\n        meta: { format: session.format, reason, nodeId: session.nodeId },\n      })\n      await this.destroySession(session)\n      await this.startSessionDecoder(session)\n      rotated += 1\n    }\n    return rotated\n  }\n\n  /**\n   * Register a frame-handle subscription. Spins up (or reuses) a\n   * `frameSink: 'shm'` decoder session producing `format`. The returned\n   * `subscriptionId` is the handle the consumer passes to\n   * `pullFrameHandles` / `unsubscribe`.\n   */\n  async subscribe(input: {\n    readonly format: FrameHandleFormat\n    readonly maxFps?: number\n    readonly tag?: string\n  }): Promise<{ subscriptionId: string; maxFps: number }> {\n    if (this.disposed) {\n      throw new Error('FrameHandlePlane: subscribe after dispose')\n    }\n    const subscriptionId = `fh-${randomUUID()}`\n    const maxFps = input.maxFps !== undefined && input.maxFps > 0\n      ? input.maxFps\n      : DEFAULT_HINT_FPS\n    const subscription: HandleSubscription = {\n      id: subscriptionId,\n      format: input.format,\n      maxFps,\n      tag: input.tag ?? 'unknown',\n      subscribedAt: Date.now(),\n      queue: new RingBuffer<FrameHandle>(HANDLE_QUEUE_CAPACITY),\n      framesDelivered: 0,\n    }\n    this.subscriptions.set(subscriptionId, subscription)\n\n    await this.ensureSession(input.format, subscriptionId)\n\n    this.logger?.info('frame-handle subscription added', {\n      meta: { subscriptionId, format: input.format, tag: subscription.tag, maxFps },\n    })\n    return { subscriptionId, maxFps }\n  }\n\n  /**\n   * Drain up to `maxCount` `FrameHandle`s for a subscription, latest-wins.\n   * An unknown subscription id returns `[]` (the consumer may have been\n   * torn down concurrently).\n   */\n  pullHandles(subscriptionId: string, maxCount: number): readonly FrameHandle[] {\n    const subscription = this.subscriptions.get(subscriptionId)\n    if (!subscription) return []\n    // Latest-wins (for real): frames are fanned in at the FULL decode rate but\n    // a consumer only pulls `maxCount` per poll. The ring drops oldest on\n    // overflow, but `drain` reads oldest-first — so a consumer slower than the\n    // decode rate would perpetually read the BACK of a full 64-deep ring, i.e.\n    // frames ~ringDepth/fps old (~2.4s at 25fps), even though downstream queues\n    // are latest-only. Discard the stale backlog beyond `maxCount` so the\n    // consumer always receives the FRESHEST handles; the skipped slots are\n    // simply not read (the decoder recycles them).\n    const stale = subscription.queue.size - maxCount\n    if (stale > 0) subscription.queue.drain(stale)\n    const handles = subscription.queue.drain(maxCount)\n    subscription.framesDelivered += handles.length\n    return handles\n  }\n\n  /**\n   * Release a subscription. When it was the last reader of its format the\n   * underlying decoder session is destroyed (and its shm segment unlinked\n   * by the decoder). Returns `true` when a known subscription was released.\n   */\n  async unsubscribe(subscriptionId: string): Promise<boolean> {\n    const subscription = this.subscriptions.get(subscriptionId)\n    if (!subscription) return false\n    this.subscriptions.delete(subscriptionId)\n\n    const session = this.sessions.get(subscription.format)\n    if (session) {\n      session.subscriberIds.delete(subscriptionId)\n      if (session.subscriberIds.size === 0) {\n        this.sessions.delete(subscription.format)\n        await this.destroySession(session)\n      }\n    }\n    this.logger?.info('frame-handle subscription removed', {\n      meta: { subscriptionId, format: subscription.format },\n    })\n    return true\n  }\n\n  /**\n   * Forward a video `EncodedPacket` to every active shm decoder session.\n   * Push-mode decoders consume this; a pull-mode decoder ignores it (it\n   * reads its own pipe via `openStream`). Audio packets are not forwarded.\n   *\n   * The packet feed doubles as the plane's stream-ready signal: the broker\n   * only calls `pushPacket` once it has a source and the first keyframe has\n   * landed — exactly the point `resolveStreamInfo` starts returning a\n   * descriptor. Each packet therefore triggers `armPendingSessions`, which\n   * re-attempts session creation for any deferred subscription so a lone\n   * first subscriber on a not-yet-started broker self-arms without a\n   * redundant re-`subscribe`.\n   */\n  pushPacket(packet: EncodedPacket): void {\n    if (this.disposed || packet.type !== 'video') return\n    void this.armPendingSessions(packet).catch((err: unknown) => {\n      this.logger?.warn('frame-handle plane: arm pending sessions error', {\n        meta: { error: errMsg(err) },\n      })\n    })\n    this.feedSessions(packet)\n  }\n\n  /** Forward a video `EncodedPacket` to every live shm decoder session. */\n  private feedSessions(packet: EncodedPacket): void {\n    for (const session of this.sessions.values()) {\n      if (!session.proxy) continue\n      session.proxy.pushPacket(packet).catch((err: unknown) => {\n        const msg = errMsg(err)\n        this.logger?.warn('frame-handle plane: decoder push error', {\n          meta: { format: session.format, error: msg },\n        })\n        // Self-heal: when the decoder reports the session is gone (crash,\n        // restart, or shutdown), invalidate the stale proxy so we don't\n        // flood with thousands of warnings per second. `armPendingSessions`\n        // on the next packet will recreate a fresh session.\n        if (msg.includes('unknown sessionId') || msg.includes('Service') || msg.includes('not found')) {\n          this.logger?.info('frame-handle plane: invalidating stale proxy', {\n            meta: { format: session.format },\n          })\n          // Best-effort cleanup: the decoder is already gone, so\n          // destroySession may also fail — that's fine.\n          void this.destroySession(session).catch(() => {})\n          this.sessions.delete(session.format)\n        }\n      })\n    }\n  }\n\n  /**\n   * Re-attempt session creation for every subscription whose format has no\n   * live `FormatSession` yet — the subscriptions whose `startSessionDecoder`\n   * was deferred because the broker had no stream at `subscribe` time. Driven\n   * off the `pushPacket` feed (event-driven, no polling timer); a no-op once\n   * every subscribed format already owns a session.\n   *\n   * `armInFlight` collapses re-entrant calls: `pushPacket` fires per packet,\n   * but `ensureSession` is async — without the guard a burst of packets would\n   * launch overlapping decoder-creation round-trips for the same format.\n   *\n   * `triggerPacket` is the packet whose arrival armed the pass. Because\n   * `ensureSession` is async, the synchronous `feedSessions` in `pushPacket`\n   * runs before the session lands in `this.sessions` — so this method feeds\n   * the triggering packet to each session it just created, otherwise that\n   * first (keyframe-carrying) packet would be lost and a push-mode H264/H265\n   * decoder would stall waiting for SPS/PPS.\n   */\n  private async armPendingSessions(triggerPacket: EncodedPacket): Promise<void> {\n    if (this.disposed || this.armInFlight) return\n    // Collect the formats that still lack a session — skip the formats that\n    // already have a live one so we never re-run `ensureSession` for them.\n    const pending = new Map<FrameHandleFormat, string>()\n    for (const subscription of this.subscriptions.values()) {\n      if (this.sessions.has(subscription.format)) continue\n      if (!pending.has(subscription.format)) {\n        pending.set(subscription.format, subscription.id)\n      }\n    }\n    if (pending.size === 0) return\n\n    this.armInFlight = true\n    try {\n      for (const [format, subscriptionId] of pending) {\n        if (this.disposed) return\n        await this.ensureSession(format, subscriptionId)\n        const armed = this.sessions.get(format)\n        // Seed the freshly-armed session with the triggering packet so its\n        // decoder receives the keyframe that drove the arm.\n        if (armed?.proxy) {\n          armed.proxy.pushPacket(triggerPacket).catch((err: unknown) => {\n            this.logger?.warn('frame-handle plane: decoder push error', {\n              meta: { format, error: errMsg(err) },\n            })\n          })\n        }\n      }\n    } finally {\n      this.armInFlight = false\n    }\n  }\n\n  /**\n   * Aggregate decoded-frame rate across every `frameSink: 'shm'` session in\n   * this plane, frames/s. Each `fanoutHandle` call is one decoded frame\n   * delivered into the plane; the rate is recomputed over a rolling\n   * `DECODE_FPS_WINDOW_MS` window. Returns the last computed value between\n   * window rolls, and `0` while no session is producing.\n   */\n  decodeFps(): number {\n    this.rollDecodeFpsWindow()\n    return this.decodedFps\n  }\n\n  /**\n   * Roll the decoded-fps window when it has elapsed: convert frames seen in\n   * the window into a per-second rate, then reset the counter. Called both\n   * on every fanout and on every `decodeFps()` read so a stalled session\n   * (no fanout) decays its rate to `0` rather than reporting a stale value.\n   */\n  private rollDecodeFpsWindow(): void {\n    const now = Date.now()\n    const windowMs = now - this.windowStartMs\n    if (windowMs < DECODE_FPS_WINDOW_MS) return\n    this.decodedFps = (this.framesInWindow / windowMs) * 1000\n    this.framesInWindow = 0\n    this.windowStartMs = now\n  }\n\n  /** Diagnostic snapshot of every active frame-handle subscription. */\n  listSubscribers(): readonly FrameHandleSubscriberInfo[] {\n    return [...this.subscriptions.values()].map((s) => ({\n      tag: s.tag,\n      subscribedAt: s.subscribedAt,\n      maxFps: s.maxFps,\n      framesDelivered: s.framesDelivered,\n    }))\n  }\n\n  /**\n   * Force-release every subscription whose tag matches `tag`. Returns the\n   * number released. Decoder sessions wind down when their last subscriber\n   * leaves, same as a normal `unsubscribe`.\n   */\n  async killByTag(tag: string): Promise<number> {\n    const victims = [...this.subscriptions.values()]\n      .filter((s) => s.tag === tag)\n      .map((s) => s.id)\n    for (const id of victims) {\n      await this.unsubscribe(id)\n    }\n    return victims.length\n  }\n\n  /** Tear down every subscription + decoder session. Idempotent. */\n  async dispose(): Promise<void> {\n    if (this.disposed) return\n    this.disposed = true\n    this.subscriptions.clear()\n    const sessions = [...this.sessions.values()]\n    this.sessions.clear()\n    await Promise.all(sessions.map((s) => this.destroySession(s)))\n  }\n\n  // ── Internal ───────────────────────────────────────────────────────\n\n  /**\n   * Ensure a `frameSink: 'shm'` decoder session for `format` exists and add\n   * `subscriptionId` to its reader set. Reuses the session when one already\n   * runs for that format.\n   *\n   * When a session is created fresh, its reader set is seeded with EVERY\n   * subscription currently reading `format`, not just `subscriptionId` — a\n   * deferred-arm pass (`armPendingSessions`) may create the session long\n   * after several subscriptions of that format have registered, and the\n   * session is reference-counted by `subscriberIds`. Seeding only the one id\n   * would let an `unsubscribe` of that id tear the session down while its\n   * sibling subscriptions still read it.\n   */\n  private async ensureSession(\n    format: FrameHandleFormat,\n    subscriptionId: string,\n  ): Promise<void> {\n    const existing = this.sessions.get(format)\n    if (existing) {\n      existing.subscriberIds.add(subscriptionId)\n      return\n    }\n\n    const subscriberIds = new Set<string>([subscriptionId])\n    for (const subscription of this.subscriptions.values()) {\n      if (subscription.format === format) subscriberIds.add(subscription.id)\n    }\n    const session: FormatSession = {\n      format,\n      proxy: null,\n      nodeId: null,\n      subscriberIds,\n    }\n    const started = await this.startSessionDecoder(session)\n    if (started) {\n      this.sessions.set(format, session)\n    }\n  }\n\n  /**\n   * Create the `frameSink: 'shm'` decoder for a `FormatSession` and start\n   * draining its handle queue. Mutates `session.proxy` / `session.nodeId` in\n   * place so a `rotate()` can rebuild a registered session without disturbing\n   * its `subscriberIds`. Returns `false` when no stream / unsupported codec\n   * means no decoder could be created.\n   */\n  private async startSessionDecoder(session: FormatSession): Promise<boolean> {\n    const { format } = session\n    const info = this.resolveStreamInfo()\n    if (!info) {\n      // No stream yet — the subscription is registered but receives no handles\n      // until a session exists. The plane self-arms: `armPendingSessions`\n      // (driven off the `pushPacket` feed) re-attempts this once the broker\n      // starts pushing packets, so a lone pending subscriber is not stranded.\n      this.logger?.info('frame-handle plane: no stream — session deferred', {\n        meta: { format },\n      })\n      return false\n    }\n\n    const supported = await this.decoderApi.supportsCodec({ codec: info.codec })\n    if (!supported) {\n      this.logger?.warn('frame-handle plane: codec unsupported — session skipped', {\n        meta: { format, codec: info.codec },\n      })\n      return false\n    }\n\n    const { sessionId, nodeId } = await this.decoderApi.createSession({\n      codec: info.codec,\n      maxFps: 0,\n      outputFormat: format,\n      scale: 1,\n      frameSink: 'shm',\n      ...(info.numericDeviceId !== undefined ? { deviceId: info.numericDeviceId } : {}),\n      tag: `${info.tag}:shm:${format}`,\n    })\n    const proxy = new DecoderSessionProxy(this.decoderApi, sessionId)\n    session.proxy = proxy\n    session.nodeId = nodeId\n\n    // Drain the decoder's shm handle queue and fan each handle into every\n    // subscription reading this format.\n    proxy.startHandlePolling((handle) => {\n      this.fanoutHandle(format, handle)\n    }).catch((err: unknown) => {\n      this.logger?.warn('frame-handle plane: handle polling error', {\n        meta: { format, error: errMsg(err) },\n      })\n    })\n\n    if (info.pullModeUrl) {\n      proxy.openStream(info.pullModeUrl).catch((err: unknown) => {\n        this.logger?.error('frame-handle plane: pull-mode openStream failed', {\n          meta: { format, error: errMsg(err) },\n        })\n      })\n    }\n\n    this.logger?.info('frame-handle plane: shm decoder session created', {\n      meta: { format, codec: info.codec, sessionId, nodeId },\n    })\n    return true\n  }\n\n  /** Push one decoded `FrameHandle` to every subscription of `format`. */\n  private fanoutHandle(format: FrameHandleFormat, handle: FrameHandle): void {\n    // One fanout call == one decoded frame delivered into the plane.\n    this.rollDecodeFpsWindow()\n    this.framesInWindow += 1\n    for (const subscription of this.subscriptions.values()) {\n      if (subscription.format !== format) continue\n      // Latest-wins: a full queue overwrites its oldest handle, so a slow\n      // consumer always pulls the newest frames and silently drops the rest.\n      subscription.queue.push(handle)\n    }\n  }\n\n  /** Destroy a decoder session, swallowing teardown errors. */\n  private async destroySession(session: FormatSession): Promise<void> {\n    const proxy = session.proxy\n    session.proxy = null\n    session.nodeId = null\n    if (!proxy) return\n    try {\n      await proxy.destroy()\n    } catch (err) {\n      this.logger?.warn('frame-handle plane: session destroy failed', {\n        meta: { format: session.format, error: errMsg(err) },\n      })\n    }\n  }\n}\n","/**\n * `AudioChunkPlane` — the broker's serialisable decoded audio-chunk surface\n * (Phase 5 / D9).\n *\n * This is the poll-based replacement for `StreamBroker.onDecodedAudioChunk`'s\n * live-object callback path. A live callback cannot cross a process boundary;\n * once the `pipeline` group is dissolved (Task 8) the audio consumer\n * (`pipeline-orchestrator`) runs in a different process from the broker, so\n * audio delivery must go over tRPC like the video frame plane.\n *\n * ## Why no shared memory\n *\n * Unlike video frames (~6 MB at 1080p RGB), a decoded audio chunk is tiny — a\n * ~500ms `AudioCodecSession` window is a few KB of Float32 PCM. Tiny payloads\n * serialise over tRPC cheaply, so audio ships its bytes INLINE on the RPC\n * wire; there is no shm ring.\n *\n * ## FIFO / no-drop queue policy\n *\n * Audio wants FIFO arrival order with no latest-wins drop: a missing chunk is\n * an audible gap and breaks the audio analyzer's fixed-length analysis\n * window. Each subscription therefore owns a generously-sized FIFO queue\n * (`AUDIO_QUEUE_CAPACITY`). Chunks arrive roughly every\n * `AudioCodecSession`-window (~500ms), so the queue buffers tens of seconds\n * of audio — a healthy poll loop never fills it. The queue only drops its\n * OLDEST chunk if a truly stalled consumer lets it overflow (a bounded safety\n * valve, not the steady-state behaviour).\n */\nimport { randomUUID } from 'node:crypto'\n\nimport { RingBuffer } from '@camstack/types'\nimport type { DecodedAudioChunk, IScopedLogger } from '@camstack/types'\n\n/**\n * Per-subscription FIFO capacity. A chunk is a few KB and arrives ~every\n * 500ms, so 64 buffers ~30s of audio between two `pullAudioChunks` polls\n * without unbounded growth.\n */\nconst AUDIO_QUEUE_CAPACITY = 64\n\n/** A consumer's decoded audio-chunk subscription. */\ninterface AudioChunkSubscription {\n  readonly id: string\n  /** Short caller-identity tag for diagnostics (`audio-analyzer`, …). */\n  readonly tag: string\n  /** Ms epoch the subscription was opened. */\n  readonly subscribedAt: number\n  /** Bounded FIFO chunk queue drained by `pullAudioChunks`. */\n  readonly queue: RingBuffer<DecodedAudioChunk>\n  chunksDelivered: number\n}\n\n/** A diagnostic row for one audio-chunk subscription (mirrors `BrokerAudioClient`). */\nexport interface AudioChunkSubscriberInfo {\n  readonly tag: string\n  readonly subscribedAt: number\n  readonly chunksDelivered: number\n}\n\n/**\n * Owns the broker's decoded audio-chunk subscriptions. The broker feeds every\n * decoded `DecodedAudioChunk` into `fanout`; each subscription's poll loop\n * drains its own FIFO queue via `pull`.\n */\nexport class AudioChunkPlane {\n  private readonly logger: IScopedLogger | undefined\n  private readonly subscriptions = new Map<string, AudioChunkSubscription>()\n\n  constructor(logger?: IScopedLogger) {\n    this.logger = logger\n  }\n\n  /** Number of active audio-chunk subscriptions — surfaced as audio demand. */\n  get subscriberCount(): number {\n    return this.subscriptions.size\n  }\n\n  /**\n   * Register an audio-chunk subscription. The returned `subscriptionId` is\n   * the handle the consumer passes to `pull` / `unsubscribe`.\n   */\n  subscribe(input?: { readonly tag?: string }): { subscriptionId: string } {\n    const subscriptionId = `ac-${randomUUID()}`\n    const subscription: AudioChunkSubscription = {\n      id: subscriptionId,\n      tag: input?.tag ?? 'unknown',\n      subscribedAt: Date.now(),\n      queue: new RingBuffer<DecodedAudioChunk>(AUDIO_QUEUE_CAPACITY),\n      chunksDelivered: 0,\n    }\n    this.subscriptions.set(subscriptionId, subscription)\n    this.logger?.info('audio-chunk subscription added', {\n      meta: { subscriptionId, tag: subscription.tag },\n    })\n    return { subscriptionId }\n  }\n\n  /**\n   * Drain up to `maxCount` `DecodedAudioChunk`s for a subscription, in FIFO\n   * arrival order. An unknown subscription id returns `[]` (the consumer may\n   * have been torn down concurrently).\n   */\n  pull(subscriptionId: string, maxCount: number): readonly DecodedAudioChunk[] {\n    const subscription = this.subscriptions.get(subscriptionId)\n    if (!subscription) return []\n    const chunks = subscription.queue.drain(maxCount)\n    subscription.chunksDelivered += chunks.length\n    return chunks\n  }\n\n  /** Release a subscription. Returns `true` when a known subscription was released. */\n  unsubscribe(subscriptionId: string): boolean {\n    const existed = this.subscriptions.delete(subscriptionId)\n    if (existed) {\n      this.logger?.info('audio-chunk subscription removed', {\n        meta: { subscriptionId },\n      })\n    }\n    return existed\n  }\n\n  /**\n   * Fan one decoded `DecodedAudioChunk` into every active subscription's FIFO\n   * queue. A full queue drops its oldest chunk (bounded safety valve for a\n   * stalled consumer). No-op when there are no subscriptions.\n   */\n  fanout(chunk: DecodedAudioChunk): void {\n    for (const subscription of this.subscriptions.values()) {\n      subscription.queue.push(chunk)\n    }\n  }\n\n  /** Diagnostic snapshot of every active audio-chunk subscription. */\n  listSubscribers(): readonly AudioChunkSubscriberInfo[] {\n    return [...this.subscriptions.values()].map((s) => ({\n      tag: s.tag,\n      subscribedAt: s.subscribedAt,\n      chunksDelivered: s.chunksDelivered,\n    }))\n  }\n\n  /**\n   * Force-release every subscription whose tag matches `tag`. Returns the\n   * number released.\n   */\n  killByTag(tag: string): number {\n    const victims = [...this.subscriptions.values()]\n      .filter((s) => s.tag === tag)\n      .map((s) => s.id)\n    for (const id of victims) {\n      this.subscriptions.delete(id)\n    }\n    return victims.length\n  }\n\n  /** Tear down every subscription. Idempotent. */\n  dispose(): void {\n    this.subscriptions.clear()\n  }\n}\n","/**\n * Helpers for AAC stream framing.\n *\n * Two pieces:\n *\n * 1. `parseAudioSpecificConfig` — decode the hex `config=` blob from the\n *    SDP `fmtp:` line for `mpeg4-generic`/`MP4A-LATM`. Returns the\n *    samplerate + channel config the camera advertises.\n *\n * 2. `depacketizeRfc3640` — strip AU-headers from an RTP `mode=AAC-hbr`\n *    payload and return one or more raw AAC AccessUnits ready to feed\n *    libavcodec. Reolink uses `sizeLength=13; indexLength=3` which is\n *    the RFC 3640 default for AAC-hbr.\n *\n * Both are pure functions over `Uint8Array` — keep them free of\n * libav/node-av deps so they're easy to unit-test.\n */\n\n// AAC sample-rate frequency table (ISO/IEC 14496-3 §1.6.3.4).\nconst SAMPLERATE_TABLE: readonly number[] = [\n  96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,\n  16000, 12000, 11025,  8000,  7350,\n]\n\nexport interface AudioSpecificConfig {\n  /** Object type (e.g. 2 = AAC-LC, 5 = HE-AAC, 29 = HE-AAC v2). */\n  readonly objectType: number\n  /** Decoded sample rate in Hz. */\n  readonly sampleRate: number\n  /** Channel configuration (1 = mono, 2 = stereo, 6 = 5.1, …). */\n  readonly channelConfig: number\n}\n\nclass BitReader {\n  private bitPos = 0\n  constructor(private readonly buf: Uint8Array) {}\n  read(n: number): number {\n    let v = 0\n    for (let i = 0; i < n; i++) {\n      const bytePos = this.bitPos >> 3\n      const bit = 7 - (this.bitPos & 7)\n      const b = bytePos < this.buf.length ? this.buf[bytePos]! : 0\n      v = (v << 1) | ((b >> bit) & 1)\n      this.bitPos++\n    }\n    return v\n  }\n  remainingBits(): number {\n    return this.buf.length * 8 - this.bitPos\n  }\n}\n\n/**\n * Decode hex `config=` blob (or raw `Uint8Array`) into an\n * `AudioSpecificConfig`. Throws on malformed input.\n *\n * Layout per ISO/IEC 14496-3 §1.6.2.1:\n *   - 5 bits: audioObjectType (extension-escape if 31)\n *   - 4 bits: samplingFrequencyIndex (0xf = explicit 24-bit value follows)\n *   - 4 bits: channelConfiguration\n */\nexport function parseAudioSpecificConfig(input: string | Uint8Array): AudioSpecificConfig {\n  const buf = typeof input === 'string' ? hexToBytes(input) : input\n  if (buf.length < 2) {\n    throw new Error(`audio-codec: AudioSpecificConfig too short (${buf.length} bytes)`)\n  }\n  const reader = new BitReader(buf)\n  let objectType = reader.read(5)\n  if (objectType === 31) {\n    objectType = 32 + reader.read(6)\n  }\n  const sampleRateIndex = reader.read(4)\n  let sampleRate: number\n  if (sampleRateIndex === 0xf) {\n    sampleRate = reader.read(24)\n  } else {\n    const v = SAMPLERATE_TABLE[sampleRateIndex]\n    if (!v) throw new Error(`audio-codec: invalid samplingFrequencyIndex ${sampleRateIndex}`)\n    sampleRate = v\n  }\n  const channelConfig = reader.read(4)\n  return { objectType, sampleRate, channelConfig }\n}\n\nfunction hexToBytes(hex: string): Uint8Array {\n  const trimmed = hex.replace(/\\s+/g, '')\n  if (trimmed.length % 2 !== 0) {\n    throw new Error(`audio-codec: hex string has odd length (${trimmed.length})`)\n  }\n  const out = new Uint8Array(trimmed.length / 2)\n  for (let i = 0; i < out.length; i++) {\n    const byte = parseInt(trimmed.slice(i * 2, i * 2 + 2), 16)\n    if (Number.isNaN(byte)) {\n      throw new Error(`audio-codec: invalid hex at offset ${i * 2}`)\n    }\n    out[i] = byte\n  }\n  return out\n}\n\n/**\n * Depacketize an RFC 3640 `mode=AAC-hbr` RTP payload (typical\n * `sizeLength=13; indexLength=3`) into one or more raw AAC AccessUnits.\n *\n * RFC 3640 §3.2 layout:\n *   AU Headers Length (16 bits, big-endian) — total bits in the AU\n *     headers section.\n *   AU headers — `sizeLength` bits AU-size, `indexLength` bits AU-index\n *     (or AU-index-delta after the first). Padding aligns to byte.\n *   AU data — `auCount` access units back-to-back, each `auSize` bytes.\n *\n * `payload` is the RTP payload (after the 12+ byte RTP header, i.e.\n * what callers pass via `onAudioRtp` minus the header strip).\n */\nexport interface Rfc3640Options {\n  readonly sizeLength: number    // bits per AU-size — default 13\n  readonly indexLength: number   // bits per AU-index — default 3\n  readonly indexDeltaLength?: number\n}\n\nexport function depacketizeRfc3640(\n  payload: Uint8Array,\n  opts: Rfc3640Options = { sizeLength: 13, indexLength: 3 },\n): Uint8Array[] {\n  if (payload.length < 2) return []\n  const sizeLen = opts.sizeLength\n  const idxLen = opts.indexLength\n  const idxDeltaLen = opts.indexDeltaLength ?? idxLen\n  const headersBitLength = (payload[0]! << 8) | payload[1]!\n  if (headersBitLength === 0) return []\n\n  const headersStartByte = 2\n  const headersByteLength = (headersBitLength + 7) >> 3\n  const headersEndByte = headersStartByte + headersByteLength\n  if (headersEndByte > payload.length) return []\n\n  const headersSlice = payload.subarray(headersStartByte, headersEndByte)\n  const reader = new BitReader(headersSlice)\n  const sizes: number[] = []\n  let consumed = 0\n  let firstAu = true\n  while (consumed < headersBitLength) {\n    if (reader.remainingBits() < sizeLen + (firstAu ? idxLen : idxDeltaLen)) break\n    const auSize = reader.read(sizeLen)\n    reader.read(firstAu ? idxLen : idxDeltaLen) // index/delta — ignored\n    sizes.push(auSize)\n    consumed += sizeLen + (firstAu ? idxLen : idxDeltaLen)\n    firstAu = false\n  }\n\n  const units: Uint8Array[] = []\n  let offset = headersEndByte\n  for (const sz of sizes) {\n    if (offset + sz > payload.length) break\n    units.push(payload.subarray(offset, offset + sz))\n    offset += sz\n  }\n  return units\n}\n","/**\n * Per-broker handle around an audio-codec decode session. Owns:\n *   - sessionId (lazily created on first push)\n *   - depacketizer choice based on RTP payload format\n *   - polling timer that pulls PCM from the cap and fans it out via\n *     a caller-supplied AudioRtpDecoder-shaped callback.\n *\n * Stays self-contained so the broker only needs to forward\n * `processPacket(rtpData)` and `close()` calls. Mirrors the surface\n * of `AudioRtpDecoder` (the legacy PCMU/PCMA path) so the broker's\n * call sites can swap implementations without further branching.\n */\nimport type { DecodedAudioChunk, IScopedLogger } from '@camstack/types'\nimport { errMsg } from '@camstack/types'\nimport type { AudioCodecCapApi } from './audio-codec-proxy.js'\nimport { depacketizeRfc3640, type Rfc3640Options } from '../rtsp/audio/aac-config.js'\n\nexport interface AudioCodecSessionConfig {\n  readonly codec: string                      // canonical lowercase: 'aac', 'opus', ...\n  readonly sourceSampleRate: number\n  readonly sourceChannels: number\n  readonly extraData?: Uint8Array             // AAC AudioSpecificConfig\n  readonly targetSampleRate: number           // typically 16000 for ASA / yamnet\n  readonly targetChannels: number             // typically 1\n  readonly tag?: string\n  /** Set when the source uses MPEG4-GENERIC mode=AAC-hbr framing. */\n  readonly rfc3640?: Rfc3640Options\n}\n\nconst POLL_INTERVAL_MS = 40\nconst MAX_PCM_PER_POLL = 16\nconst RTP_HEADER_LENGTH = 12\nconst WARN_THROTTLE_MS = 5_000\n// Audio chunks are emitted at the codec's native frame cadence (~21\n// to 128 ms depending on codec) — no accumulator window here. This\n// path used to coalesce into 500 ms chunks to match YAMNet/Apple\n// SoundAnalysis input windows; the side-effect was ~500 ms added\n// latency on every WebRTC playback (cam 15 / Reolink AAC chain).\n// Realtime consumers (WebRTC PCMU egress) now see audio with sub-\n// 100 ms decode latency. ML analyzers (YAMNet 480 ms hop / Apple SA\n// 960 ms window) accumulate at their own pipeline side; the\n// AudioChunkPlane is pull-based so they can buffer however many\n// small chunks they need before invoking `classify()`.\n// Matches the \"decode session 'dec-...' not found\" thrown by the\n// audio-codec addon when the upstream session has been reaped (idle\n// timeout, addon restart, host process resumed from sleep). We use it\n// to detect \"the cap forgot us\" vs a transient decode error.\nconst SESSION_NOT_FOUND_RE = /decode session\\b.*\\bnot found\\b/\n\nfunction isSessionGone(err: unknown): boolean {\n  return SESSION_NOT_FOUND_RE.test(errMsg(err))\n}\n\nexport class AudioCodecSession {\n  private sessionId: string | null = null\n  private sessionNodeId: string | null = null\n  private creating: Promise<string | null> | null = null\n  private pollTimer: ReturnType<typeof setInterval> | null = null\n  private closed = false\n  private packetsIn = 0\n  private packetsDropped = 0\n  // Coalesce warnings — without this a single reaped session produces\n  // ~1000 warn/s (every RTP audio packet + every poll tick), and the\n  // synchronous winston writes block the broker's event loop enough\n  // to visibly slow the video forwarders sharing the process.\n  private warnState = new Map<string, { lastLoggedAt: number; suppressed: number }>()\n  // No accumulator: each decoded PCM frame is emitted as one\n  // DecodedAudioChunk immediately for minimum WebRTC latency. ML\n  // consumers downstream apply their own window aggregation.\n\n  constructor(\n    private readonly api: AudioCodecCapApi,\n    private readonly cfg: AudioCodecSessionConfig,\n    private readonly emit: (chunk: DecodedAudioChunk) => void,\n    private readonly logger?: IScopedLogger,\n  ) {\n    // No chunk-byte target — emit happens per codec frame.\n  }\n\n  processPacket(rtpData: Buffer): void {\n    if (this.closed) return\n    if (rtpData.length <= RTP_HEADER_LENGTH) return\n    const payload = rtpData.subarray(RTP_HEADER_LENGTH)\n    this.packetsIn++\n\n    void this.ensureSession().then((sid) => {\n      if (!sid || this.closed) return\n      const nodeId = this.sessionNodeId ?? undefined\n      const units = this.cfg.rfc3640\n        ? depacketizeRfc3640(payload, this.cfg.rfc3640)\n        : [Uint8Array.from(payload)]\n      for (const unit of units) {\n        void this.api.pushEncodedFrame({\n          sessionId: sid,\n          ...(nodeId ? { nodeId } : {}),\n          data: unit,\n        }).catch((err) => {\n          this.packetsDropped++\n          this.handlePushError('audio-codec: pushEncodedFrame failed', sid, err)\n        })\n      }\n    })\n  }\n\n  /**\n   * Push a single pre-depacketized codec frame. Used by push-source\n   * providers (Reolink Baichuan, Frigate) that emit raw codec frames\n   * directly — no RTP wrapping, no rfc3640 depacketize. The payload\n   * MUST be one complete frame the underlying decoder can consume\n   * (e.g. an AAC raw frame with or without ADTS header per the\n   * codec's AudioSpecificConfig).\n   */\n  pushRawFrame(unit: Uint8Array): void {\n    if (this.closed) return\n    if (unit.length === 0) return\n    this.packetsIn++\n    void this.ensureSession().then((sid) => {\n      if (!sid || this.closed) return\n      const nodeId = this.sessionNodeId ?? undefined\n      void this.api.pushEncodedFrame({\n        sessionId: sid,\n        ...(nodeId ? { nodeId } : {}),\n        data: unit,\n      }).catch((err) => {\n        this.packetsDropped++\n        this.handlePushError('audio-codec: pushEncodedFrame (raw) failed', sid, err)\n      })\n    })\n  }\n\n  /** Stats surfaced by the broker for the streams tab / logs. */\n  getStats(): { packetsIn: number; packetsDropped: number; sessionId: string | null } {\n    return {\n      packetsIn: this.packetsIn,\n      packetsDropped: this.packetsDropped,\n      sessionId: this.sessionId,\n    }\n  }\n\n  async close(): Promise<void> {\n    if (this.closed) return\n    this.closed = true\n    if (this.pollTimer) {\n      clearInterval(this.pollTimer)\n      this.pollTimer = null\n    }\n    // No accumulator to flush — each PCM frame was emitted as it\n    // came back from `pullPcm`. close() proceeds directly to the\n    // upstream session teardown.\n    if (this.sessionId) {\n      const sid = this.sessionId\n      const nodeId = this.sessionNodeId ?? undefined\n      try { await this.api.closeSession({ sessionId: sid, ...(nodeId ? { nodeId } : {}) }) }\n      catch (err) {\n        this.logger?.warn('audio-codec: closeSession failed', {\n          meta: { error: errMsg(err) },\n        })\n      }\n      this.sessionId = null\n      this.sessionNodeId = null\n    }\n  }\n\n  private async ensureSession(): Promise<string | null> {\n    if (this.sessionId) return this.sessionId\n    if (this.creating) return this.creating\n    this.creating = (async () => {\n      try {\n        const res = await this.api.createDecodeSession({\n          codec: this.cfg.codec,\n          sourceSampleRate: this.cfg.sourceSampleRate,\n          sourceChannels: this.cfg.sourceChannels,\n          ...(this.cfg.extraData ? { extraData: this.cfg.extraData } : {}),\n          targetSampleRate: this.cfg.targetSampleRate,\n          targetChannels: this.cfg.targetChannels,\n          targetFormat: 'f32le',\n          ...(this.cfg.tag ? { tag: this.cfg.tag } : {}),\n        })\n        this.sessionId = res.sessionId\n        this.sessionNodeId = res.nodeId\n        this.startPollLoop()\n        this.logger?.info('audio-codec: decode session ready', {\n          meta: {\n            sessionId: res.sessionId,\n            nodeId: res.nodeId,\n            codec: this.cfg.codec,\n            target: `${this.cfg.targetSampleRate}Hz×${this.cfg.targetChannels}`,\n          },\n        })\n        return res.sessionId\n      } catch (err) {\n        this.logger?.error('audio-codec: createDecodeSession failed', {\n          meta: { codec: this.cfg.codec, error: errMsg(err) },\n        })\n        return null\n      } finally {\n        this.creating = null\n      }\n    })()\n    return this.creating\n  }\n\n  private startPollLoop(): void {\n    if (this.pollTimer) return\n    this.pollTimer = setInterval(() => {\n      void this.pollOnce()\n    }, POLL_INTERVAL_MS)\n    if (typeof this.pollTimer.unref === 'function') this.pollTimer.unref()\n  }\n\n  private async pollOnce(): Promise<void> {\n    const sid = this.sessionId\n    if (!sid || this.closed) return\n    const nodeId = this.sessionNodeId ?? undefined\n    try {\n      const pcms = await this.api.pullPcm({\n        sessionId: sid,\n        ...(nodeId ? { nodeId } : {}),\n        maxCount: MAX_PCM_PER_POLL,\n      })\n      for (const pcm of pcms) {\n        this.appendPcm(pcm)\n      }\n    } catch (err) {\n      // Stop polling and forget the dead session id when the upstream\n      // cap reaped it; the next pushed RTP packet will reopen via\n      // ensureSession(). Without this the timer fires every 40ms forever\n      // and the broker logs ~25 warn/s per dead session per camera.\n      if (isSessionGone(err)) {\n        this.handleSessionGone(sid)\n        return\n      }\n      this.warnThrottled('audio-codec: pullPcm failed', errMsg(err))\n    }\n  }\n\n  // ── Sample accumulator ──────────────────────────────────────────────────\n\n  /**\n   * Emit a freshly-pulled PCM frame as a DecodedAudioChunk immediately.\n   * No accumulator: realtime WebRTC consumers get audio at the codec's\n   * native cadence (~21-128ms per frame). ML analyzers that need a\n   * fixed-window input (YAMNet, Apple SA) aggregate at their own\n   * pipeline side from the pull-based AudioChunkPlane subscription.\n   *\n   * `pcm.pts` is a codec-relative PTS (often starts at 0), not a\n   * wall-clock; downstream consumers interpret the chunk timestamp as\n   * ms-since-epoch, so we stamp with `Date.now()` to match the\n   * `AudioRtpDecoder` path.\n   */\n  private appendPcm(pcm: { data: Uint8Array; sampleRate: number; channels: number; pts: number }): void {\n    if (pcm.data.byteLength === 0) return\n    // Buffer.from(uint8array) deep-copies — safe to keep across the\n    // RPC tick that owns the source buffer.\n    this.emit({\n      data: Buffer.from(pcm.data),\n      sampleRate: pcm.sampleRate,\n      channels: pcm.channels,\n      timestamp: Date.now(),\n    })\n  }\n\n  // ── Recovery + warn coalescing ──────────────────────────────────────────\n\n  /**\n   * Invoked from any push/pull path when the cap returns\n   * `decode session not found`. Drops our cached session id + tears\n   * down the poll loop so subsequent traffic re-creates the session\n   * lazily. Safe to call multiple times: second caller sees a null\n   * sessionId and noops.\n   */\n  private handleSessionGone(staleSid: string): void {\n    if (this.sessionId !== staleSid) return\n    this.sessionId = null\n    this.sessionNodeId = null\n    if (this.pollTimer) {\n      clearInterval(this.pollTimer)\n      this.pollTimer = null\n    }\n    // No accumulator to drain — each PCM frame from the reaped\n    // session was already emitted at pull time.\n    this.warnThrottled(\n      'audio-codec: session reaped upstream — will re-create on next packet',\n      `staleSessionId=${staleSid}`,\n    )\n  }\n\n  private handlePushError(label: string, sid: string, err: unknown): void {\n    if (isSessionGone(err)) {\n      this.handleSessionGone(sid)\n      return\n    }\n    this.warnThrottled(label, errMsg(err))\n  }\n\n  /**\n   * Logs at most once per WARN_THROTTLE_MS per (label,detail) pair and\n   * folds suppressed counts into the next log line so we keep visibility\n   * into chronic failures without flooding winston (whose JSON encode +\n   * sync write blocks the broker's event loop at >100/s).\n   */\n  private warnThrottled(label: string, detail: string): void {\n    const key = `${label}|${detail}`\n    const now = Date.now()\n    const prev = this.warnState.get(key)\n    if (prev && now - prev.lastLoggedAt < WARN_THROTTLE_MS) {\n      prev.suppressed++\n      return\n    }\n    const suppressed = prev?.suppressed ?? 0\n    this.warnState.set(key, { lastLoggedAt: now, suppressed: 0 })\n    this.logger?.warn(label, {\n      meta: {\n        codec: this.cfg.codec,\n        error: detail,\n        ...(suppressed > 0 ? { suppressedSinceLastLog: suppressed } : {}),\n      },\n    })\n  }\n}\n","import net from 'node:net'\nimport type { IScopedLogger } from '@camstack/types'\n\n/**\n * A lightweight TCP server that broadcasts raw encoded video bytes\n * (H.264/H.265 Annex-B) to all connected clients.\n *\n * Each StreamBroker owns one StreamPipeServer. Restreamers (e.g. go2rtc)\n * connect via `tcp://127.0.0.1:{port}` and receive the same byte stream\n * that the broker reads from the camera, eliminating duplicate RTSP\n * connections.\n */\n/** Max bytes queued in a client's write buffer before we consider it too slow */\nconst MAX_WRITE_BUFFER_BYTES = 2 * 1024 * 1024 // 2 MB\n\nexport class StreamPipeServer {\n  private readonly server: net.Server\n  private readonly clients: Set<net.Socket> = new Set()\n  private port = 0\n  private started = false\n  private readonly logger: IScopedLogger | undefined\n  private _onClientCountChanged: (() => void) | null = null\n\n  /** Register a callback invoked when pipe client count changes. */\n  onClientCountChanged(cb: () => void): void { this._onClientCountChanged = cb }\n\n  constructor(logger?: IScopedLogger) {\n    this.logger = logger\n    this.server = net.createServer((socket) => {\n      this.handleConnection(socket)\n    })\n\n    this.server.on('error', (err) => {\n      this.logger?.error('StreamPipeServer error', { meta: { error: err.message } })\n    })\n  }\n\n  async start(): Promise<void> {\n    if (this.started) {\n      return\n    }\n\n    await new Promise<void>((resolve, reject) => {\n      this.server.once('error', reject)\n      this.server.listen(0, '127.0.0.1', () => {\n        this.server.removeListener('error', reject)\n        const address = this.server.address() as net.AddressInfo\n        this.port = address.port\n        this.started = true\n        this.logger?.debug('StreamPipeServer listening', { meta: { port: this.port } })\n        resolve()\n      })\n    })\n  }\n\n  /**\n   * Broadcast raw encoded bytes to all connected clients.\n   * Silently drops data for clients that are slow or disconnected.\n   */\n  broadcast(data: Buffer): void {\n    for (const client of this.clients) {\n      if (client.destroyed) {\n        this.clients.delete(client)\n        continue\n      }\n\n      // Drop data for slow clients instead of letting buffers grow unbounded\n      if (client.writableLength > MAX_WRITE_BUFFER_BYTES) {\n        this.logger?.warn(\n          'StreamPipeServer client buffer overflow — dropping client',\n          { meta: { bufferMB: Number((client.writableLength / 1024 / 1024).toFixed(1)) } },\n        )\n        this.removeClient(client)\n        continue\n      }\n\n      client.write(data, (err) => {\n        if (err) {\n          this.logger?.debug(\n            'StreamPipeServer write error, removing client',\n            { meta: { error: err.message } },\n          )\n          this.removeClient(client)\n        }\n      })\n    }\n  }\n\n  getPort(): number {\n    return this.port\n  }\n\n  getUrl(): string {\n    return `tcp://127.0.0.1:${this.port}`\n  }\n\n  getClientCount(): number {\n    return this.clients.size\n  }\n\n  isStarted(): boolean {\n    return this.started\n  }\n\n  async stop(): Promise<void> {\n    if (!this.started) {\n      return\n    }\n\n    this.started = false\n    this._onClientCountChanged = null\n\n    for (const client of this.clients) {\n      client.destroy()\n    }\n    this.clients.clear()\n\n    await new Promise<void>((resolve) => {\n      this.server.close(() => {\n        this.logger?.debug('StreamPipeServer stopped')\n        resolve()\n      })\n    })\n  }\n\n  private handleConnection(socket: net.Socket): void {\n    this.clients.add(socket)\n    this.logger?.debug(\n      'StreamPipeServer client connected',\n      { meta: { total: this.clients.size } },\n    )\n    this._onClientCountChanged?.()\n\n    socket.on('close', () => {\n      this.removeClient(socket)\n    })\n\n    socket.on('error', () => {\n      this.removeClient(socket)\n    })\n  }\n\n  private removeClient(socket: net.Socket): void {\n    const existed = this.clients.delete(socket)\n    if (existed) {\n      this.logger?.debug(\n        'StreamPipeServer client disconnected',\n        { meta: { total: this.clients.size } },\n      )\n      this._onClientCountChanged?.()\n    }\n    if (!socket.destroyed) {\n      socket.destroy()\n    }\n  }\n}\n","import type { EncodedPacket } from '@camstack/types'\n\n/**\n * Time-based ring buffer for encoded video packets.\n *\n * Keeps the most recent N seconds of encoded packets, evicting old ones\n * based on timestamp. Used for pre-buffering: when an addon needs the last\n * N seconds of video (e.g., for clip creation), it calls `getPackets()`.\n *\n * The buffer always starts from the most recent keyframe to ensure\n * consumers can decode the returned data.\n */\nexport class EncodedRingBuffer {\n  private readonly packets: EncodedPacket[] = []\n  private maxDurationMs: number\n\n  constructor(maxDurationSec: number = 10) {\n    this.maxDurationMs = maxDurationSec * 1000\n  }\n\n  /** Update the buffer duration (in seconds). Evicts old packets immediately. */\n  setDuration(seconds: number): void {\n    this.maxDurationMs = seconds * 1000\n    this.evict()\n  }\n\n  /** Get the current buffer duration in seconds. */\n  getDuration(): number {\n    return this.maxDurationMs / 1000\n  }\n\n  /** Push a new packet into the buffer and evict expired ones. */\n  push(packet: EncodedPacket): void {\n    this.packets.push(packet)\n    this.evict()\n  }\n\n  /**\n   * Get buffered packets, starting from the most recent keyframe.\n   * Returns packets in chronological order (oldest first).\n   * Returns an empty array if no keyframe is found.\n   */\n  getPackets(): readonly EncodedPacket[] {\n    // Find the last keyframe index — consumers need a keyframe to start decoding\n    let lastKeyframeIdx = -1\n    for (let i = this.packets.length - 1; i >= 0; i--) {\n      if (this.packets[i]!.keyframe && this.packets[i]!.type === 'video') {\n        lastKeyframeIdx = i\n        break\n      }\n    }\n\n    if (lastKeyframeIdx < 0) return []\n\n    return this.packets.slice(lastKeyframeIdx)\n  }\n\n  /** Get the approximate duration of buffered content in seconds. */\n  getBufferedDurationMs(): number {\n    if (this.packets.length < 2) return 0\n    const first = this.packets[0]!\n    const last = this.packets[this.packets.length - 1]!\n    return last.pts - first.pts\n  }\n\n  /** Get the number of packets in the buffer. */\n  getPacketCount(): number {\n    return this.packets.length\n  }\n\n  /** Clear all buffered packets. */\n  clear(): void {\n    this.packets.length = 0\n  }\n\n  /** Remove packets older than maxDurationMs based on PTS. */\n  private evict(): void {\n    if (this.packets.length === 0) return\n\n    const cutoff = this.packets[this.packets.length - 1]!.pts - this.maxDurationMs\n    let removeCount = 0\n\n    while (removeCount < this.packets.length && this.packets[removeCount]!.pts < cutoff) {\n      removeCount++\n    }\n\n    if (removeCount > 0) {\n      this.packets.splice(0, removeCount)\n    }\n  }\n}\n","/** RTP header + payload ready to send over TCP interleaved. */\nexport interface RtpPacket {\n  /** RTP payload including 12-byte header. */\n  readonly data: Buffer\n  /** Whether this packet starts an access unit (for keyframe detection). */\n  readonly marker: boolean\n}\n\n/** Parsed RTSP request from a client. */\nexport interface RtspRequest {\n  readonly method: string\n  readonly uri: string\n  readonly cseq: number\n  readonly headers: ReadonlyMap<string, string>\n  readonly body?: string\n}\n\n/** Track setup state per client session. */\nexport interface TrackSetup {\n  readonly trackId: number\n  readonly rtpChannel: number\n  readonly rtcpChannel: number\n}\n\n/** Client session state. */\nexport interface ClientSession {\n  readonly sessionId: string\n  readonly tracks: ReadonlyArray<TrackSetup>\n  readonly playing: boolean\n}\n\n/** Codec parameters extracted from the stream. */\nexport interface CodecParams {\n  readonly codec: 'h264' | 'h265'\n  /** Parameter set NALs (SPS+PPS for H.264, VPS+SPS+PPS for H.265). */\n  readonly parameterSets: ReadonlyArray<Buffer>\n  readonly width?: number\n  readonly height?: number\n}\n\n/** RTSP method constants. */\nexport const RTSP_METHODS = ['OPTIONS', 'DESCRIBE', 'SETUP', 'PLAY', 'TEARDOWN', 'GET_PARAMETER'] as const\n\n/** RTP constants. */\nexport const RTP_VERSION = 2\nexport const RTP_HEADER_SIZE = 12\nexport const RTP_MAX_PAYLOAD = 1400 // Safe MTU for TCP interleaved\n\n/** H.264 NAL type masks. */\nexport const H264_NAL_TYPE_MASK = 0x1f\nexport const H264_NAL_FUA = 28\n\n/** H.265 NAL type extraction: (byte0 >> 1) & 0x3f. */\nexport const H265_NAL_TYPE_SHIFT = 1\nexport const H265_NAL_TYPE_MASK = 0x3f\nexport const H265_NAL_FU = 49\n\n/** TCP interleaved framing: magic byte. */\nexport const INTERLEAVED_MAGIC = 0x24\n\n/** Max write buffer before dropping slow client. */\nexport const MAX_WRITE_BUFFER = 2 * 1024 * 1024\n","import {\n  RTP_VERSION,\n  RTP_HEADER_SIZE,\n  RTP_MAX_PAYLOAD,\n  H264_NAL_TYPE_MASK,\n  H264_NAL_FUA,\n  H265_NAL_TYPE_SHIFT,\n  H265_NAL_TYPE_MASK,\n  H265_NAL_FU,\n  type RtpPacket,\n} from './rtsp-types.js'\n\nexport class RtpPacketizer {\n  private sequenceNumber = 0\n  private readonly ssrc: number\n\n  constructor(\n    private readonly codec: 'h264' | 'h265',\n    private readonly payloadType: number,\n  ) {\n    this.ssrc = Math.floor(Math.random() * 0xffffffff)\n  }\n\n  /**\n   * Split Annex-B byte stream into individual NAL units.\n   * Strips 3-byte (0x000001) and 4-byte (0x00000001) start codes.\n   */\n  static splitAnnexB(data: Buffer): ReadonlyArray<Buffer> {\n    const nals: Buffer[] = []\n    let start = -1\n\n    for (let i = 0; i < data.length - 2; i++) {\n      const isStartCode3 = data[i] === 0 && data[i + 1] === 0 && data[i + 2] === 1\n      const isStartCode4 = i + 3 < data.length &&\n        data[i] === 0 && data[i + 1] === 0 && data[i + 2] === 0 && data[i + 3] === 1\n\n      if (isStartCode3 || isStartCode4) {\n        if (start >= 0) {\n          // Strip trailing zero bytes — they belong to the next start code prefix.\n          // Emulation prevention (00 00 03) guarantees no false start codes in NAL payload.\n          let end = i\n          while (end > start && data[end - 1] === 0) end--\n          if (end > start) nals.push(data.subarray(start, end))\n        }\n        start = isStartCode4 ? i + 4 : i + 3\n        if (isStartCode4) i += 3\n        else i += 2\n      }\n    }\n\n    if (start >= 0 && start < data.length) {\n      nals.push(data.subarray(start))\n    }\n\n    return nals\n  }\n\n  /**\n   * Packetize a single NAL unit into one or more RTP packets.\n   * @param nal NAL unit WITHOUT start code\n   * @param timestamp RTP timestamp (90kHz clock)\n   * @param marker true if this is the last NAL of the access unit\n   */\n  packetize(nal: Buffer, timestamp: number, marker: boolean): ReadonlyArray<RtpPacket> {\n    if (this.codec === 'h264') {\n      return this.packetizeH264(nal, timestamp, marker)\n    }\n    return this.packetizeH265(nal, timestamp, marker)\n  }\n\n  private packetizeH264(nal: Buffer, timestamp: number, marker: boolean): ReadonlyArray<RtpPacket> {\n    if (nal.length <= RTP_MAX_PAYLOAD) {\n      return [this.buildRtpPacket(nal, timestamp, marker)]\n    }\n    return this.fragmentH264FUA(nal, timestamp, marker)\n  }\n\n  private packetizeH265(nal: Buffer, timestamp: number, marker: boolean): ReadonlyArray<RtpPacket> {\n    if (nal.length <= RTP_MAX_PAYLOAD) {\n      return [this.buildRtpPacket(nal, timestamp, marker)]\n    }\n    return this.fragmentH265FU(nal, timestamp, marker)\n  }\n\n  /** H.264 FU-A fragmentation (RFC 6184 Section 5.8). */\n  private fragmentH264FUA(nal: Buffer, timestamp: number, marker: boolean): ReadonlyArray<RtpPacket> {\n    const nalHeader = nal[0]!\n    const fnri = nalHeader & 0xe0\n    const nalType = nalHeader & H264_NAL_TYPE_MASK\n    const payload = nal.subarray(1)\n\n    const maxFragment = RTP_MAX_PAYLOAD - 2\n    const packets: RtpPacket[] = []\n    let offset = 0\n\n    while (offset < payload.length) {\n      const end = Math.min(offset + maxFragment, payload.length)\n      const isFirst = offset === 0\n      const isLast = end === payload.length\n\n      const fuIndicator = fnri | H264_NAL_FUA\n      const fuHeader = (isFirst ? 0x80 : 0) | (isLast ? 0x40 : 0) | nalType\n\n      const fragment = Buffer.allocUnsafe(2 + (end - offset))\n      fragment[0] = fuIndicator\n      fragment[1] = fuHeader\n      payload.copy(fragment, 2, offset, end)\n\n      packets.push(this.buildRtpPacket(fragment, timestamp, isLast && marker))\n      offset = end\n    }\n\n    return packets\n  }\n\n  /** H.265 FU fragmentation (RFC 7798 Section 4.4.3). */\n  private fragmentH265FU(nal: Buffer, timestamp: number, marker: boolean): ReadonlyArray<RtpPacket> {\n    const nalType = (nal[0]! >> H265_NAL_TYPE_SHIFT) & H265_NAL_TYPE_MASK\n    const tidLayerId = nal[1]!\n    const payload = nal.subarray(2)\n\n    const maxFragment = RTP_MAX_PAYLOAD - 3\n    const packets: RtpPacket[] = []\n    let offset = 0\n\n    while (offset < payload.length) {\n      const end = Math.min(offset + maxFragment, payload.length)\n      const isFirst = offset === 0\n      const isLast = end === payload.length\n\n      const payloadHdr0 = (H265_NAL_FU << H265_NAL_TYPE_SHIFT) | (nal[0]! & 0x81)\n      const payloadHdr1 = tidLayerId\n      const fuHeader = (isFirst ? 0x80 : 0) | (isLast ? 0x40 : 0) | nalType\n\n      const fragment = Buffer.allocUnsafe(3 + (end - offset))\n      fragment[0] = payloadHdr0\n      fragment[1] = payloadHdr1\n      fragment[2] = fuHeader\n      payload.copy(fragment, 3, offset, end)\n\n      packets.push(this.buildRtpPacket(fragment, timestamp, isLast && marker))\n      offset = end\n    }\n\n    return packets\n  }\n\n  private buildRtpPacket(payload: Buffer, timestamp: number, marker: boolean): RtpPacket {\n    const header = Buffer.allocUnsafe(RTP_HEADER_SIZE)\n    header[0] = RTP_VERSION << 6\n    header[1] = (marker ? 0x80 : 0) | (this.payloadType & 0x7f)\n    header.writeUInt16BE(this.sequenceNumber & 0xffff, 2)\n    this.sequenceNumber = (this.sequenceNumber + 1) & 0xffff\n    header.writeUInt32BE(timestamp >>> 0, 4)\n    header.writeUInt32BE(this.ssrc >>> 0, 8)\n\n    return {\n      data: Buffer.concat([header, payload]),\n      marker,\n    }\n  }\n}\n","/**\n * Annex-B NAL unit deframer.\n *\n * ffmpeg outputs Annex-B via stdout in arbitrary chunk sizes — a single\n * `data` event may contain partial NALs, multiple NALs, or NALs split\n * across events. This class accumulates bytes and emits complete NAL units.\n *\n * Usage:\n *   const deframer = new AnnexBDeframer((nal, isKeyframe) => { ... })\n *   proc.stdout.on('data', chunk => deframer.push(chunk))\n */\nexport class AnnexBDeframer {\n  private buffer: Buffer<ArrayBufferLike> = Buffer.alloc(0)\n  private readonly codec: 'h264' | 'h265'\n\n  constructor(\n    codec: 'h264' | 'h265',\n    private readonly onNal: (nal: Buffer, keyframe: boolean) => void,\n  ) {\n    this.codec = codec\n  }\n\n  /** Push raw Annex-B bytes. Complete NAL units are emitted via onNal callback. */\n  push(chunk: Buffer): void {\n    this.buffer = this.buffer.length > 0\n      ? Buffer.concat([this.buffer, chunk])\n      : chunk\n\n    this.drain()\n  }\n\n  /** Flush remaining buffer (call on stream end). */\n  flush(): void {\n    if (this.buffer.length > 0) {\n      const nal = this.stripLeadingStartCode(this.buffer)\n      if (nal.length > 0) {\n        this.onNal(nal, this.isKeyframeNal(nal))\n      }\n      this.buffer = Buffer.alloc(0)\n    }\n  }\n\n  /** Extract complete NAL units from the buffer, leaving partial data for next push. */\n  private drain(): void {\n    while (true) {\n      // Find first start code in buffer\n      const firstSc = this.findStartCode(0)\n      if (firstSc < 0) {\n        // No start code found at all — keep everything (partial NAL or leading junk)\n        return\n      }\n\n      // Find the NEXT start code after the first one\n      const nalStart = firstSc + (this.isStartCode4(firstSc) ? 4 : 3)\n      const nextSc = this.findStartCode(nalStart)\n\n      if (nextSc < 0) {\n        // Only one start code — NAL is not yet complete, wait for more data.\n        // Keep buffer from firstSc onwards (discard junk before first start code).\n        if (firstSc > 0) {\n          this.buffer = Buffer.from(this.buffer.subarray(firstSc))\n        }\n        return\n      }\n\n      // We have a complete NAL between firstSc and nextSc\n      const nal = this.buffer.subarray(nalStart, nextSc)\n      if (nal.length > 0) {\n        this.onNal(nal, this.isKeyframeNal(nal))\n      }\n\n      // Advance buffer past this NAL (keep from nextSc onwards)\n      this.buffer = Buffer.from(this.buffer.subarray(nextSc))\n    }\n  }\n\n  /** Find the byte offset of the next start code (00 00 01 or 00 00 00 01) at or after `from`. */\n  private findStartCode(from: number): number {\n    const buf = this.buffer\n    const len = buf.length - 2\n    for (let i = from; i < len; i++) {\n      // Fast skip: if buf[i+2] is not 0 or 1, no start code can begin at i\n      if (buf[i + 2]! > 1) {\n        i += 2\n        continue\n      }\n      if (buf[i] === 0 && buf[i + 1] === 0) {\n        if (buf[i + 2] === 1) return i\n        if (buf[i + 2] === 0 && i + 3 < buf.length && buf[i + 3] === 1) return i\n      }\n    }\n    return -1\n  }\n\n  /** Check if start code at `pos` is 4-byte (00 00 00 01) vs 3-byte (00 00 01). */\n  private isStartCode4(pos: number): boolean {\n    return pos + 3 < this.buffer.length &&\n      this.buffer[pos] === 0 &&\n      this.buffer[pos + 1] === 0 &&\n      this.buffer[pos + 2] === 0 &&\n      this.buffer[pos + 3] === 1\n  }\n\n  /** Strip leading start code from a buffer (if present). */\n  private stripLeadingStartCode(buf: Buffer): Buffer {\n    if (buf.length >= 4 && buf[0] === 0 && buf[1] === 0 && buf[2] === 0 && buf[3] === 1) {\n      return buf.subarray(4)\n    }\n    if (buf.length >= 3 && buf[0] === 0 && buf[1] === 0 && buf[2] === 1) {\n      return buf.subarray(3)\n    }\n    return buf\n  }\n\n  /** Check if a NAL unit (without start code) is a keyframe. */\n  private isKeyframeNal(nal: Buffer): boolean {\n    if (nal.length === 0) return false\n    if (this.codec === 'h264') {\n      const type = nal[0]! & 0x1f\n      return type === 5 // IDR\n    }\n    const type = (nal[0]! >> 1) & 0x3f\n    return type >= 16 && type <= 21 // IRAP\n  }\n}\n","export interface SdpOptions {\n  readonly codec: 'h264' | 'h265'\n  readonly parameterSets: ReadonlyArray<Buffer>\n  readonly streamName: string\n}\n\nexport function buildSdp(options: SdpOptions): string {\n  const { codec, parameterSets, streamName } = options\n  const lines: string[] = [\n    'v=0',\n    `o=- ${Date.now()} 1 IN IP4 0.0.0.0`,\n    `s=${streamName}`,\n    't=0 0',\n    'a=tool:camstack-rtsp-restream',\n    'm=video 0 RTP/AVP 96',\n    'c=IN IP4 0.0.0.0',\n    'b=AS:10000',\n  ]\n\n  if (codec === 'h264') {\n    lines.push('a=rtpmap:96 H264/90000')\n    const sps = parameterSets[0]\n    const spropSets = parameterSets.map(ps => ps.toString('base64')).join(',')\n    const profileLevelId = sps && sps.length >= 4\n      ? Buffer.from([sps[1]!, sps[2]!, sps[3]!]).toString('hex')\n      : '42001e'\n    lines.push(`a=fmtp:96 packetization-mode=1;profile-level-id=${profileLevelId};sprop-parameter-sets=${spropSets}`)\n  } else {\n    lines.push('a=rtpmap:96 H265/90000')\n    const fmtpParts = parameterSets.map((ps, i) => {\n      const prefix = i === 0 ? 'sprop-vps' : i === 1 ? 'sprop-sps' : 'sprop-pps'\n      return `${prefix}=${ps.toString('base64')}`\n    })\n    lines.push(`a=fmtp:96 ${fmtpParts.join(';')}`)\n  }\n\n  lines.push('a=control:trackID=0')\n  lines.push('')\n  return lines.join('\\r\\n')\n}\n","/**\n * Egress audio SDP grafting.\n *\n * The push-mode restreamer SDP (built by `buildSdp`) is video-only: one\n * `m=video` track on payload 96, `a=control:trackID=0`. To carry transcoded\n * audio we let ffmpeg's RTP muxer emit the authoritative audio SDP (`-sdp_file`,\n * correct fmtp/config even for AAC), extract its `m=audio` block, and graft it\n * onto the video SDP as a second track (`trackID=1`).\n *\n * Two rewrites keep the grafted block consistent with the server SDP:\n *   - the media port is forced to 0 (RTSP uses interleaved TCP, the port is\n *     irrelevant and ffmpeg writes its own UDP port);\n *   - a *dynamic* payload type (>= 96) is remapped to 97 so it never collides\n *     with the video track's 96. Static types (PCMU=0, PCMA=8) are left intact.\n *\n * Pure + side-effect-free so the grafting is unit-testable without ffmpeg.\n */\n\nconst VIDEO_PAYLOAD_TYPE = 96\nconst AUDIO_DYNAMIC_PAYLOAD_TYPE = 97\n\ninterface ExtractOptions {\n  /** trackID to assign to the grafted audio track. Defaults to 1. */\n  readonly trackId?: number\n}\n\n/**\n * Extract the `m=audio` media block from an ffmpeg-generated SDP, normalised\n * for grafting (port 0, non-colliding payload type, own `a=control`, explicit\n * connection line). Returns `null` when the SDP has no audio media section.\n */\nexport function extractAudioMediaBlock(ffmpegSdp: string, options: ExtractOptions = {}): string | null {\n  const trackId = options.trackId ?? 1\n  const lines = ffmpegSdp.split(/\\r?\\n/)\n  const startIdx = lines.findIndex((l) => l.startsWith('m=audio'))\n  if (startIdx < 0) return null\n\n  // Collect the audio media block: from `m=audio` to the next `m=` or EOF.\n  const block: string[] = []\n  for (let i = startIdx; i < lines.length; i++) {\n    const line = lines[i]!\n    if (i > startIdx && line.startsWith('m=')) break\n    if (line.trim().length === 0) continue\n    if (line.startsWith('a=control:')) continue // we add our own\n    block.push(line)\n  }\n\n  // Parse the original payload type from `m=audio <port> RTP/AVP <pt>`.\n  const mLine = block[0]!\n  const mParts = mLine.split(/\\s+/)\n  const origPt = Number.parseInt(mParts[3] ?? '', 10)\n  if (!Number.isInteger(origPt)) return null\n  const newPt = origPt >= 96 ? AUDIO_DYNAMIC_PAYLOAD_TYPE : origPt\n\n  const remapped = block.map((line) => {\n    if (line.startsWith('m=audio')) {\n      return `m=audio 0 ${mParts.slice(2).join(' ').replace(String(origPt), String(newPt))}`\n    }\n    if (origPt !== newPt && (line.startsWith('a=rtpmap:') || line.startsWith('a=fmtp:'))) {\n      return line.replace(`:${origPt} `, `:${newPt} `)\n    }\n    return line\n  })\n\n  // Ensure a connection line so the block is self-contained (ffmpeg puts c= at\n  // session level). Insert right after the m= line if absent.\n  if (!remapped.some((l) => l.startsWith('c='))) {\n    remapped.splice(1, 0, 'c=IN IP4 0.0.0.0')\n  }\n\n  remapped.push(`a=control:trackID=${trackId}`)\n  return remapped.join('\\r\\n')\n}\n\n/**\n * Append a grafted audio media block to a video-only SDP, preserving CRLF\n * framing and the trailing blank line.\n */\nexport function appendAudioMediaBlock(videoSdp: string, audioBlock: string): string {\n  const trimmed = videoSdp.replace(/[\\r\\n]+$/, '')\n  return `${trimmed}\\r\\n${audioBlock}\\r\\n`\n}\n\nexport { VIDEO_PAYLOAD_TYPE, AUDIO_DYNAMIC_PAYLOAD_TYPE }\n","import type { EncodedPacket, IScopedLogger } from '@camstack/types'\nimport { RtpPacketizer } from './rtp-packetizer.js'\nimport { AnnexBDeframer } from './annexb-deframer.js'\nimport { buildSdp } from './sdp-builder.js'\nimport { appendAudioMediaBlock } from './egress-audio-sdp.js'\nimport { RtspSession } from './rtsp-session.js'\nimport type { CodecParams, RtpPacket } from './rtsp-types.js'\n\n/**\n * Per-broker RTSP restreamer.\n *\n * Operates in two modes:\n *\n * **Annex-B mode** (legacy, ffmpeg-based):\n *   Receives raw Annex-B chunks via pushPacket(), deframes into NALs,\n *   RTP-packetizes, and broadcasts to RTSP sessions.\n *\n * **RTP passthrough mode** (native RTSP client):\n *   Receives the camera's SDP via setCameraSdp() and raw RTP packets\n *   via pushRtpPassthrough(). Forwards RTP directly to clients — zero\n *   re-packetization, preserving the camera's original framing.\n */\nexport class RtspRestreamer {\n  private readonly sessions = new Map<string, RtspSession>()\n  /** Sessions that need a keyframe before receiving regular frames. */\n  private readonly pendingKeyframe = new Set<string>()\n  private packetizer: RtpPacketizer | null = null\n  private codecParams: CodecParams | null = null\n  private sdp: string | null = null\n  private mutedSdp: string | null = null\n  private deframer: AnnexBDeframer | null = null\n  private detectedCodec: 'h264' | 'h265' | null = null\n  /**\n   * Grafted audio `m=audio` block for transcode-egress audio (set by the\n   * egress from ffmpeg's `-sdp_file`). Appended to the push-mode video SDP as\n   * trackID=1. Muted sessions still get the video-only SDP.\n   */\n  private egressAudioBlock: string | null = null\n\n  /** Cached last keyframe NALs (SPS/PPS + IDR) for late-joining clients. */\n  private lastKeyframeNals: Buffer[] = []\n  private lastKeyframeTimestamp = 0\n  /** True if we're currently accumulating a keyframe's NALs. */\n  private collectingKeyframe = false\n\n  /** Current RTP timestamp for the batch of NALs being processed. */\n  private currentTimestamp = 0\n\n  /** RTP passthrough: cached last keyframe RTP packets for late-joining clients. */\n  private lastKeyframeRtpPackets: RtpPacket[] = []\n  /** True when in RTP passthrough mode (setCameraSdp was called). */\n  private _rtpPassthroughMode = false\n  /** Accumulating keyframe RTP packets in passthrough mode. */\n  private collectingKeyframeRtp = false\n\n  private logger: IScopedLogger | null = null\n  private _onSessionCountChanged: (() => void) | null = null\n\n  constructor(private readonly streamName: string) {}\n\n  /** True when using RTP passthrough (native client) instead of Annex-B re-packetization. */\n  get rtpPassthroughMode(): boolean { return this._rtpPassthroughMode }\n\n  /** Register a callback invoked when RTSP client count changes. */\n  onSessionCountChanged(cb: () => void): void { this._onSessionCountChanged = cb }\n\n  setLogger(logger: IScopedLogger): void {\n    this.logger = logger\n  }\n\n  /**\n   * Reset all stream-session state. Called on broker suspend so stale\n   * keyframe caches / codec params from the previous session don't leak\n   * into the next one after resume.\n   */\n  resetStreamState(): void {\n    this.lastKeyframeNals = []\n    this.lastKeyframeTimestamp = 0\n    this.collectingKeyframe = false\n    this.currentTimestamp = 0\n    this.packetizer = null\n    this.codecParams = null\n    this.sdp = null\n    this.mutedSdp = null\n    this.deframer = null\n    this.detectedCodec = null\n    this.lastKeyframeRtpPackets = []\n    this._rtpPassthroughMode = false\n    this.collectingKeyframeRtp = false\n    this.egressAudioBlock = null\n  }\n\n  /**\n   * Graft (or clear) a transcode-egress audio track onto the push-mode SDP.\n   * Safe to call before or after the video codec params are detected — the SDP\n   * is recomposed whenever both are available.\n   */\n  setEgressAudioBlock(block: string | null): void {\n    this.egressAudioBlock = block\n    if (this.codecParams) this.rebuildSdp()\n  }\n\n  /** Recompose the push-mode SDP from current codec params + egress audio block. */\n  private rebuildSdp(): void {\n    if (!this.codecParams) return\n    const { codec, parameterSets } = this.codecParams\n    const videoSdp = buildSdp({ codec, parameterSets, streamName: this.streamName })\n    this.sdp = this.egressAudioBlock ? appendAudioMediaBlock(videoSdp, this.egressAudioBlock) : videoSdp\n    // Muted sessions remain video-only.\n    this.mutedSdp = buildSdp({ codec, parameterSets, streamName: `${this.streamName} (muted)` })\n  }\n\n  /** True when codec params are detected and SDP is available — safe to accept PLAY. */\n  isReady(): boolean { return this.codecParams !== null && this.packetizer !== null }\n\n  getCodecParams(): CodecParams | null { return this.codecParams }\n  getSdp(): string | null { return this.sdp }\n  getMutedSdp(): string | null { return this.mutedSdp ?? this.sdp }\n  getSessionCount(): number { return this.sessions.size }\n\n  /** Per-session diagnostic snapshots consumed by `StreamBroker.listClients`. */\n  listSessionInfos(): readonly ReturnType<RtspSession['getInfo']>[] {\n    return [...this.sessions.values()].map((s) => s.getInfo())\n  }\n\n  addSession(session: RtspSession): void {\n    this.sessions.set(session.getSessionId(), session)\n    this.pendingKeyframe.add(session.getSessionId())\n    this.logger?.info('RTSP client connected', {\n      meta: { streamName: this.streamName, total: this.sessions.size },\n    })\n    this._onSessionCountChanged?.()\n\n    if (this._rtpPassthroughMode) {\n      // RTP passthrough: serve cached keyframe RTP packets to late-joining clients\n      if (this.lastKeyframeRtpPackets.length > 0) {\n        setTimeout(() => this.servePendingSessionsRtp(), 0)\n      }\n    } else {\n      // Annex-B mode: serve cached keyframe NALs via packetizer\n      if (this.lastKeyframeNals.length > 0 && this.packetizer) {\n        setTimeout(() => this.servePendingSessions(), 0)\n      }\n    }\n  }\n\n  // ── RTP Passthrough Mode ──────────��───────────────────────────────\n\n  /**\n   * Set the camera's SDP directly (from DESCRIBE response).\n   * Switches the restreamer to RTP passthrough mode — subsequent calls\n   * should use pushRtpPassthrough() instead of pushPacket().\n   */\n  setCameraSdp(sdpText: string, codec: 'h264' | 'h265', codecParams: CodecParams | null): void {\n    this._rtpPassthroughMode = true\n    this.detectedCodec = codec\n    this.codecParams = codecParams\n    this.sdp = sdpText\n    // Build a video-only SDP for muted sessions (strip all m=audio sections)\n    this.mutedSdp = stripAudioFromSdp(sdpText)\n    this.logger?.info('RTP passthrough mode — SDP set', {\n      meta: { streamName: this.streamName, codec },\n    })\n  }\n\n  /**\n   * Push a raw RTP packet from the native RTSP client.\n   * Forwarded directly to all connected RTSP sessions — zero re-packetization.\n   *\n   * @param rtpData Complete RTP packet (header + payload)\n   * @param isKeyframe True if this packet is part of a keyframe access unit\n   * @param isParameterSet True if this packet contains SPS/PPS/VPS\n   */\n  pushRtpPassthrough(rtpData: Buffer, isKeyframe: boolean, isParameterSet: boolean): void {\n    if (this.sessions.size === 0 && !isKeyframe && !isParameterSet) return\n\n    const rtpPacket: RtpPacket = { data: rtpData, marker: (rtpData[1]! & 0x80) !== 0 }\n\n    // Cache keyframe RTP packets for late-joining clients\n    if (isKeyframe || isParameterSet) {\n      if (!this.collectingKeyframeRtp) {\n        this.collectingKeyframeRtp = true\n        this.lastKeyframeRtpPackets = []\n      }\n    }\n\n    if (this.collectingKeyframeRtp) {\n      this.lastKeyframeRtpPackets.push({ data: Buffer.from(rtpData), marker: rtpPacket.marker })\n      if (!isKeyframe && !isParameterSet) {\n        this.collectingKeyframeRtp = false\n      }\n    }\n\n    if (this.sessions.size === 0) return\n\n    // Serve pending sessions (late joiners waiting for keyframe)\n    this.servePendingSessionsRtp()\n\n    // Broadcast to all active sessions\n    for (const session of this.sessions.values()) {\n      if (this.pendingKeyframe.has(session.getSessionId())) continue\n      if (session.isPlaying()) {\n        session.sendRtp(rtpPacket)\n      }\n    }\n  }\n\n  /** Send cached keyframe RTP packets to sessions that transitioned to PLAY. */\n  private servePendingSessionsRtp(): void {\n    if (this.pendingKeyframe.size === 0) return\n    if (this.lastKeyframeRtpPackets.length === 0) return\n\n    const ready: string[] = []\n    for (const sessionId of this.pendingKeyframe) {\n      const session = this.sessions.get(sessionId)\n      if (session?.isPlaying()) ready.push(sessionId)\n    }\n\n    if (ready.length === 0) return\n\n    for (const rtpPacket of this.lastKeyframeRtpPackets) {\n      for (const sessionId of ready) {\n        const session = this.sessions.get(sessionId)\n        if (session?.isPlaying()) session.sendRtp(rtpPacket)\n      }\n    }\n\n    for (const id of ready) this.pendingKeyframe.delete(id)\n  }\n\n  /**\n   * Push a raw audio RTP packet from the native RTSP client.\n   * Forwarded to all non-muted sessions. Muted sessions receive no audio.\n   */\n  pushAudioRtpPassthrough(rtpData: Buffer): void {\n    if (this.sessions.size === 0) return\n\n    const rtpPacket: RtpPacket = { data: rtpData, marker: (rtpData[1]! & 0x80) !== 0 }\n\n    for (const session of this.sessions.values()) {\n      if (session.isMuted()) continue\n      if (this.pendingKeyframe.has(session.getSessionId())) continue\n      if (session.isPlaying()) {\n        session.sendAudioRtp(rtpPacket)\n      }\n    }\n  }\n\n  removeSession(sessionId: string): void {\n    this.sessions.delete(sessionId)\n    this.pendingKeyframe.delete(sessionId)\n    this.logger?.info('RTSP client disconnected', {\n      meta: { streamName: this.streamName, total: this.sessions.size },\n    })\n    this._onSessionCountChanged?.()\n  }\n\n  /**\n   * Force-close + remove a session by id. Used by the admin `killClient`\n   * cap to let operators kick a stuck holder. Returns `true` if the\n   * session existed and was dropped, `false` if the id doesn't match.\n   */\n  killSession(sessionId: string): boolean {\n    const session = this.sessions.get(sessionId)\n    if (!session) return false\n    try { session.close() } catch { /* best-effort close */ }\n    this.removeSession(sessionId)\n    return true\n  }\n\n  pushPacket(packet: EncodedPacket): void {\n    if (packet.type !== 'video') return\n\n    this.currentTimestamp = Math.floor(packet.pts * 90) // ms → 90kHz\n\n    // Initialize deframer on first packet (need codec to detect keyframes)\n    if (!this.deframer) {\n      this.detectedCodec = (packet.codec === 'h265' || packet.codec === 'hevc') ? 'h265' : 'h264'\n      this.deframer = new AnnexBDeframer(this.detectedCodec, (nal, isKeyframe) => {\n        this.handleNal(nal, isKeyframe)\n      })\n    }\n\n    // NOTE: keyframe collection is triggered inside handleNal() when the\n    // deframer emits a complete keyframe NAL — NOT from packet.keyframe,\n    // which is unreliable because ffmpeg chunks don't align to NAL boundaries.\n\n    // Push raw bytes — the deframer emits complete NALs via handleNal callback\n    this.deframer.push(packet.data)\n  }\n\n  destroy(): void {\n    this._onSessionCountChanged = null\n    this.deframer?.flush()\n    for (const session of this.sessions.values()) {\n      session.close()\n    }\n    this.sessions.clear()\n    this.packetizer = null\n    this.deframer = null\n  }\n\n  /** Called by deframer for each complete NAL unit. */\n  private handleNal(nal: Buffer, isKeyframe: boolean): void {\n    // Initialize codec params from parameter set NALs (SPS/PPS/VPS)\n    if (!this.codecParams) {\n      this.tryInitCodec(nal)\n    }\n\n    // Start keyframe collection when the deframer emits a keyframe or\n    // parameter set NAL. This is reliable because the deframer works on\n    // complete NAL units — unlike packet.keyframe which scans raw chunks\n    // that may split a keyframe across two ffmpeg data events.\n    if (isKeyframe || this.isParameterSetNal(nal)) {\n      if (!this.collectingKeyframe) {\n        this.collectingKeyframe = true\n        this.lastKeyframeNals = []\n        this.lastKeyframeTimestamp = this.currentTimestamp\n      }\n    }\n\n    // Cache keyframe NALs\n    if (this.collectingKeyframe) {\n      this.lastKeyframeNals.push(Buffer.from(nal))\n      // Stop collecting after the first non-parameter-set, non-keyframe NAL\n      // (parameter sets + IDR = the keyframe, then P/B frames start)\n      if (!isKeyframe && !this.isParameterSetNal(nal)) {\n        this.collectingKeyframe = false\n      }\n    }\n\n    // Skip RTP if no clients connected or no packetizer yet\n    if (this.sessions.size === 0) return\n    if (!this.packetizer) return\n\n    // Serve pending sessions: send cached keyframe when they transition to PLAY\n    this.servePendingSessions()\n\n    // RTP packetize and broadcast\n    const rtpPackets = this.packetizer.packetize(nal, this.currentTimestamp, true)\n    for (const rtp of rtpPackets) {\n      for (const session of this.sessions.values()) {\n        if (this.pendingKeyframe.has(session.getSessionId())) continue\n        if (session.isPlaying()) {\n          session.sendRtp(rtp)\n        }\n      }\n    }\n  }\n\n  /** Send cached keyframe to sessions that have transitioned to PLAY. */\n  private servePendingSessions(): void {\n    if (this.pendingKeyframe.size === 0) return\n\n    const ready: string[] = []\n    for (const sessionId of this.pendingKeyframe) {\n      const session = this.sessions.get(sessionId)\n      if (session?.isPlaying()) ready.push(sessionId)\n    }\n\n    if (ready.length === 0) return\n    if (this.lastKeyframeNals.length === 0 || !this.packetizer) return\n\n    // Send cached keyframe NALs to each ready session\n    for (const keyNal of this.lastKeyframeNals) {\n      const rtpPackets = this.packetizer.packetize(keyNal, this.lastKeyframeTimestamp, true)\n      for (const rtp of rtpPackets) {\n        for (const sessionId of ready) {\n          const session = this.sessions.get(sessionId)\n          if (session?.isPlaying()) session.sendRtp(rtp)\n        }\n      }\n    }\n\n    for (const id of ready) this.pendingKeyframe.delete(id)\n  }\n\n  /** Try to initialize codec params from a parameter set NAL (SPS/PPS for H264, VPS/SPS/PPS for H265). */\n  private tryInitCodec(nal: Buffer): void {\n    if (!this.isParameterSetNal(nal)) return\n    if (!this.detectedCodec) return\n\n    // Accumulate parameter sets until we have enough\n    if (!this.codecParams) {\n      // Start a new collection\n      const existing = this.lastKeyframeNals.filter(n => this.isParameterSetNal(n))\n      const all = [...existing, nal]\n\n      const codec = this.detectedCodec\n      const needed = codec === 'h264' ? 2 : 3 // SPS+PPS or VPS+SPS+PPS\n      if (all.length >= needed) {\n        this.codecParams = { codec, parameterSets: all.slice(0, needed) }\n        this.packetizer = new RtpPacketizer(codec, 96)\n        this.rebuildSdp()\n      }\n    }\n  }\n\n  /** Check if a NAL is a parameter set (SPS, PPS, or VPS). */\n  private isParameterSetNal(nal: Buffer): boolean {\n    if (nal.length === 0) return false\n    if (this.detectedCodec === 'h264') {\n      const type = nal[0]! & 0x1f\n      return type === 7 || type === 8 // SPS, PPS\n    }\n    const type = (nal[0]! >> 1) & 0x3f\n    return type === 32 || type === 33 || type === 34 // VPS, SPS, PPS\n  }\n}\n\n/**\n * Strip audio media sections from an SDP string.\n * Returns a video-only SDP for muted RTSP sessions.\n *\n * SDP structure: session-level lines, then m= blocks.\n * We keep session-level lines and all m= blocks that aren't audio.\n */\nfunction stripAudioFromSdp(sdpText: string): string {\n  const lines = sdpText.split(/\\r?\\n/)\n  const result: string[] = []\n  let inAudioSection = false\n\n  for (const line of lines) {\n    if (line.startsWith('m=')) {\n      inAudioSection = line.startsWith('m=audio')\n    }\n    if (!inAudioSection) {\n      result.push(line)\n    }\n  }\n\n  return result.join('\\r\\n')\n}\n","import type { CodecParams } from './rtsp-types.js'\n\n/** Parsed representation of an SDP media section (m= block). */\nexport interface SdpMediaSection {\n  /** Media type: \"video\", \"audio\", \"application\". */\n  readonly type: string\n  /** RTP port (0 for TCP interleaved). */\n  readonly port: number\n  /** Transport protocol (e.g. \"RTP/AVP\"). */\n  readonly protocol: string\n  /** Payload types listed on the m= line. */\n  readonly payloadTypes: ReadonlyArray<number>\n  /** Codec name from a=rtpmap (e.g. \"H264\", \"H265\", \"PCMU\"). */\n  readonly codec: string | null\n  /** Clock rate from a=rtpmap (e.g. 90000 for video). */\n  readonly clockRate: number | null\n  /** Audio channel count from a=rtpmap (null for video). */\n  readonly channels: number | null\n  /** Parsed a=fmtp parameters as key-value pairs. */\n  readonly fmtp: Readonly<Record<string, string>>\n  /** Track control URL from a=control (may be relative or absolute). */\n  readonly control: string | null\n  /** Raw SDP lines for this section. */\n  readonly lines: ReadonlyArray<string>\n}\n\n/** Result of parsing a full SDP response. */\nexport interface ParsedSdp {\n  /** Session-level lines (before the first m= line). */\n  readonly sessionLines: ReadonlyArray<string>\n  /** Session-level a=control (base URL for relative track controls). */\n  readonly sessionControl: string | null\n  /** All parsed media sections. */\n  readonly mediaSections: ReadonlyArray<SdpMediaSection>\n}\n\n/** Parsed video track info ready for native RTSP client use. */\nexport interface SdpVideoTrack {\n  /** Index of this media section in the SDP. */\n  readonly sectionIndex: number\n  /** Codec: \"h264\" or \"h265\". */\n  readonly codec: 'h264' | 'h265'\n  /** RTP payload type (usually 96+). */\n  readonly payloadType: number\n  /** Clock rate (90000 for video). */\n  readonly clockRate: number\n  /** Track control URL from a=control. */\n  readonly control: string | null\n  /** H.264 profile-level-id (hex string, e.g. \"42001e\"). Null for H.265. */\n  readonly profileLevelId: string | null\n  /** Codec parameters extracted from SDP fmtp (SPS/PPS for H264, VPS/SPS/PPS for H265). */\n  readonly codecParams: CodecParams | null\n}\n\n/** Parsed audio track info. */\nexport interface SdpAudioTrack {\n  /** Index of this media section in the SDP. */\n  readonly sectionIndex: number\n  /** Codec name (e.g. \"PCMU\", \"PCMA\", \"MPEG4-GENERIC\", \"opus\"). */\n  readonly codec: string\n  /** RTP payload type. */\n  readonly payloadType: number\n  /** Clock rate (e.g. 8000, 48000). */\n  readonly clockRate: number\n  /** Channel count. */\n  readonly channels: number\n  /** Track control URL from a=control. */\n  readonly control: string | null\n  /** Raw fmtp parameters. */\n  readonly fmtp: Readonly<Record<string, string>>\n}\n\n/**\n * Parse a raw SDP text into structured sections.\n *\n * Handles:\n * - Session-level and media-level attributes\n * - Multiple media sections (video, audio, application)\n * - a=rtpmap, a=fmtp, a=control extraction\n * - Tolerant of \\r\\n and \\n line endings\n */\nexport function parseSdp(sdpText: string): ParsedSdp {\n  const lines = sdpText.split(/\\r?\\n/).filter(l => l.length > 0)\n\n  const sessionLines: string[] = []\n  const sections: Array<{ mLine: string; lines: string[] }> = []\n  let currentSection: { mLine: string; lines: string[] } | null = null\n\n  for (const line of lines) {\n    if (line.startsWith('m=')) {\n      currentSection = { mLine: line, lines: [line] }\n      sections.push(currentSection)\n    } else if (currentSection) {\n      currentSection.lines.push(line)\n    } else {\n      sessionLines.push(line)\n    }\n  }\n\n  const sessionControl = extractAttribute(sessionLines, 'control')\n\n  const mediaSections = sections.map(s => parseMediaSection(s.mLine, s.lines))\n\n  return { sessionLines, sessionControl, mediaSections }\n}\n\n/**\n * Extract the first video track from a parsed SDP.\n * Returns null if no supported video codec found.\n */\nexport function extractVideoTrack(parsed: ParsedSdp): SdpVideoTrack | null {\n  for (let i = 0; i < parsed.mediaSections.length; i++) {\n    const section = parsed.mediaSections[i]!\n    if (section.type !== 'video') continue\n\n    const codecUpper = section.codec?.toUpperCase() ?? null\n    if (codecUpper !== 'H264' && codecUpper !== 'H265') continue\n\n    const codec: 'h264' | 'h265' = codecUpper === 'H264' ? 'h264' : 'h265'\n    const payloadType = section.payloadTypes[0] ?? 96\n    const clockRate = section.clockRate ?? 90000\n\n    const codecParams = extractCodecParams(codec, section.fmtp)\n    const profileLevelId = codec === 'h264'\n      ? (section.fmtp['profile-level-id'] ?? null)\n      : null\n\n    return {\n      sectionIndex: i,\n      codec,\n      payloadType,\n      clockRate,\n      control: section.control,\n      profileLevelId,\n      codecParams,\n    }\n  }\n  return null\n}\n\n/**\n * Extract the first audio track from a parsed SDP.\n * Returns null if no audio section found.\n */\nexport function extractAudioTrack(parsed: ParsedSdp): SdpAudioTrack | null {\n  for (let i = 0; i < parsed.mediaSections.length; i++) {\n    const section = parsed.mediaSections[i]!\n    if (section.type !== 'audio') continue\n\n    return {\n      sectionIndex: i,\n      codec: section.codec ?? 'unknown',\n      payloadType: section.payloadTypes[0] ?? 0,\n      clockRate: section.clockRate ?? 8000,\n      channels: section.channels ?? 1,\n      control: section.control,\n      fmtp: section.fmtp,\n    }\n  }\n  return null\n}\n\n/**\n * Resolve a track control URL against the session-level control / RTSP base URL.\n *\n * Cameras vary wildly:\n * - Absolute: \"rtsp://192.168.1.100/trackID=1\"\n * - Relative: \"trackID=1\" or \"track1\"\n * - Session control: \"rtsp://192.168.1.100/stream1\" with track \"trackID=1\"\n */\nexport function resolveTrackUrl(\n  baseUrl: string,\n  sessionControl: string | null,\n  trackControl: string | null,\n): string {\n  if (!trackControl) return baseUrl\n\n  // Absolute URL — use as-is\n  if (trackControl.startsWith('rtsp://') || trackControl.startsWith('rtsps://')) {\n    return trackControl\n  }\n\n  // Use session control as base if it's an absolute URL\n  const base = (sessionControl && (sessionControl.startsWith('rtsp://') || sessionControl.startsWith('rtsps://')))\n    ? sessionControl\n    : baseUrl\n\n  // Append track control to base\n  const separator = base.endsWith('/') ? '' : '/'\n  return `${base}${separator}${trackControl}`\n}\n\n// ── Internal helpers ────────────────────────────────────────────────\n\nfunction parseMediaSection(mLine: string, lines: ReadonlyArray<string>): SdpMediaSection {\n  // m=<type> <port> <protocol> <payloadType1> [payloadType2 ...]\n  const mParts = mLine.substring(2).split(/\\s+/)\n  const type = mParts[0] ?? 'unknown'\n  const port = parseInt(mParts[1] ?? '0', 10)\n  const protocol = mParts[2] ?? 'RTP/AVP'\n  const payloadTypes = mParts.slice(3).map(p => parseInt(p, 10)).filter(n => !isNaN(n))\n\n  const primaryPt = payloadTypes[0]\n\n  // Parse a=rtpmap:<pt> <codec>/<clockRate>[/<channels>]\n  let codec: string | null = null\n  let clockRate: number | null = null\n  let channels: number | null = null\n\n  for (const line of lines) {\n    const rtpmapMatch = line.match(/^a=rtpmap:(\\d+)\\s+([^/]+)\\/(\\d+)(?:\\/(\\d+))?/)\n    if (rtpmapMatch) {\n      const pt = parseInt(rtpmapMatch[1]!, 10)\n      if (pt === primaryPt) {\n        codec = rtpmapMatch[2]!\n        clockRate = parseInt(rtpmapMatch[3]!, 10)\n        channels = rtpmapMatch[4] ? parseInt(rtpmapMatch[4], 10) : null\n      }\n    }\n  }\n\n  // Fallback for well-known static payload types (no rtpmap line)\n  if (!codec && primaryPt !== undefined) {\n    const staticCodec = STATIC_PAYLOAD_TYPES.get(primaryPt)\n    if (staticCodec) {\n      codec = staticCodec.codec\n      clockRate = staticCodec.clockRate\n      channels = staticCodec.channels\n    }\n  }\n\n  // Parse a=fmtp:<pt> <key>=<value>;<key>=<value>;...\n  const fmtp = parseFmtp(lines, primaryPt)\n\n  // Parse a=control:<url>\n  const control = extractAttribute(lines, 'control')\n\n  return { type, port, protocol, payloadTypes, codec, clockRate, channels, fmtp, control, lines }\n}\n\nfunction parseFmtp(\n  lines: ReadonlyArray<string>,\n  payloadType: number | undefined,\n): Record<string, string> {\n  const result: Record<string, string> = {}\n\n  for (const line of lines) {\n    const fmtpMatch = line.match(/^a=fmtp:(\\d+)\\s+(.+)/)\n    if (!fmtpMatch) continue\n    const pt = parseInt(fmtpMatch[1]!, 10)\n    if (payloadType !== undefined && pt !== payloadType) continue\n\n    const paramsStr = fmtpMatch[2]!\n    // Split on ';' then on first '='\n    for (const part of paramsStr.split(/;\\s*/)) {\n      const eqIdx = part.indexOf('=')\n      if (eqIdx > 0) {\n        const key = part.substring(0, eqIdx).trim().toLowerCase()\n        const value = part.substring(eqIdx + 1).trim()\n        result[key] = value\n      }\n    }\n  }\n\n  return result\n}\n\nfunction extractAttribute(lines: ReadonlyArray<string>, name: string): string | null {\n  for (const line of lines) {\n    if (line.startsWith(`a=${name}:`)) {\n      return line.substring(name.length + 3).trim()\n    }\n  }\n  return null\n}\n\n/**\n * Extract CodecParams (parameter set buffers) from SDP fmtp.\n *\n * H.264: sprop-parameter-sets=<base64-SPS>,<base64-PPS>\n * H.265: sprop-vps=<base64>; sprop-sps=<base64>; sprop-pps=<base64>\n *\n * Returns null if parameter sets are missing from SDP (some cameras\n * only send them in-band via RTP).\n */\nfunction extractCodecParams(\n  codec: 'h264' | 'h265',\n  fmtp: Readonly<Record<string, string>>,\n): CodecParams | null {\n  if (codec === 'h264') {\n    const spropSets = fmtp['sprop-parameter-sets']\n    if (!spropSets) return null\n\n    const parts = spropSets.split(',').filter(s => s.length > 0)\n    if (parts.length < 2) return null\n\n    const parameterSets = parts.map(p => Buffer.from(p, 'base64'))\n    return { codec, parameterSets }\n  }\n\n  // H.265: need all three\n  const vpsB64 = fmtp['sprop-vps']\n  const spsB64 = fmtp['sprop-sps']\n  const ppsB64 = fmtp['sprop-pps']\n  if (!vpsB64 || !spsB64 || !ppsB64) return null\n\n  const parameterSets = [\n    Buffer.from(vpsB64, 'base64'),\n    Buffer.from(spsB64, 'base64'),\n    Buffer.from(ppsB64, 'base64'),\n  ]\n  return { codec, parameterSets }\n}\n\n/** Well-known static RTP payload types (RFC 3551). */\nconst STATIC_PAYLOAD_TYPES = new Map<number, { codec: string; clockRate: number; channels: number }>([\n  [0, { codec: 'PCMU', clockRate: 8000, channels: 1 }],\n  [8, { codec: 'PCMA', clockRate: 8000, channels: 1 }],\n])\n","import * as net from 'node:net'\nimport { createHash, randomUUID } from 'node:crypto'\nimport type { IScopedLogger } from '@camstack/types'\nimport { INTERLEAVED_MAGIC } from './rtsp-types.js'\nimport { parseSdp, extractVideoTrack, extractAudioTrack, resolveTrackUrl } from './sdp-parser.js'\nimport type { ParsedSdp, SdpVideoTrack, SdpAudioTrack } from './sdp-parser.js'\n\n// ── Types ───────────────────────────────────────────────────────────\n\nexport interface RtspClientOptions {\n  /** RTSP URL (rtsp://user:pass@host:port/path). */\n  readonly url: string\n  /** Override username (takes precedence over URL credentials). */\n  readonly username?: string\n  /** Override password (takes precedence over URL credentials). */\n  readonly password?: string\n  /** TCP connect timeout in ms (default 10000). */\n  readonly connectTimeoutMs?: number\n  /** Keepalive interval in ms (default 30000). GET_PARAMETER sent periodically. */\n  readonly keepaliveIntervalMs?: number\n  /** Logger instance. */\n  readonly logger?: IScopedLogger\n}\n\nexport interface RtspClientCallbacks {\n  /** Called for every RTP packet received on a video track. */\n  readonly onVideoRtp?: (packet: Buffer, channel: number) => void\n  /** Called for every RTP packet received on an audio track. */\n  readonly onAudioRtp?: (packet: Buffer, channel: number) => void\n  /** Called when video SDP info is available (after DESCRIBE). */\n  readonly onVideoTrack?: (track: SdpVideoTrack, sdpText: string) => void\n  /** Called when audio SDP info is available (after DESCRIBE). */\n  readonly onAudioTrack?: (track: SdpAudioTrack) => void\n  /** Called when connection is fully established (PLAY response received). */\n  readonly onPlaying?: () => void\n  /** Called on unrecoverable error or connection lost. */\n  readonly onError?: (error: Error) => void\n  /** Called when the client is cleanly torn down. */\n  readonly onTeardown?: () => void\n}\n\ntype RtspState = 'idle' | 'connecting' | 'options' | 'describe' | 'setup-video' | 'setup-audio' | 'play' | 'streaming' | 'teardown' | 'closed'\n\ninterface DigestChallenge {\n  readonly realm: string\n  readonly nonce: string\n  readonly qop?: string\n  readonly opaque?: string\n}\n\n// ── Constants ───────────────────────────────────────────────────────\n\nconst DEFAULT_CONNECT_TIMEOUT_MS = 10_000\nconst DEFAULT_KEEPALIVE_INTERVAL_MS = 30_000\nconst RTSP_PORT = 554\n\n// ── Implementation ──────────────────────────────────────────────────\n\n/**\n * Native RTSP client that connects directly to IP cameras via TCP.\n *\n * Replaces the ffmpeg RTSP reader. Instead of spawning ffmpeg to read\n * RTSP → Annex-B pipe, this client speaks RTSP natively and delivers\n * raw RTP packets — preserving RTP framing that ffmpeg's Annex-B output\n * destroys.\n *\n * State machine: connect → OPTIONS → DESCRIBE → SETUP(video) → SETUP(audio) → PLAY → streaming\n *\n * Features:\n * - TCP interleaved transport (RTP over RTSP connection)\n * - Basic + Digest authentication (auto-detect from 401 response)\n * - GET_PARAMETER keepalive\n * - Clean teardown\n */\nexport class NativeRtspClient {\n  private state: RtspState = 'idle'\n  private socket: net.Socket | null = null\n  private cseq = 0\n  private sessionId: string | null = null\n\n  private readonly url: string\n  private readonly host: string\n  private readonly port: number\n  private readonly username: string\n  private readonly password: string\n  private readonly connectTimeoutMs: number\n  private readonly keepaliveIntervalMs: number\n  private readonly logger: IScopedLogger | undefined\n  private readonly callbacks: RtspClientCallbacks\n\n  /** Parsed SDP from DESCRIBE response. */\n  private parsedSdp: ParsedSdp | null = null\n  private rawSdpText: string | null = null\n  private videoTrack: SdpVideoTrack | null = null\n  private audioTrack: SdpAudioTrack | null = null\n\n  /** Channel assignments for TCP interleaved transport. */\n  private videoRtpChannel = 0\n  private videoRtcpChannel = 1\n  private audioRtpChannel = 2\n  private audioRtcpChannel = 3\n\n  /** Authentication state. */\n  private authMethod: 'none' | 'basic' | 'digest' = 'none'\n  private digestChallenge: DigestChallenge | null = null\n  private digestNc = 0\n\n  /** TCP read buffer for interleaved framing + RTSP response parsing. */\n  private readBuffer: Buffer = Buffer.alloc(0)\n\n  /** Pending RTSP response handler (one request at a time). */\n  private pendingResponse: ((statusCode: number, headers: ReadonlyMap<string, string>, body: string) => void) | null = null\n\n  /** Keepalive timer. */\n  private keepaliveTimer: ReturnType<typeof setInterval> | undefined\n\n  /** Connect timeout. */\n  private connectTimer: ReturnType<typeof setTimeout> | undefined\n\n  /** Whether a teardown was requested (prevents reconnect on close). */\n  private tornDown = false\n\n  constructor(options: RtspClientOptions, callbacks: RtspClientCallbacks) {\n    this.callbacks = callbacks\n    this.logger = options.logger\n    this.connectTimeoutMs = options.connectTimeoutMs ?? DEFAULT_CONNECT_TIMEOUT_MS\n    this.keepaliveIntervalMs = options.keepaliveIntervalMs ?? DEFAULT_KEEPALIVE_INTERVAL_MS\n\n    // Parse URL\n    const parsed = new URL(options.url)\n    this.url = options.url\n    this.host = parsed.hostname\n    this.port = parsed.port ? parseInt(parsed.port, 10) : RTSP_PORT\n    this.username = options.username ?? decodeURIComponent(parsed.username)\n    this.password = options.password ?? decodeURIComponent(parsed.password)\n  }\n\n  /** Start the RTSP session: connect → negotiate → stream. */\n  async connect(): Promise<void> {\n    if (this.state !== 'idle' && this.state !== 'closed') {\n      throw new Error(`Cannot connect in state ${this.state}`)\n    }\n\n    this.tornDown = false\n    this.state = 'connecting'\n    this.cseq = 0\n    this.sessionId = null\n    this.authMethod = 'none'\n    this.digestChallenge = null\n    this.digestNc = 0\n    this.readBuffer = Buffer.alloc(0)\n\n    return new Promise<void>((resolve, reject) => {\n      const socket = net.createConnection({ host: this.host, port: this.port })\n      this.socket = socket\n\n      this.connectTimer = setTimeout(() => {\n        this.connectTimer = undefined\n        socket.destroy(new Error(`RTSP connect timeout after ${this.connectTimeoutMs}ms`))\n      }, this.connectTimeoutMs)\n\n      socket.on('connect', () => {\n        if (this.connectTimer) {\n          clearTimeout(this.connectTimer)\n          this.connectTimer = undefined\n        }\n        this.logger?.info('Connected', { meta: { host: this.host, port: this.port } })\n\n        this.negotiate()\n          .then(() => resolve())\n          .catch((err) => {\n            this.destroy()\n            reject(err)\n          })\n      })\n\n      socket.on('data', (chunk: Buffer) => {\n        this.onData(chunk)\n      })\n\n      socket.on('close', () => {\n        if (this.state === 'streaming' && !this.tornDown) {\n          this.callbacks.onError?.(new Error('RTSP connection closed unexpectedly'))\n        }\n        this.cleanup()\n      })\n\n      socket.on('error', (err) => {\n        if (this.connectTimer) {\n          clearTimeout(this.connectTimer)\n          this.connectTimer = undefined\n        }\n        if (this.state !== 'closed' && this.state !== 'teardown') {\n          this.callbacks.onError?.(err)\n        }\n        this.cleanup()\n        reject(err)\n      })\n    })\n  }\n\n  /** Gracefully tear down the RTSP session. */\n  async teardown(): Promise<void> {\n    if (this.state === 'closed' || this.state === 'idle') return\n    this.tornDown = true\n    this.state = 'teardown'\n\n    try {\n      if (this.socket && !this.socket.destroyed) {\n        await this.sendRequest('TEARDOWN', this.url)\n      }\n    } catch {\n      // Best-effort teardown\n    }\n\n    this.destroy()\n    this.callbacks.onTeardown?.()\n  }\n\n  /** Forcefully destroy the connection without TEARDOWN. */\n  destroy(): void {\n    this.tornDown = true\n    this.cleanup()\n  }\n\n  /** Get the parsed SDP text (available after DESCRIBE). */\n  getSdpText(): string | null { return this.rawSdpText }\n\n  /** Get the parsed SDP (available after DESCRIBE). */\n  getParsedSdp(): ParsedSdp | null { return this.parsedSdp }\n\n  /** Get the video track info (available after DESCRIBE). */\n  getVideoTrack(): SdpVideoTrack | null { return this.videoTrack }\n\n  /** Current client state. */\n  getState(): RtspState { return this.state }\n\n  // ── RTSP Negotiation ──────────────────────────────────────────────\n\n  private async negotiate(): Promise<void> {\n    // Step 1: OPTIONS (optional but standard — probes server capabilities)\n    this.state = 'options'\n    await this.sendRequest('OPTIONS', this.url)\n\n    // Step 2: DESCRIBE — get SDP\n    this.state = 'describe'\n    const describeResult = await this.sendRequest('DESCRIBE', this.url, {\n      'Accept': 'application/sdp',\n    })\n\n    if (describeResult.statusCode === 401) {\n      // Parse auth challenge and retry\n      this.parseAuthChallenge(describeResult.headers)\n      const retryResult = await this.sendRequest('DESCRIBE', this.url, {\n        'Accept': 'application/sdp',\n      })\n      if (retryResult.statusCode !== 200) {\n        throw new Error(`DESCRIBE failed after auth: ${retryResult.statusCode}`)\n      }\n      this.processSdp(retryResult.body)\n    } else if (describeResult.statusCode === 200) {\n      this.processSdp(describeResult.body)\n    } else {\n      throw new Error(`DESCRIBE failed: ${describeResult.statusCode}`)\n    }\n\n    if (!this.videoTrack) {\n      throw new Error('No supported video track in SDP')\n    }\n\n    // Step 3: SETUP video track\n    this.state = 'setup-video'\n    const videoControlUrl = resolveTrackUrl(\n      this.url,\n      this.parsedSdp?.sessionControl ?? null,\n      this.videoTrack.control,\n    )\n\n    const videoSetupResult = await this.sendRequest('SETUP', videoControlUrl, {\n      'Transport': `RTP/AVP/TCP;unicast;interleaved=${this.videoRtpChannel}-${this.videoRtcpChannel}`,\n    })\n\n    if (videoSetupResult.statusCode !== 200) {\n      throw new Error(`SETUP video failed: ${videoSetupResult.statusCode}`)\n    }\n\n    // Extract session ID from first SETUP response\n    const sessionHeader = videoSetupResult.headers.get('session')\n    if (sessionHeader) {\n      // Session header may contain timeout: \"12345678;timeout=60\"\n      this.sessionId = sessionHeader.split(';')[0]!.trim()\n    }\n\n    // Step 4: SETUP audio track (optional)\n    if (this.audioTrack) {\n      this.state = 'setup-audio'\n      const audioControlUrl = resolveTrackUrl(\n        this.url,\n        this.parsedSdp?.sessionControl ?? null,\n        this.audioTrack.control,\n      )\n\n      const audioSetupResult = await this.sendRequest('SETUP', audioControlUrl, {\n        'Transport': `RTP/AVP/TCP;unicast;interleaved=${this.audioRtpChannel}-${this.audioRtcpChannel}`,\n      })\n\n      if (audioSetupResult.statusCode !== 200) {\n        this.logger?.warn('SETUP audio failed — continuing without audio', {\n          meta: { statusCode: audioSetupResult.statusCode },\n        })\n        this.audioTrack = null\n      }\n    }\n\n    // Step 5: PLAY\n    this.state = 'play'\n    const playResult = await this.sendRequest('PLAY', this.url, {\n      'Range': 'npt=0.000-',\n    })\n\n    if (playResult.statusCode !== 200) {\n      throw new Error(`PLAY failed: ${playResult.statusCode}`)\n    }\n\n    this.state = 'streaming'\n    this.logger?.info('RTSP streaming', { meta: { host: this.host, port: this.port } })\n\n    // Start keepalive\n    this.keepaliveTimer = setInterval(() => {\n      this.sendKeepalive().catch((err) => {\n        this.logger?.warn('Keepalive failed', {\n          meta: { error: err instanceof Error ? err.message : String(err) },\n        })\n      })\n    }, this.keepaliveIntervalMs)\n\n    this.callbacks.onPlaying?.()\n  }\n\n  private processSdp(sdpText: string): void {\n    this.rawSdpText = sdpText\n    this.parsedSdp = parseSdp(sdpText)\n    this.videoTrack = extractVideoTrack(this.parsedSdp)\n    this.audioTrack = extractAudioTrack(this.parsedSdp)\n\n    if (this.videoTrack) {\n      this.callbacks.onVideoTrack?.(this.videoTrack, sdpText)\n      this.logger?.info('Video track', {\n        meta: {\n          codec: this.videoTrack.codec,\n          payloadType: this.videoTrack.payloadType,\n          clockRate: this.videoTrack.clockRate,\n        },\n      })\n    }\n\n    if (this.audioTrack) {\n      this.callbacks.onAudioTrack?.(this.audioTrack)\n      this.logger?.info('Audio track', {\n        meta: {\n          codec: this.audioTrack.codec,\n          payloadType: this.audioTrack.payloadType,\n          clockRate: this.audioTrack.clockRate,\n        },\n      })\n    }\n  }\n\n  // ── TCP Data Handling ─────────────────────────────────────────────\n\n  /**\n   * Process incoming TCP data. RTSP TCP interleaved transport mixes:\n   * - RTSP text responses (e.g. \"RTSP/1.0 200 OK\\r\\n...\")\n   * - Interleaved binary frames ($channel + 2-byte length + payload)\n   */\n  private onData(chunk: Buffer): void {\n    this.readBuffer = this.readBuffer.length > 0\n      ? Buffer.concat([this.readBuffer, chunk])\n      : chunk\n\n    while (this.readBuffer.length > 0) {\n      if (this.readBuffer[0] === INTERLEAVED_MAGIC) {\n        // Binary interleaved frame: $ + channel(1) + length(2) + payload\n        if (this.readBuffer.length < 4) return // Need more data\n\n        const channel = this.readBuffer[1]!\n        const length = this.readBuffer.readUInt16BE(2)\n\n        if (this.readBuffer.length < 4 + length) return // Need more data\n\n        const payload = Buffer.from(this.readBuffer.subarray(4, 4 + length))\n        this.readBuffer = this.readBuffer.subarray(4 + length)\n\n        this.handleInterleavedPacket(channel, payload)\n      } else {\n        // RTSP text response\n        const headerEnd = this.readBuffer.indexOf('\\r\\n\\r\\n')\n        if (headerEnd < 0) return // Need more data\n\n        const headerText = this.readBuffer.subarray(0, headerEnd).toString('ascii')\n        let bodyStart = headerEnd + 4\n\n        // Check for Content-Length\n        const clMatch = headerText.match(/Content-Length:\\s*(\\d+)/i)\n        const contentLength = clMatch ? parseInt(clMatch[1]!, 10) : 0\n\n        if (this.readBuffer.length < bodyStart + contentLength) return // Need more data\n\n        const body = contentLength > 0\n          ? this.readBuffer.subarray(bodyStart, bodyStart + contentLength).toString('utf-8')\n          : ''\n\n        this.readBuffer = this.readBuffer.subarray(bodyStart + contentLength)\n\n        this.handleRtspResponse(headerText, body)\n      }\n    }\n  }\n\n  private handleInterleavedPacket(channel: number, payload: Buffer): void {\n    if (channel === this.videoRtpChannel) {\n      this.callbacks.onVideoRtp?.(payload, channel)\n    } else if (channel === this.audioRtpChannel) {\n      this.callbacks.onAudioRtp?.(payload, channel)\n    }\n    // RTCP channels (odd) are silently ignored for now\n  }\n\n  private handleRtspResponse(headerText: string, body: string): void {\n    const lines = headerText.split('\\r\\n')\n    const statusLine = lines[0] ?? ''\n    const statusMatch = statusLine.match(/RTSP\\/1\\.\\d\\s+(\\d+)/)\n    const statusCode = statusMatch ? parseInt(statusMatch[1]!, 10) : 0\n\n    const headers = new Map<string, string>()\n    for (let i = 1; i < lines.length; i++) {\n      const colonIdx = lines[i]!.indexOf(':')\n      if (colonIdx < 0) continue\n      const key = lines[i]!.substring(0, colonIdx).trim().toLowerCase()\n      const value = lines[i]!.substring(colonIdx + 1).trim()\n      headers.set(key, value)\n    }\n\n    if (this.pendingResponse) {\n      const handler = this.pendingResponse\n      this.pendingResponse = null\n      handler(statusCode, headers, body)\n    }\n  }\n\n  // ── RTSP Request Building ─────────────────────────────────────────\n\n  private sendRequest(\n    method: string,\n    uri: string,\n    extraHeaders?: Record<string, string>,\n  ): Promise<{ statusCode: number; headers: ReadonlyMap<string, string>; body: string }> {\n    return new Promise((resolve, reject) => {\n      if (!this.socket || this.socket.destroyed) {\n        reject(new Error('Socket not connected'))\n        return\n      }\n\n      this.cseq++\n      const headers: Record<string, string> = {\n        'CSeq': String(this.cseq),\n        'User-Agent': 'camstack-rtsp-client',\n        ...extraHeaders,\n      }\n\n      if (this.sessionId) {\n        headers['Session'] = this.sessionId\n      }\n\n      // Add authentication header\n      const authHeader = this.buildAuthHeader(method, uri)\n      if (authHeader) {\n        headers['Authorization'] = authHeader\n      }\n\n      let request = `${method} ${uri} RTSP/1.0\\r\\n`\n      for (const [key, value] of Object.entries(headers)) {\n        request += `${key}: ${value}\\r\\n`\n      }\n      request += '\\r\\n'\n\n      this.pendingResponse = (statusCode, responseHeaders, body) => {\n        resolve({ statusCode, headers: responseHeaders, body })\n      }\n\n      // Timeout for response\n      const timeout = setTimeout(() => {\n        if (this.pendingResponse) {\n          this.pendingResponse = null\n          reject(new Error(`RTSP ${method} response timeout`))\n        }\n      }, this.connectTimeoutMs)\n\n      const originalResolve = this.pendingResponse\n      this.pendingResponse = (statusCode, responseHeaders, body) => {\n        clearTimeout(timeout)\n        originalResolve(statusCode, responseHeaders, body)\n      }\n\n      this.socket.write(request)\n    })\n  }\n\n  private async sendKeepalive(): Promise<void> {\n    if (this.state !== 'streaming' || !this.socket) return\n    await this.sendRequest('GET_PARAMETER', this.url)\n  }\n\n  // ── Authentication ────────────────────────────────────────────────\n\n  private parseAuthChallenge(headers: ReadonlyMap<string, string>): void {\n    const wwwAuth = headers.get('www-authenticate')\n    if (!wwwAuth) return\n\n    if (wwwAuth.toLowerCase().startsWith('digest')) {\n      this.authMethod = 'digest'\n      const realm = extractQuoted(wwwAuth, 'realm') ?? ''\n      const nonce = extractQuoted(wwwAuth, 'nonce') ?? ''\n      const qop = extractQuoted(wwwAuth, 'qop')\n      const opaque = extractQuoted(wwwAuth, 'opaque')\n      this.digestChallenge = { realm, nonce, qop, opaque }\n      this.digestNc = 0\n    } else if (wwwAuth.toLowerCase().startsWith('basic')) {\n      this.authMethod = 'basic'\n    }\n  }\n\n  private buildAuthHeader(method: string, uri: string): string | null {\n    if (!this.username && !this.password) return null\n\n    if (this.authMethod === 'basic') {\n      const encoded = Buffer.from(`${this.username}:${this.password}`).toString('base64')\n      return `Basic ${encoded}`\n    }\n\n    if (this.authMethod === 'digest' && this.digestChallenge) {\n      return this.buildDigestAuth(method, uri)\n    }\n\n    // First request — try Basic (some cameras accept it without 401)\n    if (this.username || this.password) {\n      const encoded = Buffer.from(`${this.username}:${this.password}`).toString('base64')\n      return `Basic ${encoded}`\n    }\n\n    return null\n  }\n\n  private buildDigestAuth(method: string, uri: string): string {\n    const { realm, nonce, qop, opaque } = this.digestChallenge!\n    const ha1 = md5(`${this.username}:${realm}:${this.password}`)\n\n    let response: string\n    let authParams: string\n\n    if (qop === 'auth' || qop?.includes('auth')) {\n      this.digestNc++\n      const nc = this.digestNc.toString(16).padStart(8, '0')\n      const cnonce = randomUUID().replace(/-/g, '').slice(0, 16)\n      const ha2 = md5(`${method}:${uri}`)\n      response = md5(`${ha1}:${nonce}:${nc}:${cnonce}:auth:${ha2}`)\n      authParams = `username=\"${this.username}\", realm=\"${realm}\", nonce=\"${nonce}\", uri=\"${uri}\", qop=auth, nc=${nc}, cnonce=\"${cnonce}\", response=\"${response}\"`\n    } else {\n      const ha2 = md5(`${method}:${uri}`)\n      response = md5(`${ha1}:${nonce}:${ha2}`)\n      authParams = `username=\"${this.username}\", realm=\"${realm}\", nonce=\"${nonce}\", uri=\"${uri}\", response=\"${response}\"`\n    }\n\n    if (opaque) {\n      authParams += `, opaque=\"${opaque}\"`\n    }\n\n    return `Digest ${authParams}`\n  }\n\n  // ── Cleanup ───────────────────────────────────────────────────────\n\n  private cleanup(): void {\n    this.state = 'closed'\n\n    if (this.keepaliveTimer) {\n      clearInterval(this.keepaliveTimer)\n      this.keepaliveTimer = undefined\n    }\n\n    if (this.connectTimer) {\n      clearTimeout(this.connectTimer)\n      this.connectTimer = undefined\n    }\n\n    if (this.socket) {\n      try { this.socket.destroy() } catch { /* already closed */ }\n      this.socket = null\n    }\n\n    this.pendingResponse = null\n  }\n}\n\n// ── Utilities ───────────────────────────────────────────────────────\n\nfunction md5(input: string): string {\n  return createHash('md5').update(input).digest('hex')\n}\n\nfunction extractQuoted(header: string, key: string): string | undefined {\n  // Match key=\"value\" or key=value\n  const regex = new RegExp(`${key}=\"([^\"]*)\"`, 'i')\n  const match = header.match(regex)\n  if (match) return match[1]\n\n  // Try unquoted\n  const unquotedRegex = new RegExp(`${key}=([^,\\\\s]+)`, 'i')\n  const unquotedMatch = header.match(unquotedRegex)\n  return unquotedMatch?.[1]\n}\n","/**\n * RFC 4571 framed-RTP-over-TCP reader.\n *\n * Format on the wire (per RFC 4571):\n *\n *   ┌─────────────┬────────────────────────────┐\n *   │ length (BE) │  RTP packet (length bytes) │\n *   │   uint16    │                            │\n *   └─────────────┴────────────────────────────┘\n *\n * No RTSP signalling, no '$' magic byte, no interleaved channel id —\n * just length-prefixed framed RTP. Stream descriptions arrive\n * out-of-band as an SDP string (handed in by the publisher), and\n * the reader routes packets to video / audio depacketizers based on\n * the RTP payload type each track advertised in its `m=` / rtpmap line.\n *\n * Used to ingest the local TCP server spawned by the Reolink lib's\n * `createRfc4571TcpServer` — the standard RFC 4571 length-prefixed\n * RTP-over-TCP wire format.\n *\n * The callbacks here mirror `NativeRtspClient` so the broker's\n * `startNativeRtspReader` glue code can be reused unchanged: same\n * `onVideoTrack` / `onVideoRtp` / `onAudioRtp` / `onConnected` /\n * `onError` / `onDisconnected` handles.\n */\nimport * as net from 'node:net'\nimport { parseSdp, extractVideoTrack, extractAudioTrack } from './sdp-parser.js'\nimport type { ParsedSdp, SdpVideoTrack, SdpAudioTrack } from './sdp-parser.js'\nimport type { IScopedLogger } from '@camstack/types'\n\nexport interface Rfc4571ReaderOptions {\n  /** Loopback URL the lib's RFC 4571 server emitted (`tcp://[user:pass@]host:port`). */\n  readonly url: string\n  /** SDP describing the video + (optional) audio track, returned by the lib's server. */\n  readonly sdp: string\n  readonly logger?: IScopedLogger\n}\n\n/**\n * Reader callbacks. Signature deliberately mirrors `RtspClientCallbacks`\n * (channel arg is always 0 here — RFC 4571 has no interleaved channel\n * concept, but keeping the shape lets the broker reuse the same handler\n * objects across both readers without an adapter layer).\n */\nexport interface Rfc4571ReaderCallbacks {\n  readonly onVideoTrack?: (track: SdpVideoTrack, sdpText: string) => void\n  readonly onAudioTrack?: (track: SdpAudioTrack) => void\n  readonly onVideoRtp?: (rtpData: Buffer, channel: number) => void\n  readonly onAudioRtp?: (rtpData: Buffer, channel: number) => void\n  readonly onPlaying?: () => void\n  readonly onError?: (error: Error) => void\n  readonly onTeardown?: () => void\n}\n\nexport class Rfc4571Reader {\n  private readonly options: Rfc4571ReaderOptions\n  private readonly callbacks: Rfc4571ReaderCallbacks\n  private readonly logger: IScopedLogger | undefined\n  private readonly host: string\n  private readonly port: number\n  private readonly authLine: Buffer | null\n  private socket: net.Socket | null = null\n  /** Concatenated TCP read buffer awaiting full frames. */\n  private readBuffer: Buffer = Buffer.alloc(0)\n  private parsedSdp: ParsedSdp | null = null\n  private videoTrack: SdpVideoTrack | null = null\n  private audioTrack: SdpAudioTrack | null = null\n  private destroyed = false\n\n  constructor(options: Rfc4571ReaderOptions, callbacks: Rfc4571ReaderCallbacks) {\n    this.options = options\n    this.callbacks = callbacks\n    this.logger = options.logger\n    const parsed = new URL(options.url)\n    this.host = parsed.hostname\n    this.port = parseInt(parsed.port || '0', 10)\n    if (this.port === 0) {\n      throw new Error(`Rfc4571Reader: URL must include a port — got '${options.url}'`)\n    }\n    // The lib's RFC 4571 server (when spawned with `requireAuth: true`)\n    // expects `username:password\\n` as the very first bytes on the socket\n    // before any framed RTP. After 5s without a match it drops the\n    // connection. Username/password are percent-encoded in the URL —\n    // decode them back to their raw form before composing the auth line.\n    if (parsed.username || parsed.password) {\n      const u = decodeURIComponent(parsed.username)\n      const p = decodeURIComponent(parsed.password)\n      this.authLine = Buffer.from(`${u}:${p}\\n`, 'utf8')\n    } else {\n      this.authLine = null\n    }\n  }\n\n  async connect(): Promise<void> {\n    // Parse the SDP up front. RFC 4571 has no DESCRIBE phase; the publisher\n    // handed us the stream description out-of-band so the broker knows\n    // codec + payload-type → track routing before the first byte arrives.\n    this.parsedSdp = parseSdp(this.options.sdp)\n    this.videoTrack = extractVideoTrack(this.parsedSdp)\n    this.audioTrack = extractAudioTrack(this.parsedSdp)\n\n    if (this.videoTrack) {\n      this.callbacks.onVideoTrack?.(this.videoTrack, this.options.sdp)\n    }\n    if (this.audioTrack) {\n      this.callbacks.onAudioTrack?.(this.audioTrack)\n    }\n\n    await this.openSocket()\n  }\n\n  destroy(): void {\n    if (this.destroyed) return\n    this.destroyed = true\n    if (this.socket) {\n      try { this.socket.destroy() } catch { /* ignore */ }\n      this.socket = null\n    }\n  }\n\n  private openSocket(): Promise<void> {\n    return new Promise((resolve, reject) => {\n      let settled = false\n      const socket = net.createConnection({ host: this.host, port: this.port }, () => {\n        this.logger?.info('rfc4571: tcp connected', { meta: { host: this.host, port: this.port } })\n        // The lib's server requires `username:password\\n` as the first\n        // bytes when started with `requireAuth: true`. Send it before\n        // anything else — the server starts framed-RTP delivery only\n        // after a successful auth line.\n        if (this.authLine) socket.write(this.authLine)\n        // RFC 4571 has no PLAY phase — `onPlaying` fires the moment the TCP\n        // socket is up, signalling \"data is about to flow\". Mirrors the\n        // semantic the broker's RTSP path expects post-PLAY.\n        this.callbacks.onPlaying?.()\n        settled = true\n        resolve()\n      })\n      this.socket = socket\n      // Disable Nagle: video packets are small, latency matters more than\n      // coalescing for a loopback connection.\n      socket.setNoDelay(true)\n\n      socket.on('data', (chunk) => this.onData(chunk))\n      socket.on('error', (err) => {\n        this.logger?.warn('rfc4571: tcp error', { meta: { error: err.message } })\n        if (!this.destroyed) {\n          this.callbacks.onError?.(err)\n        }\n        if (!settled) reject(err)\n      })\n      socket.on('close', () => {\n        this.logger?.info('rfc4571: tcp closed')\n        if (!this.destroyed) {\n          this.callbacks.onTeardown?.()\n        }\n      })\n    })\n  }\n\n  private onData(chunk: Buffer): void {\n    this.readBuffer = this.readBuffer.length > 0\n      ? Buffer.concat([this.readBuffer, chunk])\n      : chunk\n\n    while (this.readBuffer.length >= 2) {\n      const length = this.readBuffer.readUInt16BE(0)\n      if (this.readBuffer.length < 2 + length) return // need more bytes\n\n      const rtpPacket = Buffer.from(this.readBuffer.subarray(2, 2 + length))\n      this.readBuffer = this.readBuffer.subarray(2 + length)\n\n      this.dispatchRtp(rtpPacket)\n    }\n  }\n\n  private dispatchRtp(rtpPacket: Buffer): void {\n    if (rtpPacket.length < 2) return\n    // RTP payload type is the lower 7 bits of byte 1.\n    const payloadType = rtpPacket[1]! & 0x7f\n    if (this.videoTrack && payloadType === this.videoTrack.payloadType) {\n      this.callbacks.onVideoRtp?.(rtpPacket, 0)\n      return\n    }\n    if (this.audioTrack && payloadType === this.audioTrack.payloadType) {\n      this.callbacks.onAudioRtp?.(rtpPacket, 0)\n      return\n    }\n    // Unknown PT — drop silently. Cameras occasionally send RTCP-style\n    // packets multiplexed on the same channel; the broker doesn't care.\n  }\n}\n","/**\n * FLV Audio/Video tag payload parser (RTMP messages).\n *\n * Standard FLV (Adobe spec) supports H.264 (codecId=7) and AAC.\n *\n * Enhanced RTMP / Enhanced FLV (Veovera spec, used by YouTube Live and most\n * modern Reolink firmware for HEVC streams) extends the layout:\n *   byte 0:\n *     bit 7      : IsExHeader (1 = enhanced layout follows)\n *     bits 4..6  : PacketType  (0=SequenceStart, 1=CodedFrames, 2=SequenceEnd,\n *                              3=CodedFramesX, 4=Metadata, 5=MPEG2TS-SequenceStart)\n *     bits 0..3  : FrameType   (1=key, 2=inter, ...)\n *   bytes 1..4   : FOURCC ('hvc1' / 'hev1' = H.265, 'av01' = AV1, 'vp09' = VP9)\n *   remainder    : codec/packet-type-specific payload\n *\n * H.265 / enhanced-FLV branches added here.\n */\n\n// ── Standard FLV enums ────────────────────────────────────────────────\n\nexport enum VideoCodecId {\n  JPEG = 1,\n  SORENSON_H263 = 2,\n  SCREEN_VIDEO = 3,\n  ON2_VP6 = 4,\n  ON2_VP6_WITH_ALPHA = 5,\n  SCREEN_VIDEO_V2 = 6,\n  H264 = 7,\n  /**\n   * Non-standard codecId for HEVC, predating the Veovera Enhanced-FLV\n   * spec. Used by Reolink RLC-810/820/etc., ZLMediaKit, ffmpeg builds\n   * with HEVC FLV patches, and MediaMTX. Layout matches codecId=7\n   * (H.264) — the only difference is that SEQUENCE_HEADER carries\n   * HEVCDecoderConfigurationRecord and NALU payloads contain HEVC NAL\n   * units.\n   */\n  H265_LEGACY = 12,\n}\n\nexport enum VideoFrameType {\n  KEY = 1,\n  INTER = 2,\n  DISPOSABLE_INTER = 3,\n  GENERATED_KEYFRAME = 4,\n  VIDEO_INFO = 5,\n}\n\nexport enum AvcPacketType {\n  SEQUENCE_HEADER = 0,\n  NALU = 1,\n  END_OF_SEQUENCE = 2,\n}\n\nexport enum AudioSoundFormat {\n  PCM_BE = 0,\n  ADPCM = 1,\n  MP3 = 2,\n  PCM_LE = 3,\n  NELLYMOSER_16K = 4,\n  NELLYMOSER_8K = 5,\n  NELLYMOSER = 6,\n  G711_A = 7,\n  G711_U = 8,\n  AAC = 10,\n  SPEEX = 11,\n  MP3_8K = 14,\n}\n\nexport enum AacPacketType {\n  SEQUENCE_HEADER = 0,\n  RAW = 1,\n}\n\n// ── Enhanced FLV (Veovera) ────────────────────────────────────────────\n\nexport enum EnhancedPacketType {\n  SEQUENCE_START = 0,\n  CODED_FRAMES = 1,\n  SEQUENCE_END = 2,\n  CODED_FRAMES_X = 3,\n  METADATA = 4,\n  MPEG2TS_SEQUENCE_START = 5,\n}\n\n/** Canonical codec discriminator for the parsed video tag. */\nexport type VideoCodec = 'h264' | 'h265'\n\n// ── Decoder configuration record types ────────────────────────────────\n\nexport interface AvcDecoderConfigurationRecord {\n  readonly configurationVersion: number\n  readonly avcProfileIndication: number\n  readonly profileCompatibility: number\n  readonly avcLevelIndication: number\n  /** NALU length-size = lengthSizeMinusOne + 1 (1, 2, or 4) */\n  readonly lengthSizeMinusOne: number\n  readonly sps: ReadonlyArray<Buffer>\n  readonly pps: ReadonlyArray<Buffer>\n}\n\nexport interface HevcDecoderConfigurationRecord {\n  readonly configurationVersion: number\n  readonly generalProfileSpace: number\n  readonly generalTierFlag: number\n  readonly generalProfileIdc: number\n  readonly generalProfileCompatibilityFlags: number\n  readonly generalLevelIdc: number\n  /** NALU length-size = lengthSizeMinusOne + 1 (1, 2, or 4) */\n  readonly lengthSizeMinusOne: number\n  readonly vps: ReadonlyArray<Buffer>\n  readonly sps: ReadonlyArray<Buffer>\n  readonly pps: ReadonlyArray<Buffer>\n}\n\n// ── Audio decoder configuration ───────────────────────────────────────\n\nexport interface AudioSpecificConfig {\n  readonly audioObjectType: number\n  readonly samplingFrequencyIndex: number\n  readonly channelConfiguration: number\n}\n\n// ── Parsed tag types ──────────────────────────────────────────────────\n\ninterface FlvVideoTagBase {\n  readonly frameType: VideoFrameType\n  readonly codec: VideoCodec\n  /** Composition-time offset in milliseconds (0 for enhanced FLV CODED_FRAMES_X). */\n  readonly compositionTime: number\n}\n\nexport interface FlvVideoSequenceHeaderH264 extends FlvVideoTagBase {\n  readonly kind: 'video-sequence-header'\n  readonly codec: 'h264'\n  readonly avcConfig: AvcDecoderConfigurationRecord\n}\n\nexport interface FlvVideoSequenceHeaderH265 extends FlvVideoTagBase {\n  readonly kind: 'video-sequence-header'\n  readonly codec: 'h265'\n  readonly hevcConfig: HevcDecoderConfigurationRecord\n}\n\nexport interface FlvVideoCodedFrames extends FlvVideoTagBase {\n  readonly kind: 'video-coded-frames'\n  readonly nalus: ReadonlyArray<Buffer>\n}\n\nexport interface FlvVideoSequenceEnd extends FlvVideoTagBase {\n  readonly kind: 'video-sequence-end'\n}\n\nexport type FlvVideoTag =\n  | FlvVideoSequenceHeaderH264\n  | FlvVideoSequenceHeaderH265\n  | FlvVideoCodedFrames\n  | FlvVideoSequenceEnd\n\nexport interface FlvAudioSequenceHeader {\n  readonly kind: 'audio-sequence-header'\n  readonly soundFormat: AudioSoundFormat\n  readonly audioSpecificConfig: AudioSpecificConfig\n  /** Raw 2-byte AudioSpecificConfig — useful for SDP / pushAudioInfo. */\n  readonly audioSpecificConfigBytes: Buffer\n}\n\nexport interface FlvAudioRaw {\n  readonly kind: 'audio-raw'\n  readonly soundFormat: AudioSoundFormat\n  readonly data: Buffer\n}\n\nexport type FlvAudioTag = FlvAudioSequenceHeader | FlvAudioRaw\n\n// ── Errors ────────────────────────────────────────────────────────────\n\nexport class FlvParseError extends Error {\n  constructor(message: string) {\n    super(message)\n    this.name = 'FlvParseError'\n  }\n}\n\nexport class UnsupportedCodecError extends FlvParseError {\n  constructor(message: string) {\n    super(message)\n    this.name = 'UnsupportedCodecError'\n  }\n}\n\n// ── Configuration record parsers ──────────────────────────────────────\n\nfunction parseAvcDecoderConfigurationRecord(buffer: Buffer): AvcDecoderConfigurationRecord {\n  if (buffer.length < 7) {\n    throw new FlvParseError('AVCDecoderConfigurationRecord too short')\n  }\n\n  const configurationVersion = buffer.readUInt8(0)\n  const avcProfileIndication = buffer.readUInt8(1)\n  const profileCompatibility = buffer.readUInt8(2)\n  const avcLevelIndication = buffer.readUInt8(3)\n  const lengthSizeMinusOne = buffer.readUInt8(4) & 0x03\n\n  const numSPS = buffer.readUInt8(5) & 0x1f\n  let pos = 6\n\n  const sps: Buffer[] = []\n  for (let i = 0; i < numSPS; i++) {\n    if (pos + 2 > buffer.length) throw new FlvParseError('AVCDecoderConfigurationRecord truncated reading SPS length')\n    const len = buffer.readUInt16BE(pos)\n    pos += 2\n    if (pos + len > buffer.length) throw new FlvParseError('AVCDecoderConfigurationRecord SPS exceeds buffer')\n    sps.push(buffer.subarray(pos, pos + len))\n    pos += len\n  }\n\n  const pps: Buffer[] = []\n  if (pos < buffer.length) {\n    const numPPS = buffer.readUInt8(pos)\n    pos += 1\n    for (let i = 0; i < numPPS; i++) {\n      if (pos + 2 > buffer.length) throw new FlvParseError('AVCDecoderConfigurationRecord truncated reading PPS length')\n      const len = buffer.readUInt16BE(pos)\n      pos += 2\n      if (pos + len > buffer.length) throw new FlvParseError('AVCDecoderConfigurationRecord PPS exceeds buffer')\n      pps.push(buffer.subarray(pos, pos + len))\n      pos += len\n    }\n  }\n\n  return {\n    configurationVersion,\n    avcProfileIndication,\n    profileCompatibility,\n    avcLevelIndication,\n    lengthSizeMinusOne,\n    sps,\n    pps,\n  }\n}\n\n/**\n * Parse HEVCDecoderConfigurationRecord (HVCC) per ISO/IEC 14496-15 §8.\n * Layout (relevant fields only — reserved bits skipped):\n *\n *   u8  configurationVersion       (= 1)\n *   u8  general_profile_space(2) | general_tier_flag(1) | general_profile_idc(5)\n *   u32 general_profile_compatibility_flags\n *   u48 general_constraint_indicator_flags\n *   u8  general_level_idc\n *   u16 reserved(4) | min_spatial_segmentation_idc(12)\n *   u8  reserved(6) | parallelismType(2)\n *   u8  reserved(6) | chromaFormat(2)\n *   u8  reserved(5) | bitDepthLumaMinus8(3)\n *   u8  reserved(5) | bitDepthChromaMinus8(3)\n *   u16 avgFrameRate\n *   u8  constantFrameRate(2) | numTemporalLayers(3) | temporalIdNested(1) | lengthSizeMinusOne(2)\n *   u8  numOfArrays\n *   for each array:\n *     u8  array_completeness(1) | reserved(1) | NAL_unit_type(6)\n *     u16 numNalus\n *     for each nalu: u16 nalUnitLength, then nalUnit bytes\n *\n * The fixed-size header is exactly 22 bytes; arrays start at offset 22.\n * NAL unit types of interest: 32 = VPS, 33 = SPS, 34 = PPS.\n */\nfunction parseHevcDecoderConfigurationRecord(buffer: Buffer): HevcDecoderConfigurationRecord {\n  if (buffer.length < 23) {\n    throw new FlvParseError('HEVCDecoderConfigurationRecord too short')\n  }\n\n  const configurationVersion = buffer.readUInt8(0)\n  const byte1 = buffer.readUInt8(1)\n  const generalProfileSpace = (byte1 >> 6) & 0x03\n  const generalTierFlag = (byte1 >> 5) & 0x01\n  const generalProfileIdc = byte1 & 0x1f\n  const generalProfileCompatibilityFlags = buffer.readUInt32BE(2)\n  // bytes 6..11 = general_constraint_indicator_flags (48 bits) — preserved but not exposed\n  const generalLevelIdc = buffer.readUInt8(12)\n  // bytes 13..14 = min_spatial_segmentation_idc (low 12 bits)\n  // byte 15      = parallelismType\n  // byte 16      = chromaFormat\n  // byte 17      = bitDepthLumaMinus8\n  // byte 18      = bitDepthChromaMinus8\n  // bytes 19..20 = avgFrameRate\n  // byte 21      = constantFrameRate(2) | numTemporalLayers(3) | temporalIdNested(1) | lengthSizeMinusOne(2)\n  const lengthSizeMinusOne = buffer.readUInt8(21) & 0x03\n\n  const numOfArrays = buffer.readUInt8(22)\n  let pos = 23\n\n  const vps: Buffer[] = []\n  const sps: Buffer[] = []\n  const pps: Buffer[] = []\n\n  for (let a = 0; a < numOfArrays; a++) {\n    if (pos + 3 > buffer.length) throw new FlvParseError('HEVCDecoderConfigurationRecord truncated reading array header')\n    const nalUnitType = buffer.readUInt8(pos) & 0x3f\n    const numNalus = buffer.readUInt16BE(pos + 1)\n    pos += 3\n\n    for (let n = 0; n < numNalus; n++) {\n      if (pos + 2 > buffer.length) throw new FlvParseError('HEVCDecoderConfigurationRecord truncated reading NALU length')\n      const len = buffer.readUInt16BE(pos)\n      pos += 2\n      if (pos + len > buffer.length) throw new FlvParseError('HEVCDecoderConfigurationRecord NALU exceeds buffer')\n      const nal = buffer.subarray(pos, pos + len)\n      pos += len\n\n      if (nalUnitType === 32) vps.push(nal)\n      else if (nalUnitType === 33) sps.push(nal)\n      else if (nalUnitType === 34) pps.push(nal)\n      // SEI / other NAL arrays are dropped — not needed for decoder init.\n    }\n  }\n\n  return {\n    configurationVersion,\n    generalProfileSpace,\n    generalTierFlag,\n    generalProfileIdc,\n    generalProfileCompatibilityFlags,\n    generalLevelIdc,\n    lengthSizeMinusOne,\n    vps,\n    sps,\n    pps,\n  }\n}\n\nfunction parseAudioSpecificConfig(buffer: Buffer): AudioSpecificConfig {\n  if (buffer.length < 2) throw new FlvParseError('AudioSpecificConfig too short')\n  const b0 = buffer.readUInt8(0)\n  const b1 = buffer.readUInt8(1)\n  return {\n    audioObjectType: (b0 >> 3) & 0x1f,\n    samplingFrequencyIndex: ((b0 & 0x07) << 1) | ((b1 >> 7) & 0x01),\n    channelConfiguration: (b1 >> 3) & 0x0f,\n  }\n}\n\n// ── NALU extraction (length-prefixed → list of NAL bodies) ────────────\n\nfunction extractLengthPrefixedNalus(payload: Buffer, lengthSize: number): Buffer[] {\n  if (lengthSize < 1 || lengthSize > 4) {\n    throw new FlvParseError(`Invalid NALU length-size: ${lengthSize}`)\n  }\n  const out: Buffer[] = []\n  let pos = 0\n  while (pos + lengthSize <= payload.length) {\n    const len = payload.readUIntBE(pos, lengthSize)\n    pos += lengthSize\n    if (len === 0) continue\n    if (pos + len > payload.length) {\n      throw new FlvParseError(`NALU exceeds payload at pos=${pos} len=${len}`)\n    }\n    out.push(payload.subarray(pos, pos + len))\n    pos += len\n  }\n  return out\n}\n\n// ── AAC ADTS sniffing (audio path runtime helper) ─────────────────────\n\nconst ADTS_SYNC_WORD = 0xfff\n\n/**\n * Returns true when `buffer` starts with the 12-bit AAC ADTS sync word\n * (0xFFF). Used by the RTMP reader to disambiguate AAC raw frames coming\n * directly from cameras that bypass the AAC sequence header step.\n */\nexport function looksLikeAdts(buffer: Buffer): boolean {\n  if (buffer.length < 2) return false\n  return ((buffer.readUInt8(0) << 4) | (buffer.readUInt8(1) >> 4)) === ADTS_SYNC_WORD\n}\n\n// ── Public entry points ───────────────────────────────────────────────\n\n/**\n * Parse an RTMP video tag (FLV message-type 9) payload.\n *\n * Throws `UnsupportedCodecError` for codecs we cannot route to AnnexB\n * downstream (anything other than H.264 / H.265).\n */\nexport function parseFlvVideoTag(buffer: Buffer): FlvVideoTag {\n  if (buffer.length < 1) throw new FlvParseError('Video tag empty')\n  const byte0 = buffer.readUInt8(0)\n  const isExHeader = (byte0 & 0x80) !== 0\n\n  if (isExHeader) {\n    return parseEnhancedVideoTag(buffer)\n  }\n\n  // Standard FLV layout (codecId=7 H.264, or codecId=12 H.265-legacy)\n  const frameType = (byte0 >> 4) as VideoFrameType\n  const codecId = byte0 & 0x0f\n\n  const codec: VideoCodec | null =\n    codecId === VideoCodecId.H264 ? 'h264'\n      : codecId === VideoCodecId.H265_LEGACY ? 'h265'\n        : null\n  if (codec === null) {\n    throw new UnsupportedCodecError(`FLV video codecId=${codecId} not supported (only H.264 / H.265-legacy / H.265-enhanced)`)\n  }\n\n  return parseStandardLayoutVideoTag(buffer, codec, frameType, 4)\n}\n\n/**\n * Shared parser for the codecId=7 (H.264) / codecId=12 (H.265 legacy)\n * layout: byte0 frameType+codecId, byte1 packetType, bytes2..4 composition\n * time offset (signed), then payload (decoder config or length-prefixed\n * NALUs).\n */\nfunction parseStandardLayoutVideoTag(\n  buffer: Buffer,\n  codec: VideoCodec,\n  frameType: VideoFrameType,\n  lengthSize: number,\n): FlvVideoTag {\n  if (buffer.length < 5) throw new FlvParseError(`${codec.toUpperCase()} video tag header truncated`)\n  const packetType = buffer.readUInt8(1) as AvcPacketType\n  const compositionTime = buffer.readIntBE(2, 3)\n\n  if (packetType === AvcPacketType.SEQUENCE_HEADER) {\n    if (codec === 'h264') {\n      const avcConfig = parseAvcDecoderConfigurationRecord(buffer.subarray(5))\n      return { kind: 'video-sequence-header', codec: 'h264', frameType, compositionTime, avcConfig }\n    }\n    const hevcConfig = parseHevcDecoderConfigurationRecord(buffer.subarray(5))\n    return { kind: 'video-sequence-header', codec: 'h265', frameType, compositionTime, hevcConfig }\n  }\n\n  if (packetType === AvcPacketType.NALU) {\n    const nalus = extractLengthPrefixedNalus(buffer.subarray(5), lengthSize)\n    return { kind: 'video-coded-frames', codec, frameType, compositionTime, nalus }\n  }\n\n  if (packetType === AvcPacketType.END_OF_SEQUENCE) {\n    return { kind: 'video-sequence-end', codec, frameType, compositionTime }\n  }\n\n  throw new FlvParseError(`Unknown AVC packet type: ${packetType}`)\n}\n\n/**\n * Same as `parseFlvVideoTag` but uses the supplied NALU length-size\n * when extracting CodedFrames payloads. Callers that have parsed a\n * prior SequenceHeader should prefer this overload for correctness on\n * cameras that ship a non-default `lengthSizeMinusOne`.\n */\nexport function parseFlvVideoTagWithLengthSize(buffer: Buffer, lengthSize: number): FlvVideoTag {\n  if (buffer.length < 1) throw new FlvParseError('Video tag empty')\n  const byte0 = buffer.readUInt8(0)\n  const isExHeader = (byte0 & 0x80) !== 0\n\n  if (isExHeader) {\n    return parseEnhancedVideoTag(buffer, lengthSize)\n  }\n\n  const frameType = (byte0 >> 4) as VideoFrameType\n  const codecId = byte0 & 0x0f\n  const codec: VideoCodec | null =\n    codecId === VideoCodecId.H264 ? 'h264'\n      : codecId === VideoCodecId.H265_LEGACY ? 'h265'\n        : null\n  if (codec === null) {\n    throw new UnsupportedCodecError(`FLV video codecId=${codecId} not supported`)\n  }\n  return parseStandardLayoutVideoTag(buffer, codec, frameType, lengthSize)\n}\n\nfunction parseEnhancedVideoTag(buffer: Buffer, lengthSize: number = 4): FlvVideoTag {\n  // byte 0 layout: bit7 IsExHeader | bits4..6 PacketType | bits0..3 FrameType\n  if (buffer.length < 5) throw new FlvParseError('Enhanced FLV video tag truncated (need FOURCC)')\n  const byte0 = buffer.readUInt8(0)\n  const packetType = ((byte0 >> 4) & 0x07) as EnhancedPacketType\n  const frameType = (byte0 & 0x0f) as VideoFrameType\n\n  const fourcc = buffer.subarray(1, 5).toString('ascii')\n  const codec = mapFourCcToCodec(fourcc)\n\n  if (codec !== 'h265') {\n    throw new UnsupportedCodecError(`Enhanced FLV FOURCC '${fourcc}' not supported (only hvc1/hev1)`)\n  }\n\n  if (packetType === EnhancedPacketType.SEQUENCE_START) {\n    const hevcConfig = parseHevcDecoderConfigurationRecord(buffer.subarray(5))\n    return {\n      kind: 'video-sequence-header',\n      codec: 'h265',\n      frameType,\n      compositionTime: 0,\n      hevcConfig,\n    }\n  }\n\n  if (packetType === EnhancedPacketType.CODED_FRAMES) {\n    if (buffer.length < 8) throw new FlvParseError('Enhanced FLV CODED_FRAMES truncated')\n    const compositionTime = buffer.readIntBE(5, 3)\n    const nalus = extractLengthPrefixedNalus(buffer.subarray(8), lengthSize)\n    return { kind: 'video-coded-frames', codec: 'h265', frameType, compositionTime, nalus }\n  }\n\n  if (packetType === EnhancedPacketType.CODED_FRAMES_X) {\n    const nalus = extractLengthPrefixedNalus(buffer.subarray(5), lengthSize)\n    return { kind: 'video-coded-frames', codec: 'h265', frameType, compositionTime: 0, nalus }\n  }\n\n  if (packetType === EnhancedPacketType.SEQUENCE_END) {\n    return { kind: 'video-sequence-end', codec: 'h265', frameType, compositionTime: 0 }\n  }\n\n  throw new FlvParseError(`Unsupported enhanced FLV packet type: ${packetType}`)\n}\n\nfunction mapFourCcToCodec(fourcc: string): VideoCodec | null {\n  if (fourcc === 'hvc1' || fourcc === 'hev1') return 'h265'\n  if (fourcc === 'avc1') return 'h264'\n  return null\n}\n\n/** Parse an RTMP audio tag (FLV message-type 8) payload. */\nexport function parseFlvAudioTag(buffer: Buffer): FlvAudioTag {\n  if (buffer.length < 1) throw new FlvParseError('Audio tag empty')\n  const byte0 = buffer.readUInt8(0)\n  const soundFormat = ((byte0 >> 4) & 0x0f) as AudioSoundFormat\n\n  if (soundFormat !== AudioSoundFormat.AAC) {\n    return { kind: 'audio-raw', soundFormat, data: buffer.subarray(1) }\n  }\n\n  if (buffer.length < 2) throw new FlvParseError('AAC audio tag truncated')\n  const aacPacketType = buffer.readUInt8(1) as AacPacketType\n\n  if (aacPacketType === AacPacketType.SEQUENCE_HEADER) {\n    const ascBytes = buffer.subarray(2)\n    const audioSpecificConfig = parseAudioSpecificConfig(ascBytes)\n    return {\n      kind: 'audio-sequence-header',\n      soundFormat,\n      audioSpecificConfig,\n      audioSpecificConfigBytes: Buffer.from(ascBytes),\n    }\n  }\n\n  return {\n    kind: 'audio-raw',\n    soundFormat,\n    data: buffer.subarray(2),\n  }\n}\n\n// ── AnnexB assembly helper ────────────────────────────────────────────\n\n/** 4-byte AnnexB start code. */\nconst ANNEXB_START_CODE = Buffer.from([0x00, 0x00, 0x00, 0x01])\n\n/**\n * Assemble a sequence of NAL unit bodies into a single AnnexB buffer.\n * Each NAL is prefixed with `00 00 00 01`.\n */\nexport function nalusToAnnexB(nalus: ReadonlyArray<Buffer>): Buffer {\n  if (nalus.length === 0) return Buffer.alloc(0)\n  const parts: Buffer[] = []\n  for (const nal of nalus) {\n    parts.push(ANNEXB_START_CODE, nal)\n  }\n  return Buffer.concat(parts)\n}\n\n/**\n * Build the AnnexB parameter-set prefix for an H.264 IDR — concatenates\n * SPS + PPS NAL bodies with start codes.\n */\nexport function buildH264ParamSetsAnnexB(record: AvcDecoderConfigurationRecord): Buffer {\n  return nalusToAnnexB([...record.sps, ...record.pps])\n}\n\n/**\n * Build the AnnexB parameter-set prefix for an H.265 IDR — concatenates\n * VPS + SPS + PPS NAL bodies with start codes.\n */\nexport function buildH265ParamSetsAnnexB(record: HevcDecoderConfigurationRecord): Buffer {\n  return nalusToAnnexB([...record.vps, ...record.sps, ...record.pps])\n}\n\n// ── Frequency-index → sample-rate (AAC) ───────────────────────────────\n\nconst AAC_SAMPLE_RATES: ReadonlyArray<number> = [\n  96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,\n  16000, 12000, 11025, 8000, 7350,\n]\n\n/**\n * Resolve the sample-rate for an AAC `samplingFrequencyIndex`. Returns\n * `null` for the reserved indices 13–15 so the caller can decide whether\n * to fall back or reject the stream.\n */\nexport function aacSamplingIndexToHz(index: number): number | null {\n  return AAC_SAMPLE_RATES[index] ?? null\n}\n","/**\n * RTMP client (TCP, handshake, AMF0 connect/createStream/play, chunk parser).\n *\n * RTMP client with the following adaptations from the original implementation:\n *   - `console` replaced with project `IScopedLogger`.\n *   - Inlined `readLength` helper (no external dependency).\n *   - Strict TS — AMF0 encoder accepts a tagged union instead of `any`.\n *   - Async-friendly `destroy()` for the broker's reconnect cycle.\n */\n\nimport { Socket, type SocketConnectOpts } from 'net'\nimport { Readable } from 'stream'\nimport { once } from 'events'\nimport type { IScopedLogger } from '@camstack/types'\n\n// ── Inlined readLength helper ──\n\nclass StreamEndError extends Error {\n  constructor(where: string) {\n    super(`stream ended: ${where}`)\n    this.name = 'StreamEndError'\n  }\n}\n\nasync function readLength(readable: Readable, length: number): Promise<Buffer> {\n  if (readable.readableEnded || readable.destroyed) {\n    throw new StreamEndError('readLength start')\n  }\n  if (!length) return Buffer.alloc(0)\n\n  const initial = readable.read(length) as Buffer | null\n  if (initial) return initial\n\n  return new Promise<Buffer>((resolve, reject) => {\n    const onReadable = (): void => {\n      const chunk = readable.read(length) as Buffer | null\n      if (chunk) {\n        cleanup()\n        resolve(chunk)\n        return\n      }\n      if (readable.readableEnded || readable.destroyed) {\n        cleanup()\n        reject(new StreamEndError('readLength readable'))\n      }\n    }\n    const onEnd = (): void => {\n      cleanup()\n      reject(new StreamEndError('readLength end'))\n    }\n    const cleanup = (): void => {\n      readable.removeListener('readable', onReadable)\n      readable.removeListener('end', onEnd)\n    }\n    readable.on('readable', onReadable)\n    readable.on('end', onEnd)\n  })\n}\n\n// ── RTMP wire enums ────────────────────────────────────────────────────\n\nconst HANDSHAKE_SIZE = 1536\nconst RTMP_VERSION = 3\nconst DEFAULT_RTMP_PORT = 1935\n\nenum ChunkFormat {\n  TYPE_0 = 0,\n  TYPE_1 = 1,\n  TYPE_2 = 2,\n  TYPE_3 = 3,\n}\n\nenum RtmpMessageType {\n  CHUNK_SIZE = 1,\n  ABORT = 2,\n  ACKNOWLEDGEMENT = 3,\n  USER_CONTROL = 4,\n  WINDOW_ACKNOWLEDGEMENT_SIZE = 5,\n  SET_PEER_BANDWIDTH = 6,\n  AUDIO = 8,\n  VIDEO = 9,\n  DATA_AMF0 = 18,\n  COMMAND_AMF0 = 20,\n}\n\ninterface ChunkStreamState {\n  chunkStreamId: number\n  messageStreamId: number\n  messageLength: number\n  messageTypeId: number\n  timestamp: number\n  messageData: Buffer[]\n  totalReceived: number\n  hasExtendedTimestamp: boolean\n}\n\n// ── AMF0 encoder (strict types) ────────────────────────────────────────\n\ntype Amf0Value =\n  | number\n  | string\n  | boolean\n  | null\n  | undefined\n  | { readonly [key: string]: Amf0Value }\n\nfunction encodeAmf0(value: Amf0Value): Buffer {\n  if (typeof value === 'number') {\n    const buf = Buffer.alloc(9)\n    buf[0] = 0x00 // Number marker\n    buf.writeDoubleBE(value, 1)\n    return buf\n  }\n  if (typeof value === 'string') {\n    const buf = Buffer.alloc(3 + value.length)\n    buf[0] = 0x02 // String marker\n    buf.writeUInt16BE(value.length, 1)\n    buf.write(value, 3, 'utf8')\n    return buf\n  }\n  if (typeof value === 'boolean') {\n    const buf = Buffer.alloc(2)\n    buf[0] = 0x01 // Boolean marker\n    buf[1] = value ? 1 : 0\n    return buf\n  }\n  if (value === null || value === undefined) {\n    return Buffer.from([0x05]) // Null marker\n  }\n  // Object\n  const parts: Buffer[] = [Buffer.from([0x03])] // Object marker\n  for (const [key, val] of Object.entries(value)) {\n    const keyBuf = Buffer.alloc(2 + key.length)\n    keyBuf.writeUInt16BE(key.length, 0)\n    keyBuf.write(key, 2, 'utf8')\n    parts.push(keyBuf)\n    parts.push(encodeAmf0(val))\n  }\n  parts.push(Buffer.from([0x00, 0x00, 0x09])) // End-of-object marker\n  return Buffer.concat(parts)\n}\n\nfunction encodeAmf0Command(\n  commandName: string,\n  transactionId: number,\n  commandObject: Amf0Value,\n  ...args: ReadonlyArray<Amf0Value>\n): Buffer {\n  const parts: Buffer[] = [\n    encodeAmf0(commandName),\n    encodeAmf0(transactionId),\n    encodeAmf0(commandObject),\n  ]\n  for (const arg of args) parts.push(encodeAmf0(arg))\n  return Buffer.concat(parts)\n}\n\nfunction writeUInt24BE(buffer: Buffer, value: number, offset: number): void {\n  buffer[offset] = (value >> 16) & 0xff\n  buffer[offset + 1] = (value >> 8) & 0xff\n  buffer[offset + 2] = value & 0xff\n}\n\n// ── Public RTMP packet shape ───────────────────────────────────────────\n\nexport type RtmpMediaCodec = 'video' | 'audio'\n\nexport interface RtmpMediaPacket {\n  readonly codec: RtmpMediaCodec\n  readonly packet: Buffer\n  /** Timestamp in milliseconds, as advertised by the server. */\n  readonly timestamp: number\n}\n\nexport class RtmpClientClosedError extends Error {\n  constructor(message: string) {\n    super(message)\n    this.name = 'RtmpClientClosedError'\n  }\n}\n\n// ── RtmpClient ─────────────────────────────────────────────────────────\n\nexport class RtmpClient {\n  readonly url: string\n  private readonly logger: IScopedLogger | undefined\n  private socket: Socket\n  private chunkSize = 128\n  private outgoingChunkSize = 128\n  private windowAckSize = 5_000_000\n  private streamId = 0\n  private lastAcknowledgementBytes = 0\n  private totalBytesReceived = 0\n  private transactionId = 1\n  private chunkStreams: Map<number, ChunkStreamState> = new Map()\n  private destroyed = false\n\n  constructor(url: string, logger?: IScopedLogger) {\n    this.url = url\n    this.logger = logger\n    this.socket = new Socket()\n  }\n\n  /** Connect, handshake, and play. Throws on any failure. */\n  async setup(): Promise<void> {\n    await this.connect()\n\n    this.logger?.debug('rtmp connect command sent')\n    await this.sendConnect()\n\n    while (true) {\n      const msg = await this.readMessage()\n      const { messageTypeId } = msg.chunkStream\n      if (messageTypeId === RtmpMessageType.WINDOW_ACKNOWLEDGEMENT_SIZE) continue\n      if (messageTypeId === RtmpMessageType.SET_PEER_BANDWIDTH) continue\n      if (messageTypeId === RtmpMessageType.CHUNK_SIZE) {\n        this.chunkSize = msg.message.readUInt32BE(0)\n        this.logger?.debug('rtmp server chunk size', { meta: { chunkSize: this.chunkSize } })\n        continue\n      }\n      if (messageTypeId === RtmpMessageType.COMMAND_AMF0) {\n        const commandName = msg.message.subarray(3, 10).toString('utf8')\n        if (commandName === '_result') {\n          this.logger?.debug('rtmp connect _result received')\n          break\n        }\n        throw new Error(`Unexpected RTMP command: ${commandName}`)\n      }\n      throw new Error(`Unexpected RTMP message type: ${messageTypeId}`)\n    }\n\n    this.sendWindowAckSize(5_000_000)\n\n    this.streamId = await this.sendCreateStream()\n    const createStreamResult = await this.readMessage()\n    const { messageTypeId } = createStreamResult.chunkStream\n    if (messageTypeId !== RtmpMessageType.COMMAND_AMF0) {\n      throw new Error(`Unexpected message type for createStream result: ${messageTypeId}`)\n    }\n    this.logger?.debug('rtmp createStream _result received')\n\n    const parsedUrl = new URL(this.url)\n    const parts = parsedUrl.pathname.split('/')\n    const streamName = parts.length > 2 ? parts.slice(2).join('/') : ''\n    const playPath = streamName + parsedUrl.search\n\n    const getStreamLengthData = encodeAmf0Command('getStreamLength', this.transactionId++, null, playPath)\n    this.sendMessage(5, 0, RtmpMessageType.COMMAND_AMF0, 0, getStreamLengthData)\n\n    this.sendPlay(this.streamId, playPath)\n    this.setBufferLength(this.streamId, 3000)\n\n    this.logger?.info('rtmp play sent', { meta: { playPath } })\n  }\n\n  /** Async-iterator over received video/audio packets. */\n  async *readLoop(): AsyncGenerator<RtmpMediaPacket, void, void> {\n    while (!this.destroyed) {\n      const msg = await this.readMessage()\n      const typeId = msg.chunkStream.messageTypeId\n      if (typeId === RtmpMessageType.VIDEO) {\n        yield { codec: 'video', packet: msg.message, timestamp: msg.chunkStream.timestamp }\n      } else if (typeId === RtmpMessageType.AUDIO) {\n        yield { codec: 'audio', packet: msg.message, timestamp: msg.chunkStream.timestamp }\n      } else {\n        this.logger?.debug('rtmp ignoring message', { meta: { messageTypeId: typeId } })\n      }\n    }\n  }\n\n  destroy(): void {\n    if (this.destroyed) return\n    this.destroyed = true\n    try {\n      this.socket.destroy()\n    } catch {\n      // Socket already destroyed — ignore.\n    }\n  }\n\n  // ── Connection / handshake ────────────────────────────────────────────\n\n  private async connect(): Promise<void> {\n    const parsedUrl = new URL(this.url)\n    const host = parsedUrl.hostname\n    const port = parseInt(parsedUrl.port || `${DEFAULT_RTMP_PORT}`, 10) || DEFAULT_RTMP_PORT\n\n    this.logger?.debug('rtmp tcp connecting', { meta: { host, port } })\n\n    this.socket.on('error', (err) => {\n      this.logger?.warn('rtmp socket error', { meta: { error: err.message } })\n    })\n\n    await new Promise<void>((resolve, reject) => {\n      const onError = (err: Error): void => reject(err)\n      this.socket.once('error', onError)\n      const opts: SocketConnectOpts = { host, port }\n      this.socket.connect(opts, () => {\n        this.socket.removeListener('error', onError)\n        resolve()\n      })\n    })\n\n    await this.performHandshake()\n  }\n\n  private async performHandshake(): Promise<void> {\n    // C0\n    const c0 = Buffer.from([RTMP_VERSION])\n    // C1: timestamp(4) + zero(4) + random(1528)\n    const c1 = Buffer.alloc(HANDSHAKE_SIZE)\n    c1.writeUInt32BE(Math.floor(Date.now() / 1000), 0)\n    c1.writeUInt32BE(0, 4)\n    this.socket.write(Buffer.concat([c0, c1]))\n\n    const s0 = await readLength(this.socket, 1)\n    const serverVersion = s0.readUInt8(0)\n    if (serverVersion !== RTMP_VERSION) {\n      throw new Error(`Unsupported RTMP server version: ${serverVersion}`)\n    }\n    const s1 = await readLength(this.socket, HANDSHAKE_SIZE)\n    await readLength(this.socket, HANDSHAKE_SIZE) // S2 (discarded)\n\n    // C2 = echo of S1\n    this.socket.write(s1)\n  }\n\n  // ── Chunk parser ──────────────────────────────────────────────────────\n\n  private async readMessage(): Promise<{ message: Buffer; chunkStream: ChunkStreamState }> {\n    const stream = this.socket\n    while (true) {\n      const basicHeader = await readLength(stream, 1)\n      const basic0 = basicHeader.readUInt8(0)\n      const fmt = (basic0 >> 6) & 0x03\n      let csId = basic0 & 0x3f\n\n      if (csId === 0) {\n        const secondByte = await readLength(stream, 1)\n        csId = secondByte.readUInt8(0) + 64\n      } else if (csId === 1) {\n        const bytes = await readLength(stream, 2)\n        csId = (bytes.readUInt8(1) << 8) | (bytes.readUInt8(0) + 64)\n      }\n\n      let chunkStream = this.chunkStreams.get(csId)\n      if (!chunkStream) {\n        chunkStream = {\n          chunkStreamId: csId,\n          messageStreamId: 0,\n          messageLength: 0,\n          messageTypeId: 0,\n          timestamp: 0,\n          messageData: [],\n          totalReceived: 0,\n          hasExtendedTimestamp: false,\n        }\n        this.chunkStreams.set(csId, chunkStream)\n      }\n\n      let hasExtendedTimestamp = false\n      let headerSize: number\n\n      if (fmt === ChunkFormat.TYPE_0) {\n        headerSize = 11\n        const header = await readLength(stream, 11)\n        const timestamp = header.readUIntBE(0, 3)\n        const messageLength = header.readUIntBE(3, 3)\n        const messageTypeId = header.readUInt8(6)\n        const messageStreamId = header.readUInt32LE(7)\n\n        chunkStream.messageStreamId = messageStreamId\n        chunkStream.messageLength = messageLength\n        chunkStream.messageTypeId = messageTypeId\n        chunkStream.timestamp = timestamp\n        chunkStream.totalReceived = 0\n        chunkStream.messageData = []\n\n        if (timestamp >= 0xffffff) {\n          hasExtendedTimestamp = true\n          chunkStream.hasExtendedTimestamp = true\n        }\n      } else if (fmt === ChunkFormat.TYPE_1) {\n        headerSize = 7\n        const header = await readLength(stream, 7)\n        const timestampDelta = header.readUIntBE(0, 3)\n        const messageLength = header.readUIntBE(3, 3)\n        const messageTypeId = header.readUInt8(6)\n        chunkStream.messageLength = messageLength\n        chunkStream.messageTypeId = messageTypeId\n        chunkStream.timestamp += timestampDelta\n        chunkStream.totalReceived = 0\n        chunkStream.messageData = []\n        if (timestampDelta >= 0xffffff) {\n          hasExtendedTimestamp = true\n          chunkStream.hasExtendedTimestamp = true\n        }\n      } else if (fmt === ChunkFormat.TYPE_2) {\n        headerSize = 3\n        const header = await readLength(stream, 3)\n        const timestampDelta = header.readUIntBE(0, 3)\n        chunkStream.timestamp += timestampDelta\n        chunkStream.totalReceived = 0\n        chunkStream.messageData = []\n        if (timestampDelta >= 0xffffff) {\n          hasExtendedTimestamp = true\n          chunkStream.hasExtendedTimestamp = true\n        }\n      } else {\n        headerSize = 0\n        if (chunkStream.totalReceived === 0) {\n          throw new Error('Type 3 chunk but no previous chunk in stream')\n        }\n      }\n\n      if (hasExtendedTimestamp || chunkStream.hasExtendedTimestamp) {\n        const extTs = await readLength(stream, 4)\n        const extendedTimestamp = extTs.readUInt32BE(0)\n        if (fmt === ChunkFormat.TYPE_0) {\n          chunkStream.timestamp = extendedTimestamp\n        }\n      }\n\n      const remainingInMessage = chunkStream.messageLength - chunkStream.totalReceived\n      const chunkDataSize = Math.min(this.chunkSize, remainingInMessage)\n\n      const MAX_CHUNK_SIZE = 1024 * 1024\n      if (chunkDataSize > MAX_CHUNK_SIZE) {\n        throw new Error(`Chunk size ${chunkDataSize} exceeds max ${MAX_CHUNK_SIZE}`)\n      }\n\n      const chunkData = await readLength(stream, chunkDataSize)\n      chunkStream.messageData.push(chunkData)\n      chunkStream.totalReceived += chunkDataSize\n\n      const extTimestampSize = (hasExtendedTimestamp || chunkStream.hasExtendedTimestamp) ? 4 : 0\n      this.totalBytesReceived += 1 + headerSize + extTimestampSize + chunkDataSize\n      this.sendAcknowledgementIfNeeded()\n\n      if (chunkStream.totalReceived >= chunkStream.messageLength) {\n        const message = Buffer.concat(chunkStream.messageData)\n        chunkStream.messageData = []\n        chunkStream.totalReceived = 0\n        chunkStream.hasExtendedTimestamp = false\n        return { chunkStream, message }\n      }\n    }\n  }\n\n  private sendAcknowledgementIfNeeded(): void {\n    const bytesToAck = this.totalBytesReceived - this.lastAcknowledgementBytes\n    if (bytesToAck >= this.windowAckSize) {\n      this.lastAcknowledgementBytes = this.totalBytesReceived\n      const data = Buffer.alloc(4)\n      data.writeUInt32BE(this.lastAcknowledgementBytes & 0xffffffff, 0)\n      this.sendMessage(2, 0, RtmpMessageType.ACKNOWLEDGEMENT, 0, data)\n    }\n  }\n\n  // ── Outgoing message helpers ─────────────────────────────────────────\n\n  private sendMessage(\n    chunkStreamId: number,\n    messageStreamId: number,\n    messageTypeId: number,\n    timestamp: number,\n    data: Buffer,\n  ): void {\n    const chunks: Buffer[] = []\n    let offset = 0\n\n    while (offset < data.length) {\n      const chunkDataSize = Math.min(this.outgoingChunkSize, data.length - offset)\n      const isType0 = offset === 0\n      const headerSize = isType0 ? 12 : 1\n      const header = Buffer.alloc(headerSize)\n\n      if (chunkStreamId < 64) {\n        header[0] = ((isType0 ? ChunkFormat.TYPE_0 : ChunkFormat.TYPE_3) << 6) | chunkStreamId\n      } else {\n        header[0] = ((isType0 ? ChunkFormat.TYPE_0 : ChunkFormat.TYPE_3) << 6) | 1\n      }\n\n      if (isType0) {\n        writeUInt24BE(header, timestamp, 1)\n        writeUInt24BE(header, data.length, 4)\n        header[7] = messageTypeId\n        header.writeUInt32LE(messageStreamId, 8)\n      }\n\n      chunks.push(header, data.subarray(offset, offset + chunkDataSize))\n      offset += chunkDataSize\n    }\n\n    for (const chunk of chunks) this.socket.write(chunk)\n  }\n\n  private async sendConnect(): Promise<void> {\n    const parsedUrl = new URL(this.url)\n    const tcUrl = `${parsedUrl.protocol}//${parsedUrl.host}/${parsedUrl.pathname.split('/')[1]}`\n    const connectObject = {\n      app: parsedUrl.pathname.split('/')[1],\n      flashVer: 'LNX 9,0,124,2',\n      tcUrl,\n      fpad: false,\n      capabilities: 15,\n      audioCodecs: 4071,\n      videoCodecs: 252,\n      videoFunction: 1,\n    }\n    const data = encodeAmf0Command('connect', this.transactionId++, connectObject)\n    this.sendMessage(3, 0, RtmpMessageType.COMMAND_AMF0, 0, data)\n  }\n\n  private async sendCreateStream(): Promise<number> {\n    const data = encodeAmf0Command('createStream', this.transactionId++, null)\n    this.sendMessage(3, 0, RtmpMessageType.COMMAND_AMF0, 0, data)\n    return 1\n  }\n\n  private sendPlay(streamId: number, playPath: string): void {\n    const data = encodeAmf0Command('play', this.transactionId++, null, playPath, -2000)\n    this.sendMessage(4, streamId, RtmpMessageType.COMMAND_AMF0, 0, data)\n  }\n\n  private setBufferLength(streamId: number, bufferLength: number): void {\n    const data = Buffer.alloc(10)\n    data.writeUInt16BE(3, 0)\n    data.writeUInt32BE(streamId, 2)\n    data.writeUInt32BE(bufferLength, 6)\n    this.sendMessage(2, 0, RtmpMessageType.USER_CONTROL, 1, data)\n  }\n\n  private sendWindowAckSize(windowSize: number): void {\n    const data = Buffer.alloc(4)\n    data.writeUInt32BE(windowSize, 0)\n    this.sendMessage(2, 0, RtmpMessageType.WINDOW_ACKNOWLEDGEMENT_SIZE, 0, data)\n  }\n\n  /** Wait for the underlying socket to close. */\n  async waitForClose(): Promise<void> {\n    if (this.socket.destroyed) return\n    await once(this.socket, 'close')\n  }\n\n  get isDestroyed(): boolean {\n    return this.destroyed\n  }\n}\n","/**\n * RtmpReader — drives an `RtmpClient`, parses FLV tags from each RTMP\n * VIDEO/AUDIO message, builds AnnexB access units, and surfaces them to\n * the stream-broker via the supplied callbacks.\n *\n * Codec support:\n *   - Video: H.264 (standard FLV) and H.265 (Enhanced FLV / Veovera spec\n *           with FOURCC 'hvc1' or 'hev1'). Param sets are kept as the\n *           AnnexB prefix and prepended to every IDR access unit so\n *           Chrome's HEVC decoder can initialise from a single packet.\n *   - Audio: AAC only. Other codecs are skipped with a warn log.\n *\n * The reader does NOT own the broker's reconnect/backoff loop — it\n * exposes `connect()` / `destroy()` and reports terminal failures via\n * `onError`. The broker re-instantiates a fresh reader on reconnect.\n */\n\nimport type { EncodedPacket, IScopedLogger, PushAudioInfo } from '@camstack/types'\nimport {\n  parseFlvVideoTagWithLengthSize,\n  parseFlvAudioTag,\n  buildH264ParamSetsAnnexB,\n  buildH265ParamSetsAnnexB,\n  nalusToAnnexB,\n  aacSamplingIndexToHz,\n  AudioSoundFormat,\n  UnsupportedCodecError,\n  FlvParseError,\n} from './flv-parser.js'\nimport { RtmpClient, RtmpClientClosedError } from './rtmp-client.js'\n\nexport interface RtmpReaderCallbacks {\n  /** Called when codec is first identified from the FLV sequence header. */\n  onVideoCodec?: (codec: 'h264' | 'h265') => void\n  /** Called for every assembled AnnexB access unit. */\n  onEncodedPacket: (packet: EncodedPacket) => void\n  /** Called once when the AAC sequence header is parsed. */\n  onAudioInfo?: (info: PushAudioInfo) => void\n  /** Called the first time the connection successfully starts streaming media. */\n  onPlaying?: () => void\n  /** Called on any terminal failure (TCP close, parse error, unsupported codec). */\n  onError: (error: Error) => void\n  /** Called once after the read-loop exits (successfully or otherwise). */\n  onTeardown?: () => void\n}\n\nexport class RtmpReader {\n  private readonly url: string\n  private readonly logger: IScopedLogger | undefined\n  private readonly callbacks: RtmpReaderCallbacks\n  private client: RtmpClient | null = null\n  private destroyed = false\n\n  // Per-stream codec state — captured from the first SequenceHeader we see.\n  private videoCodec: 'h264' | 'h265' | null = null\n  private h264ParamPrefix: Buffer | null = null\n  private h265ParamPrefix: Buffer | null = null\n  private naluLengthSize = 4\n\n  private audioInfoEmitted = false\n  private firstMediaPacketSeen = false\n  private packetCounter = 0\n\n  constructor(url: string, logger: IScopedLogger | undefined, callbacks: RtmpReaderCallbacks) {\n    this.url = url\n    this.logger = logger\n    this.callbacks = callbacks\n  }\n\n  async connect(): Promise<void> {\n    const client = new RtmpClient(this.url, this.logger?.child('rtmp-client'))\n    this.client = client\n\n    try {\n      await client.setup()\n    } catch (err) {\n      const error = err instanceof Error ? err : new Error(String(err))\n      this.logger?.warn('rtmp setup failed', { meta: { error: error.message } })\n      this.fail(error)\n      return\n    }\n\n    void this.runReadLoop(client)\n  }\n\n  destroy(): void {\n    if (this.destroyed) return\n    this.destroyed = true\n    this.client?.destroy()\n  }\n\n  private async runReadLoop(client: RtmpClient): Promise<void> {\n    try {\n      for await (const media of client.readLoop()) {\n        if (this.destroyed) break\n\n        if (media.codec === 'video') {\n          this.handleVideoPacket(media.packet, media.timestamp)\n        } else {\n          this.handleAudioPacket(media.packet, media.timestamp)\n        }\n\n        if (!this.firstMediaPacketSeen) {\n          this.firstMediaPacketSeen = true\n          this.callbacks.onPlaying?.()\n        }\n      }\n      // readLoop exited cleanly — the only way this happens is destroy()\n      this.callbacks.onTeardown?.()\n    } catch (err) {\n      if (this.destroyed) {\n        this.callbacks.onTeardown?.()\n        return\n      }\n      const error = err instanceof Error ? err : new Error(String(err))\n      this.fail(error)\n    }\n  }\n\n  private fail(error: Error): void {\n    if (this.destroyed) return\n    this.destroyed = true\n    this.client?.destroy()\n    this.callbacks.onError(error)\n  }\n\n  // ── Video path ────────────────────────────────────────────────────────\n\n  private handleVideoPacket(payload: Buffer, timestamp: number): void {\n    let tag\n    try {\n      tag = parseFlvVideoTagWithLengthSize(payload, this.naluLengthSize)\n    } catch (err) {\n      if (err instanceof UnsupportedCodecError) {\n        this.fail(err)\n        return\n      }\n      if (err instanceof FlvParseError) {\n        this.logger?.warn('rtmp: skipping malformed video tag', { meta: { error: err.message } })\n        return\n      }\n      throw err\n    }\n\n    if (tag.kind === 'video-sequence-header') {\n      this.handleVideoSequenceHeader(tag.codec, tag)\n      return\n    }\n    if (tag.kind === 'video-sequence-end') {\n      // Treated as an end-of-stream signal — let the read loop exit naturally.\n      this.logger?.info('rtmp: video sequence end received')\n      return\n    }\n    // video-coded-frames\n    if (!this.videoCodec) {\n      // Coded frames before sequence header — skip silently. Some servers\n      // start mid-GOP; the next SequenceHeader (or restart) will recover.\n      return\n    }\n\n    const annexB = this.buildAnnexBAccessUnit(tag.codec, tag.nalus, this.isKeyframeFrameType(tag.frameType))\n    if (annexB.length === 0) return\n\n    this.packetCounter++\n    const packet: EncodedPacket = {\n      type: 'video',\n      data: annexB,\n      pts: timestamp + tag.compositionTime,\n      dts: timestamp,\n      keyframe: this.isKeyframeFrameType(tag.frameType),\n      codec: tag.codec,\n    }\n    this.callbacks.onEncodedPacket(packet)\n  }\n\n  private handleVideoSequenceHeader(\n    codec: 'h264' | 'h265',\n    tag: ReturnType<typeof parseFlvVideoTagWithLengthSize>,\n  ): void {\n    if (tag.kind !== 'video-sequence-header') return\n\n    if (this.videoCodec !== null && this.videoCodec !== codec) {\n      this.fail(new Error(`RTMP video codec changed mid-stream: ${this.videoCodec} → ${codec}`))\n      return\n    }\n\n    if (codec === 'h264' && tag.codec === 'h264') {\n      this.videoCodec = 'h264'\n      this.naluLengthSize = tag.avcConfig.lengthSizeMinusOne + 1\n      this.h264ParamPrefix = buildH264ParamSetsAnnexB(tag.avcConfig)\n      this.logger?.info('rtmp: H.264 sequence header', {\n        meta: {\n          profile: tag.avcConfig.avcProfileIndication,\n          level: tag.avcConfig.avcLevelIndication,\n          spsCount: tag.avcConfig.sps.length,\n          ppsCount: tag.avcConfig.pps.length,\n          lengthSize: this.naluLengthSize,\n        },\n      })\n      this.callbacks.onVideoCodec?.('h264')\n      return\n    }\n\n    if (codec === 'h265' && tag.codec === 'h265') {\n      this.videoCodec = 'h265'\n      this.naluLengthSize = tag.hevcConfig.lengthSizeMinusOne + 1\n      this.h265ParamPrefix = buildH265ParamSetsAnnexB(tag.hevcConfig)\n      this.logger?.info('rtmp: H.265 sequence header', {\n        meta: {\n          profile: tag.hevcConfig.generalProfileIdc,\n          level: tag.hevcConfig.generalLevelIdc,\n          vpsCount: tag.hevcConfig.vps.length,\n          spsCount: tag.hevcConfig.sps.length,\n          ppsCount: tag.hevcConfig.pps.length,\n          lengthSize: this.naluLengthSize,\n        },\n      })\n      this.callbacks.onVideoCodec?.('h265')\n    }\n  }\n\n  /**\n   * Build the AnnexB access unit. Prepends VPS/SPS/PPS prefix when the\n   * frame is a keyframe so downstream consumers (decoder, prebuffer,\n   * WebRTC) receive a self-contained access unit on every IDR.\n   */\n  private buildAnnexBAccessUnit(\n    codec: 'h264' | 'h265',\n    nalus: ReadonlyArray<Buffer>,\n    keyframe: boolean,\n  ): Buffer {\n    if (nalus.length === 0) return Buffer.alloc(0)\n    const body = nalusToAnnexB(nalus)\n    if (!keyframe) return body\n\n    const prefix = codec === 'h264' ? this.h264ParamPrefix : this.h265ParamPrefix\n    if (!prefix || prefix.length === 0) return body\n    return Buffer.concat([prefix, body])\n  }\n\n  private isKeyframeFrameType(frameType: number): boolean {\n    // FrameType 1 = KEY, 4 = GENERATED_KEYFRAME\n    return frameType === 1 || frameType === 4\n  }\n\n  // ── Audio path ────────────────────────────────────────────────────────\n\n  private handleAudioPacket(payload: Buffer, timestamp: number): void {\n    let tag\n    try {\n      tag = parseFlvAudioTag(payload)\n    } catch (err) {\n      if (err instanceof FlvParseError) {\n        this.logger?.warn('rtmp: skipping malformed audio tag', { meta: { error: err.message } })\n        return\n      }\n      throw err\n    }\n\n    if (tag.kind === 'audio-sequence-header') {\n      if (tag.soundFormat !== AudioSoundFormat.AAC) {\n        this.logger?.warn('rtmp: non-AAC sequence header — audio skipped', {\n          meta: { soundFormat: tag.soundFormat },\n        })\n        return\n      }\n      const sampleRate = aacSamplingIndexToHz(tag.audioSpecificConfig.samplingFrequencyIndex)\n      const channels = tag.audioSpecificConfig.channelConfiguration === 0\n        ? 1\n        : tag.audioSpecificConfig.channelConfiguration\n      if (sampleRate === null) {\n        this.logger?.warn('rtmp: AAC reserved sample-rate index — audio skipped', {\n          meta: { samplingFrequencyIndex: tag.audioSpecificConfig.samplingFrequencyIndex },\n        })\n        return\n      }\n      this.audioInfoEmitted = true\n      this.callbacks.onAudioInfo?.({\n        codec: 'aac',\n        sampleRate,\n        channels,\n        extraData: new Uint8Array(tag.audioSpecificConfigBytes),\n      })\n      this.logger?.info('rtmp: AAC sequence header', {\n        meta: {\n          audioObjectType: tag.audioSpecificConfig.audioObjectType,\n          sampleRate,\n          channels,\n        },\n      })\n      return\n    }\n\n    if (tag.soundFormat !== AudioSoundFormat.AAC) {\n      // Non-AAC audio (G.711, ADPCM, …) is not supported on the RTMP path.\n      // Drop silently after the first warning to avoid log floods.\n      return\n    }\n\n    if (!this.audioInfoEmitted) {\n      // AAC raw arrived before the sequence header — uncommon but legal.\n      // Skip until we see the header (decoder needs ASC to initialise).\n      return\n    }\n\n    const packet: EncodedPacket = {\n      type: 'audio',\n      data: tag.data,\n      pts: timestamp,\n      dts: timestamp,\n      keyframe: false,\n      codec: 'aac',\n    }\n    this.callbacks.onEncodedPacket(packet)\n  }\n}\n\nexport { RtmpClientClosedError }\n","import {\n  RTP_HEADER_SIZE,\n  H264_NAL_TYPE_MASK,\n  H264_NAL_FUA,\n  H265_NAL_TYPE_SHIFT,\n  H265_NAL_TYPE_MASK,\n  H265_NAL_FU,\n} from './rtsp-types.js'\n\n// ── Types ───────────────────────────────────────────────────────────\n\nexport interface DepacketizedNal {\n  /** Complete NAL unit (without start code). */\n  readonly nal: Buffer\n  /** Whether this NAL is a keyframe (IDR for H.264, IRAP for H.265). */\n  readonly keyframe: boolean\n  /** Whether this NAL is a parameter set (SPS/PPS/VPS). */\n  readonly parameterSet: boolean\n  /** RTP timestamp (90kHz clock). */\n  readonly timestamp: number\n  /** True if the RTP marker bit was set (last packet of access unit). */\n  readonly marker: boolean\n}\n\nexport type NalCallback = (nal: DepacketizedNal) => void\n\n// ── H.264 NAL types ────────────────────────────────────────────────\n\nconst H264_NAL_SPS = 7\nconst H264_NAL_PPS = 8\nconst H264_NAL_IDR = 5\nconst H264_NAL_STAP_A = 24\n\n// ── H.265 NAL types ────────────────────────────────────────────────\n\nconst H265_NAL_VPS = 32\nconst H265_NAL_SPS = 33\nconst H265_NAL_PPS = 34\nconst H265_NAL_IRAP_MIN = 16\nconst H265_NAL_IRAP_MAX = 21\nconst H265_NAL_AP = 48\n\n// ── Implementation ──────────────────────────────────────────────────\n\n/**\n * RTP → Annex-B NAL depacketizer.\n *\n * Takes raw RTP packets (as received from TCP interleaved RTSP) and\n * reassembles complete NAL units. Handles:\n *\n * H.264 (RFC 6184):\n *   - Single NAL unit packets (type 1-23)\n *   - STAP-A aggregation packets (type 24)\n *   - FU-A fragmentation packets (type 28)\n *\n * H.265 (RFC 7798):\n *   - Single NAL unit packets (type 0-47, except 48/49)\n *   - AP aggregation packets (type 48)\n *   - FU fragmentation packets (type 49)\n */\nexport class RtpDepacketizer {\n  private readonly codec: 'h264' | 'h265'\n  private readonly onNal: NalCallback\n\n  /** FU reassembly state. */\n  private fuBuffer: Buffer[] = []\n  private fuTimestamp = 0\n  /** H.264: stores the reconstructed first byte (NRI + type). */\n  private fuFirstByte = 0\n  /** H.265: stores the 2-byte NAL header from the FU payload header. */\n  private fuH265Header: Buffer | null = null\n\n  constructor(codec: 'h264' | 'h265', onNal: NalCallback) {\n    this.codec = codec\n    this.onNal = onNal\n  }\n\n  /** Reset reassembly state (e.g. on stream restart). */\n  reset(): void {\n    this.fuBuffer = []\n    this.fuTimestamp = 0\n\n    this.fuFirstByte = 0\n    this.fuH265Header = null\n  }\n\n  /**\n   * Process a raw RTP packet (including 12-byte RTP header).\n   * Emits zero or more complete NAL units via the callback.\n   */\n  processPacket(rtpData: Buffer): void {\n    if (rtpData.length < RTP_HEADER_SIZE) return\n\n    const marker = (rtpData[1]! & 0x80) !== 0\n    const timestamp = rtpData.readUInt32BE(4)\n    const payload = rtpData.subarray(RTP_HEADER_SIZE)\n\n    if (payload.length === 0) return\n\n    if (this.codec === 'h264') {\n      this.processH264(payload, timestamp, marker)\n    } else {\n      this.processH265(payload, timestamp, marker)\n    }\n  }\n\n  // ── H.264 ─────────────────────────────────────────────────────────\n\n  private processH264(payload: Buffer, timestamp: number, marker: boolean): void {\n    const firstByte = payload[0]!\n    const nalType = firstByte & H264_NAL_TYPE_MASK\n\n    if (nalType >= 1 && nalType <= 23) {\n      // Single NAL unit packet\n      this.emitNal(payload, timestamp, marker)\n    } else if (nalType === H264_NAL_STAP_A) {\n      this.processH264StapA(payload, timestamp, marker)\n    } else if (nalType === H264_NAL_FUA) {\n      this.processH264FuA(payload, timestamp, marker)\n    }\n    // Types 25-27 (STAP-B, MTAP16, MTAP24) and 29 (FU-B) are extremely rare\n    // in IP camera streams — ignore them.\n  }\n\n  /** STAP-A: Single-Time Aggregation Packet (multiple NALs in one RTP). */\n  private processH264StapA(payload: Buffer, timestamp: number, marker: boolean): void {\n    let offset = 1 // Skip STAP-A header byte\n    while (offset + 2 <= payload.length) {\n      const nalSize = payload.readUInt16BE(offset)\n      offset += 2\n      if (offset + nalSize > payload.length) break\n\n      const nal = Buffer.from(payload.subarray(offset, offset + nalSize))\n      offset += nalSize\n      this.emitNal(nal, timestamp, marker && offset >= payload.length)\n    }\n  }\n\n  /** FU-A: Fragmentation Unit (large NAL split across multiple RTP packets). */\n  private processH264FuA(payload: Buffer, timestamp: number, marker: boolean): void {\n    if (payload.length < 2) return\n\n    const fuIndicator = payload[0]!\n    const fuHeader = payload[1]!\n    const isStart = (fuHeader & 0x80) !== 0\n    const isEnd = (fuHeader & 0x40) !== 0\n    const nalType = fuHeader & H264_NAL_TYPE_MASK\n\n    if (isStart) {\n      // Start fragment — reconstruct NAL first byte: NRI from indicator + type from header\n      this.fuBuffer = [payload.subarray(2)]\n      this.fuTimestamp = timestamp\n\n      this.fuFirstByte = (fuIndicator & 0xe0) | nalType\n    } else if (this.fuBuffer.length > 0 && this.fuTimestamp === timestamp) {\n      // Middle or end fragment\n      this.fuBuffer.push(payload.subarray(2))\n\n      if (isEnd) {\n        // Reassemble: prepend reconstructed NAL header byte\n        const totalSize = this.fuBuffer.reduce((sum, b) => sum + b.length, 0)\n        const nal = Buffer.allocUnsafe(1 + totalSize)\n        nal[0] = this.fuFirstByte\n        let offset = 1\n        for (const fragment of this.fuBuffer) {\n          fragment.copy(nal, offset)\n          offset += fragment.length\n        }\n        this.fuBuffer = []\n        this.emitNal(nal, timestamp, marker)\n      }\n    } else {\n      // Out-of-order or gap — discard partial reassembly\n      this.fuBuffer = []\n    }\n  }\n\n  // ── H.265 ─────────────────────────────────────────────────────────\n\n  private processH265(payload: Buffer, timestamp: number, marker: boolean): void {\n    if (payload.length < 2) return\n\n    const nalType = (payload[0]! >> H265_NAL_TYPE_SHIFT) & H265_NAL_TYPE_MASK\n\n    if (nalType < 48) {\n      // Single NAL unit packet (types 0-47)\n      this.emitNal(payload, timestamp, marker)\n    } else if (nalType === H265_NAL_AP) {\n      this.processH265Ap(payload, timestamp, marker)\n    } else if (nalType === H265_NAL_FU) {\n      this.processH265Fu(payload, timestamp, marker)\n    }\n  }\n\n  /** AP: Aggregation Packet (multiple NALs in one RTP). */\n  private processH265Ap(payload: Buffer, timestamp: number, marker: boolean): void {\n    let offset = 2 // Skip 2-byte AP payload header\n    while (offset + 2 <= payload.length) {\n      const nalSize = payload.readUInt16BE(offset)\n      offset += 2\n      if (offset + nalSize > payload.length) break\n\n      const nal = Buffer.from(payload.subarray(offset, offset + nalSize))\n      offset += nalSize\n      this.emitNal(nal, timestamp, marker && offset >= payload.length)\n    }\n  }\n\n  /** FU: Fragmentation Unit (large NAL split across multiple RTP packets). */\n  private processH265Fu(payload: Buffer, timestamp: number, marker: boolean): void {\n    if (payload.length < 3) return\n\n    const fuHeader = payload[2]!\n    const isStart = (fuHeader & 0x80) !== 0\n    const isEnd = (fuHeader & 0x40) !== 0\n    const nalType = fuHeader & H265_NAL_TYPE_MASK\n\n    if (isStart) {\n      // Start fragment — store payload header bytes for NAL header reconstruction\n      this.fuBuffer = [payload.subarray(3)]\n      this.fuTimestamp = timestamp\n\n      // Reconstruct 2-byte NAL header: replace FU type with actual NAL type\n      const header = Buffer.allocUnsafe(2)\n      header[0] = (payload[0]! & 0x81) | (nalType << H265_NAL_TYPE_SHIFT)\n      header[1] = payload[1]!\n      this.fuH265Header = header\n    } else if (this.fuBuffer.length > 0 && this.fuTimestamp === timestamp) {\n      // Middle or end fragment\n      this.fuBuffer.push(payload.subarray(3))\n\n      if (isEnd && this.fuH265Header) {\n        // Reassemble: prepend 2-byte NAL header\n        const totalSize = this.fuBuffer.reduce((sum, b) => sum + b.length, 0)\n        const nal = Buffer.allocUnsafe(2 + totalSize)\n        this.fuH265Header.copy(nal, 0)\n        let offset = 2\n        for (const fragment of this.fuBuffer) {\n          fragment.copy(nal, offset)\n          offset += fragment.length\n        }\n        this.fuBuffer = []\n        this.fuH265Header = null\n        this.emitNal(nal, timestamp, marker)\n      }\n    } else {\n      // Gap — discard\n      this.fuBuffer = []\n      this.fuH265Header = null\n    }\n  }\n\n  // ── Emit ──────────────────────────────────────────────────────────\n\n  private emitNal(nal: Buffer, timestamp: number, marker: boolean): void {\n    const keyframe = this.codec === 'h264'\n      ? this.isH264Keyframe(nal)\n      : this.isH265Keyframe(nal)\n\n    const parameterSet = this.codec === 'h264'\n      ? this.isH264ParameterSet(nal)\n      : this.isH265ParameterSet(nal)\n\n    this.onNal({ nal, keyframe, parameterSet, timestamp, marker })\n  }\n\n  private isH264Keyframe(nal: Buffer): boolean {\n    if (nal.length === 0) return false\n    return (nal[0]! & H264_NAL_TYPE_MASK) === H264_NAL_IDR\n  }\n\n  private isH264ParameterSet(nal: Buffer): boolean {\n    if (nal.length === 0) return false\n    const type = nal[0]! & H264_NAL_TYPE_MASK\n    return type === H264_NAL_SPS || type === H264_NAL_PPS\n  }\n\n  private isH265Keyframe(nal: Buffer): boolean {\n    if (nal.length < 2) return false\n    const type = (nal[0]! >> H265_NAL_TYPE_SHIFT) & H265_NAL_TYPE_MASK\n    return type >= H265_NAL_IRAP_MIN && type <= H265_NAL_IRAP_MAX\n  }\n\n  private isH265ParameterSet(nal: Buffer): boolean {\n    if (nal.length < 2) return false\n    const type = (nal[0]! >> H265_NAL_TYPE_SHIFT) & H265_NAL_TYPE_MASK\n    return type === H265_NAL_VPS || type === H265_NAL_SPS || type === H265_NAL_PPS\n  }\n}\n","/**\n * Native audio RTP decoder for G.711 PCMU (μ-law) and PCMA (A-law).\n *\n * IP cameras almost universally use G.711 for audio (8kHz, mono).\n * This decoder converts raw RTP payloads → PCM float32 samples,\n * matching the format of the ffmpeg audio path (f32le mono 16kHz)\n * so downstream consumers (audio-classifier, VAD) receive identical data.\n *\n * Output: 16kHz mono f32le — upsampled from 8kHz via linear interpolation.\n */\n\nimport type { DecodedAudioChunk } from '@camstack/types'\n\nexport type AudioCodec = 'PCMU' | 'PCMA'\n\n/** Target sample rate for decoded audio (matches ffmpeg path). */\nconst TARGET_SAMPLE_RATE = 16000\n/** Buffer ~0.5s of decoded audio before emitting a chunk (matches ffmpeg). */\nconst CHUNK_DURATION_MS = 500\nconst CHUNK_SAMPLES = (TARGET_SAMPLE_RATE * CHUNK_DURATION_MS) / 1000 // 8000 samples\n\n/**\n * Decodes G.711 audio RTP packets into PCM f32le chunks.\n *\n * Buffers decoded samples and emits DecodedAudioChunk when enough\n * samples accumulate (~0.5s), matching the ffmpeg audio output cadence.\n */\nexport class AudioRtpDecoder {\n  private readonly codec: AudioCodec\n  private readonly onChunk: (chunk: DecodedAudioChunk) => void\n  private buffer: Float32Array\n  private bufferOffset = 0\n\n  constructor(codec: AudioCodec, onChunk: (chunk: DecodedAudioChunk) => void) {\n    this.codec = codec\n    this.onChunk = onChunk\n    this.buffer = new Float32Array(CHUNK_SAMPLES)\n  }\n\n  /**\n   * Process a raw RTP packet (header + payload).\n   * Strips the 12-byte RTP header, decodes G.711 payload, upsamples 8→16kHz.\n   */\n  processPacket(rtpData: Buffer): void {\n    // RTP header is 12 bytes minimum (no CSRC, no extensions handled here)\n    if (rtpData.length <= 12) return\n\n    const payload = rtpData.subarray(12)\n    const decode = this.codec === 'PCMU' ? decodeUlaw : decodeAlaw\n\n    // Each byte = one G.711 sample at 8kHz.\n    // Upsample 8→16kHz via linear interpolation (2x).\n    for (let i = 0; i < payload.length; i++) {\n      const current = decode(payload[i]!)\n      const next = i + 1 < payload.length ? decode(payload[i + 1]!) : current\n\n      // Original sample\n      this.pushSample(current)\n      // Interpolated sample (midpoint)\n      this.pushSample((current + next) * 0.5)\n    }\n  }\n\n  /** Flush any remaining buffered samples as a partial chunk. */\n  flush(): void {\n    if (this.bufferOffset === 0) return\n\n    const data = Buffer.from(\n      this.buffer.buffer,\n      this.buffer.byteOffset,\n      this.bufferOffset * 4,\n    )\n\n    this.onChunk({\n      data: Buffer.from(data), // copy — buffer will be reused\n      sampleRate: TARGET_SAMPLE_RATE,\n      channels: 1,\n      timestamp: Date.now(),\n    })\n\n    this.bufferOffset = 0\n  }\n\n  /** Reset the decoder state. */\n  reset(): void {\n    this.bufferOffset = 0\n  }\n\n  private pushSample(sample: number): void {\n    this.buffer[this.bufferOffset++] = sample\n\n    if (this.bufferOffset >= CHUNK_SAMPLES) {\n      // Emit full chunk\n      const data = Buffer.from(\n        this.buffer.buffer,\n        this.buffer.byteOffset,\n        CHUNK_SAMPLES * 4,\n      )\n\n      this.onChunk({\n        data: Buffer.from(data), // copy\n        sampleRate: TARGET_SAMPLE_RATE,\n        channels: 1,\n        timestamp: Date.now(),\n      })\n\n      this.bufferOffset = 0\n    }\n  }\n}\n\n// ── G.711 μ-law decoding ──────────────────────────────────────────────\n\n/** μ-law compressed byte → normalized float [-1.0, 1.0]. */\nfunction decodeUlaw(byte: number): number {\n  return ULAW_TABLE[byte]!\n}\n\n/** A-law compressed byte → normalized float [-1.0, 1.0]. */\nfunction decodeAlaw(byte: number): number {\n  return ALAW_TABLE[byte]!\n}\n\n/**\n * Build the μ-law decode table (ITU-T G.711).\n * Each of the 256 possible byte values maps to a 16-bit PCM sample,\n * which we normalize to [-1.0, 1.0] for f32le output.\n */\nfunction buildUlawTable(): Float32Array {\n  const table = new Float32Array(256)\n  for (let i = 0; i < 256; i++) {\n    const complemented = ~i & 0xff\n    const sign = (complemented & 0x80) !== 0 ? -1 : 1\n    const exponent = (complemented >> 4) & 0x07\n    const mantissa = complemented & 0x0f\n    const magnitude = ((2 * mantissa + 33) << exponent) - 33\n    table[i] = (sign * magnitude) / 32768\n  }\n  return table\n}\n\n/**\n * Build the A-law decode table (ITU-T G.711).\n */\nfunction buildAlawTable(): Float32Array {\n  const table = new Float32Array(256)\n  for (let i = 0; i < 256; i++) {\n    const xored = i ^ 0x55\n    const sign = (xored & 0x80) !== 0 ? -1 : 1\n    const exponent = (xored >> 4) & 0x07\n    const mantissa = xored & 0x0f\n    let magnitude: number\n    if (exponent === 0) {\n      magnitude = (2 * mantissa + 1)\n    } else {\n      magnitude = (2 * mantissa + 33) << (exponent - 1)\n    }\n    table[i] = (sign * magnitude) / 32768\n  }\n  return table\n}\n\nconst ULAW_TABLE = buildUlawTable()\nconst ALAW_TABLE = buildAlawTable()\n","/**\n * G.711 µ-law encoder — converts 16-bit signed PCM samples to µ-law (ITU-T G.711).\n *\n * Used to bridge non-native audio codecs (AAC, Opus) into WebRTC's\n * PCMU negotiated track. The audio-codec cap decodes to linear PCM;\n * this module encodes the PCM to µ-law for the RTP payload.\n */\n\n// µ-law compression bias\nconst MULAW_BIAS = 0x84\nconst MULAW_CLIP = 32635\n\n/** Encode a single 16-bit signed PCM sample to µ-law byte. */\nfunction linearToMulaw(sample: number): number {\n  const sign = (sample >> 8) & 0x80\n  if (sign !== 0) sample = -sample\n  if (sample > MULAW_CLIP) sample = MULAW_CLIP\n  sample = sample + MULAW_BIAS\n\n  const exponent = MU_LAW_COMPRESS_TABLE[(sample >> 7) & 0xff]!\n  const mantissa = (sample >> (exponent + 3)) & 0x0f\n  const byte = ~(sign | (exponent << 4) | mantissa) & 0xff\n  return byte\n}\n\n// Lookup table: maps (sample >> 7) to exponent (0-7)\nconst MU_LAW_COMPRESS_TABLE = new Uint8Array(256)\n;(() => {\n  for (let i = 0; i < 256; i++) {\n    let val = i\n    let exp = 0\n    val >>= 1\n    while (val > 0 && exp < 7) { val >>= 1; exp++ }\n    MU_LAW_COMPRESS_TABLE[i] = exp\n  }\n})()\n\n/** Convert an Int16Array of PCM samples to a Uint8Array of µ-law bytes. */\nexport function pcmToMulaw(pcm: Int16Array): Uint8Array {\n  const result = new Uint8Array(pcm.length)\n  for (let i = 0; i < pcm.length; i++) {\n    result[i] = linearToMulaw(pcm[i]!)\n  }\n  return result\n}\n\n/**\n * Downsample µ-law by picking every Nth sample. Used when source\n * sample rate exceeds 8kHz (e.g. AAC at 16kHz → G.711 at 8kHz).\n * Simple decimation is acceptable for speech/ambient audio.\n */\nexport function downsampleUlaw(data: Uint8Array, ratio: number): Uint8Array {\n  const outLen = Math.floor(data.length / ratio)\n  const result = new Uint8Array(outLen)\n  for (let i = 0; i < outLen; i++) {\n    result[i] = data[i * ratio]!\n  }\n  return result\n}\n","/**\n * Pre-encoded Annex-B placeholder keyframes for both H.264 and H.265.\n * Single 1280×720 black frame with a centred text label\n * (\"RECONNECTING\", \"SLEEPING\", …), encoded with x264/x265 ultrafast.\n * Embedded as base64 to avoid spawning ffmpeg at runtime.\n *\n * Frames carry SPS+PPS+IDR (H.264) or VPS+SPS+PPS+IDR (H.265) in a\n * single Annex-B blob so a fresh consumer (RTSP restream, WebRTC\n * AnnexB feed) decodes them without waiting for the next \"real\"\n * keyframe.\n *\n * Codec selection happens at the broker level: a broker carrying an\n * H.265 source emits H.265 placeholders so a WebRTC session\n * negotiated for H.265 sees correctly-coded fill while waking up.\n *\n * To regenerate (regression/refresh after font change):\n *   for label in reconnecting sleeping offline disabled waking; do\n *     upper=$(echo $label | tr '[:lower:]' '[:upper:]')\n *     # H.264\n *     ffmpeg -hide_banner -loglevel error -y -f lavfi \\\n *       -i \"color=c=0x101010:s=1280x720:r=1:d=1\" \\\n *       -vf \"drawtext=fontfile=/System/Library/Fonts/Supplemental/Arial.ttf:text='${upper}':fontsize=64:fontcolor=0xb8b8b8:x=(w-text_w)/2:y=(h-text_h)/2\" \\\n *       -frames:v 1 -c:v libx264 -preset ultrafast -tune stillimage \\\n *       -profile:v baseline -x264-params \"nal-hrd=none:aud=0:repeat-headers=1\" \\\n *       -f h264 -bsf:v \"h264_mp4toannexb,filter_units=remove_types=6\" \\\n *       ${label}.h264\n *     # H.265\n *     ffmpeg -hide_banner -loglevel error -y -f lavfi \\\n *       -i \"color=c=0x101010:s=1280x720:r=1:d=1\" \\\n *       -vf \"drawtext=fontfile=/System/Library/Fonts/Supplemental/Arial.ttf:text='${upper}':fontsize=64:fontcolor=0xb8b8b8:x=(w-text_w)/2:y=(h-text_h)/2\" \\\n *       -frames:v 1 -c:v libx265 -preset ultrafast \\\n *       -x265-params \"keyint=1:repeat-headers=1\" \\\n *       -f hevc ${label}.h265\n *   done\n */\n\n/**\n * State the broker is currently in. Drives which placeholder frame is\n * emitted to encoded subscribers (RTSP restream, WebRTC AnnexB feed)\n * while no live RTP is flowing. Default `reconnecting` is the\n * legacy behaviour for the dial-fail/reconnect path.\n */\nexport type PlaceholderKind =\n  | 'reconnecting'\n  | 'sleeping'\n  | 'offline'\n  | 'disabled'\n  | 'waking'\n\n/**\n * Codec the placeholder frame is encoded as. Selection lives at the\n * broker level — an H.265 source surfaces H.265 placeholders so a\n * negotiated WebRTC session decodes them with the same codec as the\n * live stream. Only `'h264'` and `'h265'` are supported (the only\n * codecs that flow through the broker's encoded subscribers today).\n */\nexport type PlaceholderCodec = 'h264' | 'h265'\n\nconst FRAME_B64: Readonly<Record<PlaceholderCodec, Readonly<Record<PlaceholderKind, string>>>> = {\n  h264: {\n    reconnecting: '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   sleeping: 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',\n    sleeping: 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   waking: 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',\n  },\n}\n\nconst cache: Partial<Record<PlaceholderCodec, Partial<Record<PlaceholderKind, Buffer>>>> = {}\n\n/**\n * Get the pre-encoded placeholder frame for `(kind, codec)`. Buffer\n * is created once on first use per (codec, kind) pair and cached\n * permanently — safe to call from hot paths (~5-9KB per pair, ~60KB\n * worst-case across both codecs and all five states).\n */\nexport function getPlaceholderFrame(kind: PlaceholderKind, codec: PlaceholderCodec): Buffer {\n  const codecCache = cache[codec] ?? (cache[codec] = {})\n  const cached = codecCache[kind]\n  if (cached) return cached\n  const buf = Buffer.from(FRAME_B64[codec][kind], 'base64')\n  codecCache[kind] = buf\n  return buf\n}\n\n/**\n * Backwards-compatible alias for the legacy single-frame API. Returns\n * the H.264 `reconnecting` placeholder (the original \"black screen\n * during reconnect\" frame).\n *\n * @deprecated prefer `getPlaceholderFrame(kind, codec)` so the\n * consumer surfaces both the actual broker state and the matching\n * codec to the operator.\n */\nexport function getBlackPlaceholderFrame(): Buffer {\n  return getPlaceholderFrame('reconnecting', 'h264')\n}\n","import type {\n  IStreamBroker,\n  StreamSource,\n  StreamSourceType,\n  EncodedPacket,\n  DecodedAudioChunk,\n  BrokerStatus,\n  BrokerStats,\n  Unsubscribe,\n  IRestreamer,\n  IScopedLogger,\n  FrameHandle,\n  SubscribeFramesInput,\n  SubscribeFramesResult,\n  BrokerConsumerAttribution,\n  BrokerEncodedClient,\n} from '@camstack/types'\n\ninterface EncodedSubscriberEntry {\n  readonly id: string\n  readonly channel: 'annexb' | 'rtp'\n  attribution: BrokerConsumerAttribution\n  readonly subscribedAt: number\n  packetsDelivered: number\n}\n\nconst UNKNOWN_ATTRIBUTION: BrokerConsumerAttribution = { kind: 'unknown' }\n\nimport type {\n  SubscribeAudioChunksResult,\n} from '@camstack/types'\n\nimport type { DecoderCapApi } from './decoder-session-proxy.js'\nimport { FrameHandlePlane, type FrameHandlePlaneStreamInfo } from './frame-handle-plane.js'\nimport { AudioChunkPlane } from './audio-chunk-plane.js'\nimport type { AudioCodecCapApi } from './audio-codec-proxy.js'\nimport { AudioCodecSession } from './audio-codec-session.js'\nimport { parseAudioSpecificConfig } from '../rtsp/audio/aac-config.js'\nimport { StreamPipeServer } from './stream-pipe-server'\nimport { EncodedRingBuffer } from './encoded-ring-buffer'\nimport { RtspRestreamer } from '../rtsp/rtsp-restreamer.js'\nimport { NativeRtspClient } from '../rtsp/rtsp-client.js'\nimport { Rfc4571Reader } from '../rtsp/rfc4571-reader.js'\nimport { RtmpReader } from '../rtmp/rtmp-reader.js'\nimport { RtpDepacketizer } from '../rtsp/rtp-depacketizer.js'\nimport type { DepacketizedNal } from '../rtsp/rtp-depacketizer.js'\nimport { AudioRtpDecoder } from '../rtsp/audio-rtp-decoder.js'\nimport type { AudioCodec } from '../rtsp/audio-rtp-decoder.js'\nimport { pcmToMulaw, downsampleUlaw } from './g711-mulaw.js'\nimport { getPlaceholderFrame, type PlaceholderKind, type PlaceholderCodec } from './placeholder-frames.js'\nimport { errMsg, maskUrlCredentials } from '@camstack/types'\n\nconst INITIAL_RECONNECT_DELAY_MS = 1_000\nconst MAX_RECONNECT_DELAY_MS = 30_000\n/** Grace period before suspending after last consumer leaves. */\nconst SUSPEND_GRACE_MS = 5_000\n/** Timeout for push-mode sources: if no packet arrives within this window, emit error placeholder. */\nconst PUSH_STALL_TIMEOUT_MS = 15_000\n\n\nexport class StreamBroker implements IStreamBroker {\n  readonly deviceId: string\n\n  private _status: BrokerStatus = 'idle'\n  private source: StreamSource | undefined\n  /**\n   * Manager-supplied resolver that derives the current `StreamSource`\n   * from the live cam-stream registry. When set, every dial attempt\n   * goes through it so a re-publish (e.g. Reolink updating an\n   * `rfc4571` URL after the lib's TCP server idle-tore-down and\n   * rebound) is observed transparently — the closure re-resolves a\n   * fresh URL on every dial. `null` for tests and stand-alone callers;\n   * falls back to the source passed to `start()`.\n   */\n  private sourceProvider: (() => StreamSource | null) | null = null\n  /**\n   * Manager-supplied notifier that asks the publishing addon to\n   * refresh its loopback transport (e.g. re-run\n   * `ensureRfc4571Server`). Called when a managed-loopback dial fails\n   * — the publisher republishes with a fresh URL and the next\n   * reconnect resolves it via `sourceProvider`. `null` when the\n   * broker has no manager attached (tests).\n   */\n  private requestSourceRefresh: (() => void) | null = null\n  /**\n   * Last time we asked the publisher to refresh the source — used to\n   * dedup the refresh notify. A single connect failure fans out into\n   * two events (`socket.on('error')` → onError callback AND the\n   * rejected `reader.connect()` promise → .catch); without dedup the\n   * publisher's `publishToBroker` runs twice in the same tick.\n   * Idempotent on the publisher side (the lib's shared pool catches\n   * the second one as `awaiting in-flight create`), but the noise\n   * isn't useful. 500 ms is wider than the worst-case fan-out delay.\n   */\n  private lastRefreshRequestAt = 0\n  /** Detected or configured codec — persists across reconnects */\n  private detectedCodec: string | undefined\n  /**\n   * The shared-memory frame-handle surface (Phase 5 / D9) — the broker's sole\n   * decoded-frame path. Lazily created on the first `subscribeFrameHandles`\n   * call; owns the per-format `frameSink: 'shm'` decoder sessions a handle\n   * consumer reads from. `null` until a consumer subscribes.\n   */\n  private frameHandlePlane: FrameHandlePlane | null = null\n  private readonly decoderApi: DecoderCapApi | null\n  private readonly audioCodecApi: AudioCodecCapApi | null\n  private readonly logger: IScopedLogger | undefined\n  private readonly startedAt: number = Date.now()\n\n  private nativeClient: NativeRtspClient | null = null\n  private rfc4571Reader: Rfc4571Reader | null = null\n  private rtmpReader: RtmpReader | null = null\n  private rtpDepacketizer: RtpDepacketizer | null = null\n  /** Flag-only depacketizer for the `push-rtp` source-RTP pre-buffer path\n   *  (sets keyframe/param-set flags without re-emitting AnnexB packets). */\n  private pushRtpFlagDepacketizer: RtpDepacketizer | null = null\n  private audioRtpDecoder: AudioRtpDecoder | null = null\n  private audioCodecSession: AudioCodecSession | null = null\n  private audioRtpSeen = 0\n  /** Detected audio codec from RTSP DESCRIBE (e.g. 'PCMU', 'PCMA'). */\n  private audioCodec: string | null = null\n  /**\n   * Full audio track snapshot from the SDP — surfaced in `BrokerStats.audio`\n   * so the UI overview / audio-analyzer model picker can pre-select the\n   * right model based on per-camera codec + sample rate.\n   */\n  private audioTrackInfo: { codec: string; sampleRate: number; channels: number; supported: boolean } | null = null\n  private reconnectTimer: ReturnType<typeof setTimeout> | undefined\n  private placeholderTimer: ReturnType<typeof setInterval> | undefined\n  private reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n  /**\n   * \"First video\" watchdog. Battery-cam rfc4571 sources sometimes\n   * complete TCP handshake on the loopback but the lib's dedicated\n   * session never delivers video bytes (camera fails to start the\n   * upstream Baichuan stream after wake, or stream subscription gets\n   * stuck). The user's manual workaround was to click restart on the\n   * WebRTC viewer — that closed the broker, tore the lib's session\n   * down, and on reopen the dedicated session attached cleanly.\n   *\n   * This watchdog automates that: it arms when the rfc4571 reader's\n   * `onVideoTrack` fires (TCP up, SDP parsed) and cancels at the first\n   * `onVideoRtp` callback. If it elapses without a single video RTP\n   * packet, it destroys the reader, asks the publisher for a fresh\n   * source, and re-dials — exactly what the manual restart did.\n   */\n  private firstVideoWatchdog: ReturnType<typeof setTimeout> | undefined\n  /**\n   * Wall-clock timeout for the first-video watchdog. Long enough that\n   * a healthy live cam comfortably emits its first IDR within the\n   * window (typical: ~1-2s after rfc4571 TCP connect on PoE cams,\n   * ~2-5s on battery cams that need to align stream subscription with\n   * the video pipeline post-wake), but short enough that a stuck\n   * dedicated session is detected before the WebRTC client gives up\n   * on the offer (Chrome's default ICE-disconnect timeout is ~15s).\n   */\n  private static readonly FIRST_VIDEO_TIMEOUT_MS = 8_000\n\n  /**\n   * Rolling source-RTP activity watchdog. Distinct from the first-video\n   * watchdog: this one ARMS on the first RTP packet (video or audio) and\n   * RESETS on every subsequent packet. If no packet arrives within\n   * `RTP_ACTIVITY_TIMEOUT_MS`, the reader is torn down and reconnected.\n   *\n   * Wi-Fi-attached IP cameras silently stall the RTSP/TCP source under\n   * packet loss: ICE/DTLS stay healthy, `rtpPacketsSent` keeps advancing\n   * (replaying the GOP cache), but the source-side audio/video time\n   * progresses only in tiny bursts. The user sees a frozen viewport with\n   * no recovery because the existing code path never noticed the stall.\n   *\n   * The watchdog resets on every RTP packet and kills+restarts the reader\n   * when no data arrives within `RTP_ACTIVITY_TIMEOUT_MS` (~10s).\n   *\n   * Reused across the native RTSP and RFC 4571 readers — both flow through\n   * `buildRtpStreamCallbacks` and call `kickRtpActivityWatchdog()` per\n   * packet. push-rtp / push (provider-driven) and RTMP sources sit out\n   * because they have no native reader to tear down.\n   */\n  private rtpActivityWatchdog: ReturnType<typeof setTimeout> | undefined\n  /** Number of times the activity watchdog fired in this broker's lifetime\n   *  (incremented on expiry, NOT on arm/cancel). Surfaced via the test\n   *  seam so unit tests can assert recovery behavior without inspecting\n   *  private timer handles. */\n  private rtpActivityWatchdogFiredCount = 0\n  /** Activity watchdog timeout. Matches `FIRST_VIDEO_TIMEOUT_MS` so a\n   *  healthy live source comfortably stays ahead (PCMU audio is 50pps,\n   *  H.264 video is 30fps → 80pps minimum on a working stream) while a\n   *  real stall is detected before Chrome's ~15s ICE-disconnect timer\n   *  has any chance to fire. */\n  private static readonly RTP_ACTIVITY_TIMEOUT_MS = 8_000\n  private manualStop = false\n  private stopping = false\n\n  private restreamers: readonly IRestreamer[] = []\n  private readonly pipeServer: StreamPipeServer\n  private readonly rtspRestreamer: import('../rtsp/rtsp-restreamer.js').RtspRestreamer\n  private readonly preBuffer: EncodedRingBuffer\n\n  private readonly encodedCallbacks = new Map<\n    (packet: EncodedPacket) => void,\n    EncodedSubscriberEntry\n  >()\n  /**\n   * Source-RTP video subscribers — receive raw on-wire RTP bytes\n   * (no depacketization). Used by the WebRTC server's H.265 path to\n   * feed a packet-level repacketizer. See `onVideoRtp`.\n   */\n  private readonly rtpVideoCallbacks = new Map<\n    (rtpData: Buffer) => void,\n    EncodedSubscriberEntry\n  >()\n  private subscriberIdCounter = 0\n  /**\n   * Codec parameter sets harvested from the camera's SDP (a=fmtp\n   * sprop-vps / sprop-sps / sprop-pps). Cameras that ONLY put their\n   * VPS/SPS/PPS in the SDP (Reolink high-profile streams, IP-cam\n   * vendors that strip param sets from the wire) leave WebRTC sessions\n   * without an HEVC decoder configuration — Chrome receives every IDR\n   * but VideoToolbox never initialises.\n   *\n   * We capture them in `onVideoTrack` and replay them as a synthetic\n   * EncodedPacket the moment a new encoded-data subscriber registers\n   * AND on every keyframe whose NAL stream is missing them. This way\n   * `session.ts` populates `lastVps/Sps/Pps` regardless of how the\n   * camera ships its parameter sets.\n   */\n  private sdpParameterSets: ReadonlyArray<Buffer> | null = null\n  /**\n   * Per-device streaming debug flag. Gates verbose info-level logs\n   * around RTP/encoded subscriber lifecycle, SDP param set replay,\n   * and codec parameter handling. Toggled via the device override\n   * `streamingDebug` field. Off by default — production logs stay\n   * quiet unless an operator explicitly enables debug for one device.\n   */\n  private streamingDebug = false\n  /**\n   * Decoded audio-chunk plane (Phase 5 / D9) — the poll-based, tRPC-reachable\n   * replacement for the live-object `onDecodedAudioChunk` callback path.\n   * Holds the per-subscription FIFO chunk queues. Assigned in the constructor\n   * so it gets a scoped logger child.\n   */\n  private readonly audioChunkPlane: AudioChunkPlane\n  /**\n   * Gate for feeding encoded packets to the decoder: H.264/H.265 decoders\n   * need the SPS/PPS carried in the first keyframe to initialise, so the\n   * broker withholds packets from the shm frame plane until the first\n   * keyframe arrives.\n   */\n  private firstKeyframeReceived = false\n\n  /** Timer for push-mode stall detection. */\n  private pushStallTimer: ReturnType<typeof setTimeout> | undefined\n  /** Whether pre-buffer counts as demand (keeps the stream alive even with 0 subscribers). */\n  private _preBufferEnabled = true\n  /** Timer for delayed suspend after last consumer leaves. */\n  private suspendTimer: ReturnType<typeof setTimeout> | undefined\n  /** True while the stream is suspended (ffmpeg killed, no RTSP connection). */\n  private _suspended = false\n\n  /** Counter for video RTP packets — drives the milestone diagnostic\n   *  log alongside `audioRtpSeen`, so a wake-up cycle that brings audio\n   *  back but never video is observable from the structured logs. */\n  private videoRtpSeen = 0\n\n  /** Tracking flags set synchronously by the RTP depacketizer callback. */\n  private _lastNalKeyframe = false\n  private _lastNalParamSet = false\n\n  /**\n   * Source-RTP pre-buffer: a ring of raw source RTP packets trimmed to\n   * begin at the most recent keyframe access unit — i.e. the current GOP so\n   * far. A late-joining WebRTC viewer on the repacketizer path replays this\n   * on connect (`getRtpPreBuffer`) so its decoder initialises immediately\n   * from a keyframe instead of waiting for the camera's next IDR (which can\n   * be many seconds away on a long-GOP 4K stream). This is the RTP-level\n   * counterpart to the AnnexB `preBuffer`, kept separately because the\n   * repacketizer path needs the original RTP wire packets, not AnnexB.\n   * Only populated for RTP sources (`isRtpSource()`).\n   */\n  private rtpRing: Buffer[] = []\n  private rtpRingCurAuStart = 0\n  private rtpRingPrevMarker = true\n  private rtpRingHasKeyframe = false\n  private rtpRingBytes = 0\n  /** Memory ceiling for the RTP ring. A GOP that exceeds this (pathological\n   *  long-GOP high-bitrate stream) drops the bootstrap rather than balloon\n   *  memory — late joiners fall back to waiting for the next keyframe. */\n  private static readonly RTP_RING_MAX_BYTES = 24 * 1024 * 1024\n\n  /** Stream stats tracking */\n  private totalBytes = 0\n  private bytesInWindow = 0\n  private packetsInWindow = 0\n  private windowStartMs = Date.now()\n  private lastKeyframeMs = 0\n  private idrIntervalMs = 0\n  private bitrateKbps = 0\n  private encodedInputFps = 0\n  private packetCount = 0\n  /**\n   * Timestamp of the most recent video packet observed on this broker.\n   * Used by the manager-level watchdog to drive `stream.online` /\n   * `stream.offline` transitions. `0` until the first packet arrives.\n   */\n  private lastPacketAt = 0\n\n  /** Default pre-buffer duration in seconds */\n  static readonly DEFAULT_PRE_BUFFER_SEC = 10\n  /** Maximum allowed pre-buffer duration in seconds */\n  static readonly MAX_PRE_BUFFER_SEC = 30\n\n  constructor(\n    deviceId: string,\n    decoderApi: DecoderCapApi | null,\n    logger?: IScopedLogger,\n    audioCodecApi?: AudioCodecCapApi | null,\n  ) {\n    this.deviceId = deviceId\n    this.decoderApi = decoderApi\n    this.audioCodecApi = audioCodecApi ?? null\n    this.logger = logger\n    this.pipeServer = new StreamPipeServer(logger?.child('pipe-server'))\n    this.rtspRestreamer = new RtspRestreamer(deviceId)\n    if (logger) this.rtspRestreamer.setLogger(logger.child('rtsp'))\n    this.preBuffer = new EncodedRingBuffer(StreamBroker.DEFAULT_PRE_BUFFER_SEC)\n    this.audioChunkPlane = new AudioChunkPlane(logger?.child('audio-chunk-plane'))\n\n    // Wire client-count callbacks so demand tracking reacts to RTSP/pipe changes\n    this.rtspRestreamer.onSessionCountChanged(() => this.checkDemand())\n    this.pipeServer.onClientCountChanged(() => this.checkDemand())\n  }\n\n  get status(): BrokerStatus {\n    return this._status\n  }\n\n  /**\n   * Ms epoch of the most recent video packet observed by the broker.\n   * `0` if none yet. Read by the manager-level health watchdog.\n   */\n  getLastPacketAt(): number {\n    return this.lastPacketAt\n  }\n\n  /** Source type of the active stream (rtsp/push/rtmp/placeholder/error). */\n  getActiveSourceType(): string {\n    return this.source?.type ?? 'unknown'\n  }\n\n  /**\n   * Diagnostic snapshot of the active source for log-meta enrichment.\n   * Always returns a stable shape so log readers can rely on the keys\n   * being present whenever a connect / read error is reported. URL is\n   * masked so credentials never reach disk.\n   */\n  private sourceMetaForLog(): {\n    brokerId: string\n    sourceType: string\n    url: string\n  } {\n    const src = this.source\n    const url = src && src.type !== 'placeholder' && src.type !== 'push' && src.type !== 'push-rtp'\n      ? maskUrlCredentials(src.url ?? '')\n      : ''\n    return {\n      brokerId: this.deviceId,\n      sourceType: src?.type ?? 'unknown',\n      url,\n    }\n  }\n\n  setRestreamers(restreamers: readonly IRestreamer[]): void {\n    this.restreamers = restreamers\n  }\n\n  /**\n   * Inject a manager-owned closure that derives the current\n   * `StreamSource` from the live cam-stream registry. Every dial\n   * attempt (initial start, scheduled reconnect, demand-driven resume)\n   * routes through this resolver so URL changes from a re-publish are\n   * picked up transparently. Calling without a provider keeps the\n   * legacy \"use the source from start()\" behaviour for tests.\n   */\n  setSourceProvider(provider: () => StreamSource | null): void {\n    this.sourceProvider = provider\n  }\n\n  /**\n   * Inject a manager-owned notifier the broker fires when a\n   * managed-loopback source (`rfc4571`) dial fails. The notifier\n   * asks the publishing addon to republish with a fresh transport;\n   * the next reconnect picks it up via `sourceProvider`.\n   */\n  setRequestSourceRefresh(notify: () => void): void {\n    this.requestSourceRefresh = notify\n  }\n\n  /**\n   * Fire `requestSourceRefresh` at most once per ~500 ms. The\n   * dial-failure path fans into onError + reader.connect().catch; we\n   * only want one round-trip to the publisher per real failure.\n   */\n  private notifySourceRefresh(): void {\n    if (!this.requestSourceRefresh) return\n    const now = Date.now()\n    if (now - this.lastRefreshRequestAt < 500) return\n    this.lastRefreshRequestAt = now\n    this.requestSourceRefresh()\n  }\n\n  /**\n   * Manager-side kick: the publisher just re-published with a fresh\n   * URL after a connect-fail / refresh round-trip (e.g. Reolink\n   * battery cam finished waking and bound a new rfc4571 port). Cancel\n   * any pending exponential-backoff retry, reset the backoff so the\n   * next failure starts from `INITIAL_RECONNECT_DELAY_MS` again, and\n   * arm a fast retry. No-op when the broker isn't currently waiting\n   * for a reconnect (already dialing, idle, or stopped).\n   *\n   * Without this, on a battery-cam wake-up the broker stays on its\n   * exponential schedule (1s → 2s → 4s …) even though the publisher\n   * has already published a working URL — wasting up to a full\n   * `MAX_RECONNECT_DELAY_MS` window dialing stale loopback ports.\n   */\n  kickReconnect(): void {\n    if (this.manualStop || this.stopping || !this.source) return\n    if (!this.hasDemand()) return\n    if (!this.reconnectTimer) return\n    clearTimeout(this.reconnectTimer)\n    this.reconnectTimer = undefined\n    this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n    this.lastRefreshRequestAt = 0\n    this.logger?.info('reconnect kicked — publisher refreshed source')\n    this.scheduleReconnect()\n  }\n\n  /**\n   * Resolve the source to use for the next dial attempt. Prefers the\n   * manager-supplied resolver — re-derives from the cam-stream\n   * registry, so a re-publish that updated the URL is honoured on the\n   * very next dial. Falls back to the cached `this.source` (set at\n   * `start()` time) when no resolver is wired.\n   */\n  private resolveCurrentSource(): StreamSource | null {\n    if (this.sourceProvider) {\n      const fresh = this.sourceProvider()\n      if (fresh) {\n        this.source = fresh\n        return fresh\n      }\n    }\n    return this.source ?? null\n  }\n\n\n\n  async start(source: StreamSource): Promise<void> {\n    this.source = source\n    this.manualStop = false\n    this.stopping = false\n\n    // Seed `detectedCodec` from the source's declared `videoCodec`\n    // so consumers (notably `BrokerWebrtcServer.createSession`) see\n    // the right codec immediately — without waiting for the reader to\n    // parse the SDP track. The race used to manifest on `adaptive`\n    // selection: a fresh session created against a still-suspended /\n    // connecting rfc4571 broker fell back to `'h264'` from\n    // `getStats().codec === undefined`, negotiated an H.264 SDP, then\n    // received H.265 packets once the reader got going → black\n    // viewport. The SDP `onVideoTrack` callback overwrites this later\n    // if it disagrees with the publisher's declaration.\n    if (source.videoCodec) {\n      this.detectedCodec = source.videoCodec\n    }\n\n    this.logger?.info('Broker starting', {\n      meta: {\n        sourceType: source.type,\n        ...(source.type !== 'placeholder' ? { url: maskUrlCredentials(source.url) } : {}),\n        ...(this.detectedCodec ? { codec: this.detectedCodec } : {}),\n      },\n    })\n\n    await this.pipeServer.start()\n    this.logger?.debug(\n      'Pipe server started',\n      { meta: { url: this.pipeServer.getUrl() } },\n    )\n\n    // Strict on-demand semantics for every pull source:\n    //\n    //   - `rtsp` / `rtmp` are dialed lazily because hammering a remote\n    //     camera with reconnect storms when nobody's watching is\n    //     useless and bandwidth-expensive.\n    //   - `rfc4571` is ALSO dialed lazily — keeping the loopback open\n    //     keeps the lib's\n    //     upstream Baichuan socket alive (good for reaction time) but\n    //     violates the user-visible rule \"stream open only when\n    //     someone's watching\". The lib's `ensureRfc4571Server` self-\n    //     heals if its TCP server idle-tore-down between dials, and\n    //     the manager's `sourceProvider` re-resolves a fresh URL on\n    //     every dial — a re-publish after recreate is observed\n    //     transparently.\n    //\n    // `checkDemand()` resumes the source the moment a subscriber\n    // arrives or the pre-buffer flips on (both feed `hasDemand`).\n    const isLazyDialSource =\n      source.type === 'rtsp'\n      || source.type === 'rtmp'\n      || source.type === 'rfc4571'\n    if (isLazyDialSource && !this.hasDemand()) {\n      this._suspended = true\n      this._status = 'idle'\n      this.logger?.info('Broker registered idle — no demand, source dial deferred')\n      return\n    }\n\n    this._status = 'connecting'\n\n    if (source.type === 'rtsp') {\n      this.startRtspReader(source)\n    } else if (source.type === 'rfc4571') {\n      this.startRfc4571Reader(source)\n    } else if (source.type === 'placeholder') {\n      this.startPlaceholderReader(source)\n    } else if (source.type === 'push' || source.type === 'push-rtp') {\n      // Push modes: the camera provider addon delivers data via\n      // pushEncodedPacket() (and pushVideoRtp() for `push-rtp`).\n      // Codec already seeded from `source.videoCodec` at the top of\n      // `start()` — the push branch used to repeat that assignment;\n      // it's redundant now and removed for clarity.\n      this._status = 'streaming'\n      this.logger?.info('Broker in push mode — waiting for data', {\n        meta: { sourceType: source.type, codec: this.detectedCodec ?? null },\n      })\n      this.resetPushStallTimer()\n    } else if (source.type === 'rtmp') {\n      this.startRtmpReader(source)\n    } else {\n      // Unknown source type — assume external push.\n      this._status = 'streaming'\n    }\n  }\n\n  async stop(): Promise<void> {\n    this.logger?.info('Broker stopping')\n    this.stopping = true\n    this.manualStop = true\n    this._status = 'stopped'\n\n    this.destroyNativeClient()\n    this.destroyRtmpReader()\n    this.stopErrorPlaceholder()\n    this.cancelFirstVideoWatchdog()\n    this.cancelRtpActivityWatchdog()\n\n    if (this.placeholderTimer) {\n      clearInterval(this.placeholderTimer)\n      this.placeholderTimer = undefined\n    }\n\n    if (this.reconnectTimer) {\n      clearTimeout(this.reconnectTimer)\n      this.reconnectTimer = undefined\n    }\n\n    if (this.suspendTimer) {\n      clearTimeout(this.suspendTimer)\n      this.suspendTimer = undefined\n    }\n\n    if (this.pushStallTimer) {\n      clearTimeout(this.pushStallTimer)\n      this.pushStallTimer = undefined\n    }\n\n    if (this.frameHandlePlane) {\n      const plane = this.frameHandlePlane\n      this.frameHandlePlane = null\n      await plane.dispose()\n    }\n\n    await this.pipeServer.stop()\n    this.rtspRestreamer.destroy()\n    this.preBuffer.clear()\n\n    this.pendingParamNals = []\n    this.firstKeyframeReceived = false\n    this.encodedCallbacks.clear()\n    this.audioChunkPlane.dispose()\n  }\n\n  // ── Auto-suspend / resume ─────────────────────────────────────────────\n\n  /** Whether the broker is currently suspended (idle, no RTSP connection). */\n  get suspended(): boolean { return this._suspended }\n\n  /** Set whether pre-buffer counts as demand. When false and no other consumers, the broker suspends. */\n  setPreBufferEnabled(enabled: boolean): void {\n    this._preBufferEnabled = enabled\n    this.checkDemand()\n  }\n\n  /** True when at least one consumer needs the stream alive. */\n  private hasDemand(): boolean {\n    if (this.encodedCallbacks.size > 0) return true\n    if (this.rtpVideoCallbacks.size > 0) return true\n    // Frame-handle subscribers (Phase 5 / D9 shm plane) count as demand —\n    // their decoder sessions need a live encoded-packet feed.\n    if ((this.frameHandlePlane?.subscriberCount ?? 0) > 0) return true\n    // Audio analysis subscribers count as demand. Without this, a\n    // camera with audio classification enabled but no decoded-video\n    // consumer (motion-wasm off + motion phase idle) suspends the\n    // broker between motion bursts; audio chunks stop flowing and\n    // `device.state.audioMetrics` stays frozen — the operator sees a\n    // stale \"X seconds ago\" chip in the UI even though the camera is\n    // online and audible. Mirrors video demand semantics.\n    if (this.audioChunkPlane.subscriberCount > 0) return true\n    if (this.rtspRestreamer.getSessionCount() > 0) return true\n    if (this.pipeServer.getClientCount() > 0) return true\n    // `_preBufferEnabled=true` keeps the stream warm even with zero\n    // active subscribers. Earlier the gate also required\n    // `preBuffer.getDuration() > 0` — meaning the broker could only\n    // STAY warm AFTER it had already received its first packets. On\n    // `suspend()` the pre-buffer gets cleared (intentional, to drop\n    // stale GOPs across the idle gap), so a momentarily empty buffer\n    // immediately voided demand again and the broker collapsed back\n    // to idle on the very next `checkDemand` poll. The natural read\n    // of the flag is \"keep the stream alive permanently\", so drop the\n    // duration gate: a camera that wants always-on stays always-on,\n    // and a battery camera explicitly sets `setPreBufferEnabled(false)`.\n    if (this._preBufferEnabled) return true\n    return false\n  }\n\n  /**\n   * Check demand and schedule suspend/resume. Called after every subscriber\n   * add/remove and after pre-buffer enable/disable.\n   */\n  private checkDemand(): void {\n    if (this.manualStop || this.stopping) return\n\n    if (this.hasDemand()) {\n      // Cancel pending suspend\n      if (this.suspendTimer) {\n        clearTimeout(this.suspendTimer)\n        this.suspendTimer = undefined\n      }\n      // Resume if suspended — re-resolve the source from the manager\n      // so a re-publish that landed during the idle window (e.g.\n      // Reolink rebound the rfc4571 server on a different port) is\n      // honoured on the very first dial of the resume.\n      //\n      // No proactive `notifySourceRefresh` here: forcing a publisher\n      // re-publish on every resume would call `ensureApi` ->\n      // `buildVideoStreamOptions` -> `ensureRfc4571Server` on the\n      // provider side, which wakes battery cameras even when the\n      // existing URL is still valid. The dial-failure path\n      // (`startRfc4571Reader`'s `.catch`) already triggers a refresh\n      // when the cached URL goes stale; that's the only case where\n      // we actually need the publisher to wake the cam.\n      if (this._suspended) {\n        const fresh = this.resolveCurrentSource()\n        if (fresh) {\n          this._suspended = false\n          this.logger?.info('demand detected — resuming stream')\n          this.start(fresh).catch((err) => {\n            this.logger?.error('resume failed', { meta: { error: errMsg(err) } })\n            this._suspended = true\n            this._status = 'idle'\n          })\n        }\n      }\n    } else if (!this._suspended) {\n      // Schedule suspend after grace period\n      if (!this.suspendTimer) {\n        this.suspendTimer = setTimeout(() => {\n          this.suspendTimer = undefined\n          if (!this.hasDemand() && !this.manualStop && !this.stopping) {\n            this.suspend()\n          }\n        }, SUSPEND_GRACE_MS)\n      }\n    }\n  }\n\n  /** Suspend the stream: kill ffmpeg/native client, clear buffers, go idle. Source is preserved for resume. */\n  private suspend(): void {\n    if (this._suspended) return\n    this._suspended = true\n    this._status = 'idle'\n\n    this.logger?.info('no demand — suspending stream')\n\n    this.destroyNativeClient()\n    this.destroyRtmpReader()\n    // Strict on-demand: drop the rfc4571 reader too. Holding the\n    // loopback TCP connection keeps the lib's upstream Baichuan\n    // socket pinned even with zero subscribers — exactly what the\n    // user-visible \"stream open only on demand\" rule rejects. The\n    // next demand triggers `start()` → `resolveCurrentSource()` →\n    // dial against the (potentially refreshed) loopback URL.\n    this.destroyRfc4571Reader()\n    // Activity watchdog has no purpose when the broker is intentionally\n    // idle — let `start()` re-arm it on the next packet.\n    this.cancelRtpActivityWatchdog()\n\n    if (this.reconnectTimer) {\n      clearTimeout(this.reconnectTimer)\n      this.reconnectTimer = undefined\n    }\n    if (this.placeholderTimer) {\n      clearInterval(this.placeholderTimer)\n      this.placeholderTimer = undefined\n    }\n\n    this.preBuffer.clear()\n    this.firstKeyframeReceived = false\n    this.pendingParamNals = []\n    this.rtspRestreamer.resetStreamState()\n\n    // Reset stats\n    this.totalBytes = 0\n    this.bytesInWindow = 0\n    this.packetsInWindow = 0\n    this.packetCount = 0\n    this.bitrateKbps = 0\n    this.encodedInputFps = 0\n  }\n\n  /**\n   * Declare push-source audio metadata. Push providers (Reolink Baichuan,\n   * Frigate) call this BEFORE pushing the first audio EncodedPacket so\n   * the broker can wire the audio-codec capability and emit decoded PCM\n   * to audio subscribers + transcode to G.711 µ-law for WebRTC.\n   *\n   * Pass `null` to retract — broker tears down the audio decode session.\n   */\n  pushAudioInfo(info: import('@camstack/types').PushAudioInfo | null): void {\n    if (this.stopping) return\n\n    if (info === null) {\n      if (this.audioCodecSession) {\n        const session = this.audioCodecSession\n        this.audioCodecSession = null\n        void session.close().catch((err) => {\n          this.logger?.warn('audio-codec: push session close failed', {\n            meta: { error: errMsg(err) },\n          })\n        })\n      }\n      this.audioTrackInfo = null\n      return\n    }\n\n    const codec = info.codec.toLowerCase()\n    this.audioCodec = codec.toUpperCase()\n    this.audioTrackInfo = {\n      codec: this.audioCodec,\n      sampleRate: info.sampleRate,\n      channels: info.channels,\n      supported: true,\n    }\n\n    // Native G.711 — no codec cap needed; pushEncodedPacket fans out\n    // directly to encodedCallbacks.\n    if (codec === 'pcmu' || codec === 'pcma') {\n      this.logger?.info('push-audio: native codec — no decoder cap required', {\n        meta: { codec, sampleRate: info.sampleRate, channels: info.channels },\n      })\n      return\n    }\n\n    if (!this.audioCodecApi) {\n      this.logger?.warn('push-audio: codec needs audio-codec cap but cap is unavailable', {\n        meta: { codec, sampleRate: info.sampleRate, channels: info.channels },\n      })\n      return\n    }\n\n    const cfg = this.buildAudioCodecSessionConfigFromPush(info)\n    if (!cfg) {\n      this.logger?.info('push-audio: codec not handled by cap — audio skipped', {\n        meta: { codec, sampleRate: info.sampleRate, channels: info.channels },\n      })\n      return\n    }\n\n    if (this.audioCodecSession) {\n      const prior = this.audioCodecSession\n      this.audioCodecSession = null\n      void prior.close().catch((err) => {\n        this.logger?.debug('audio-codec: prior session close failed', {\n          meta: { error: errMsg(err) },\n        })\n      })\n    }\n\n    this.audioCodecSession = new AudioCodecSession(\n      this.audioCodecApi,\n      cfg,\n      this.buildAudioCodecEmitCallback(),\n      this.logger?.child('audio-codec'),\n    )\n    this.logger?.info('push-audio: decode session armed', {\n      meta: {\n        codec: cfg.codec,\n        sourceSampleRate: cfg.sourceSampleRate,\n        sourceChannels: cfg.sourceChannels,\n      },\n    })\n  }\n\n  pushEncodedPacket(packet: EncodedPacket): void {\n    if (this.stopping) return\n\n    // Reset push-mode stall timer on every packet\n    if (this.source?.type === 'push') {\n      this.stopErrorPlaceholder()\n      if (this._status === 'error') this._status = 'streaming'\n      // Clear suspension if we were waiting on a battery camera to wake.\n      if (this._suspended && this.source.allowStall) {\n        this._suspended = false\n        this.logger?.info('push source resumed after stall — first packet received')\n      }\n      this.resetPushStallTimer()\n    }\n\n    // Push audio: route non-native codecs (e.g. AAC) through the\n    // audio-codec cap session if one was armed via pushAudioInfo. The\n    // session emits decoded PCM to audioSubscribers and transcodes to\n    // G.711 µ-law for WebRTC via the shared emit callback. Native\n    // codecs (PCMU/PCMA) fall through to the encodedCallbacks fanout\n    // below — no transcode required.\n    if (packet.type === 'audio' && this.source?.type === 'push' && this.audioCodecSession) {\n      const codec = packet.codec.toLowerCase()\n      if (codec !== 'pcmu' && codec !== 'pcma') {\n        this.audioCodecSession.pushRawFrame(packet.data)\n        return\n      }\n    }\n\n    // Track bitrate + IDR interval\n    if (packet.type === 'video') {\n      this.packetCount++\n      this.totalBytes += packet.data.length\n      this.bytesInWindow += packet.data.length\n      this.packetsInWindow++\n      this.lastPacketAt = Date.now()\n\n      const now = Date.now()\n      const windowMs = now - this.windowStartMs\n      if (windowMs >= 2000) {\n        this.bitrateKbps = Math.round((this.bytesInWindow * 8) / windowMs)\n        this.encodedInputFps = (this.packetsInWindow / windowMs) * 1000\n        this.bytesInWindow = 0\n        this.packetsInWindow = 0\n        this.windowStartMs = now\n      }\n\n      if (packet.keyframe) {\n        if (this.lastKeyframeMs > 0) {\n          this.idrIntervalMs = now - this.lastKeyframeMs\n        }\n        this.lastKeyframeMs = now\n      }\n    }\n\n    // Buffer encoded video packets for pre-buffer (late-join).\n    // ONE pre-buffer per source: RTP sources (RTSP) use the raw source-RTP\n    // ring (`captureRtpForPreBuffer` / `getRtpPreBuffer`) for the repacketizer\n    // path, so the AnnexB pre-buffer would be a redundant second copy of the\n    // same GOP — its video is never flushed for RTP sources (the WebRTC\n    // AnnexB flush is gated on push sources) and profile derivation reads SDP\n    // first. So only AnnexB/push sources populate this one.\n    if (packet.type === 'video' && !this.isRtpSource()) {\n      this.preBuffer.push(packet)\n    }\n\n    for (const [cb, entry] of this.encodedCallbacks) {\n      cb(packet)\n      entry.packetsDelivered += 1\n    }\n\n    // Broadcast raw video bytes to TCP pipe clients (restreamers)\n    if (packet.type === 'video') {\n      this.pipeServer.broadcast(packet.data)\n\n      // Feed RTSP restreamer (Annex-B mode only — in passthrough mode, raw RTP\n      // is forwarded directly via pushRtpPassthrough in the onVideoRtp callback)\n      if (!this.rtspRestreamer.rtpPassthroughMode) {\n        this.rtspRestreamer.pushPacket(packet)\n      }\n\n      // Gate: wait for the first keyframe before sending data to the decoder.\n      // H264/H265 decoders need SPS/PPS (present in keyframes) to initialize.\n      if (!this.firstKeyframeReceived && packet.keyframe) {\n        this.firstKeyframeReceived = true\n        this.logger?.info('First keyframe received — decoder can start')\n      }\n    }\n\n    for (const restreamer of this.restreamers) {\n      const streamId = `${this.deviceId}/video`\n      restreamer.pushPacket(streamId, packet)\n    }\n\n    // Only push to the decoder after the first keyframe — the decoder needs\n    // SPS/PPS to start. Without this, streams that start mid-GOP silently\n    // produce 0 decoded frames. Feed the shm frame-handle plane's per-format\n    // decoder sessions the encoded packet (Phase 5 / D9) — a no-op when no\n    // consumer has subscribed; the plane ignores audio packets internally.\n    if (packet.type === 'video' && this.firstKeyframeReceived) {\n      this.frameHandlePlane?.pushPacket(packet)\n    }\n  }\n\n  onEncodedData(\n    callback: (packet: EncodedPacket) => void,\n    attribution: BrokerConsumerAttribution = UNKNOWN_ATTRIBUTION,\n  ): Unsubscribe {\n    this.subscriberIdCounter += 1\n    const entry: EncodedSubscriberEntry = {\n      id: `enc-${this.deviceId}-${this.subscriberIdCounter}`,\n      channel: 'annexb',\n      attribution,\n      subscribedAt: Date.now(),\n      packetsDelivered: 0,\n    }\n    this.encodedCallbacks.set(callback, entry)\n    if (this.streamingDebug) {\n      this.logger?.info('encoded subscriber added', {\n        meta: {\n          total: this.encodedCallbacks.size,\n          subscriberId: entry.id,\n          kind: attribution.kind,\n          label: attribution.label,\n        },\n      })\n    }\n\n    // Seed the new subscriber with the SDP-derived param sets so\n    // WebRTC sessions get VPS/SPS/PPS even when the camera doesn't\n    // emit them in-band (Reolink high-profile streams).\n    this.emitSdpParamSetsTo(callback)\n    this.checkDemand()\n\n    return () => {\n      this.encodedCallbacks.delete(callback)\n      if (this.streamingDebug) {\n        this.logger?.info('encoded subscriber removed', {\n          meta: { total: this.encodedCallbacks.size, subscriberId: entry.id },\n        })\n      }\n      this.checkDemand()\n    }\n  }\n\n  /**\n   * Emit the cached SDP param sets as a synthetic Annex-B EncodedPacket\n   * to a single subscriber. No-op if no param sets are cached or no\n   * codec is detected yet.\n   */\n  private emitSdpParamSetsTo(callback: (packet: EncodedPacket) => void): void {\n    if (!this.sdpParameterSets || this.sdpParameterSets.length === 0) return\n    const codec = this.detectedCodec\n    if (!codec) return\n    const startCode = Buffer.from([0x00, 0x00, 0x00, 0x01])\n    const annexB = Buffer.concat(\n      this.sdpParameterSets.flatMap((ps) => [startCode, ps]),\n    )\n    if (this.streamingDebug) {\n      this.logger?.info('Emitting SDP param sets to encoded subscriber', {\n        meta: {\n          codec,\n          psCount: this.sdpParameterSets.length,\n          psSizes: this.sdpParameterSets.map((p) => p.length).join(','),\n          psFirstBytesHex: this.sdpParameterSets.map((p) => p.subarray(0, 4).toString('hex')).join('|'),\n          annexBLength: annexB.length,\n        },\n      })\n    }\n    callback({\n      type: 'video',\n      data: annexB,\n      pts: 0,\n      dts: 0,\n      keyframe: false,\n      codec,\n    })\n  }\n\n  /** Replay SDP param sets to every existing encoded subscriber. */\n  private replaySdpParamSetsToEncodedSubscribers(): void {\n    if (!this.sdpParameterSets || this.encodedCallbacks.size === 0) return\n    for (const cb of this.encodedCallbacks.keys()) this.emitSdpParamSetsTo(cb)\n  }\n\n  /**\n   * Subscribers waiting on the SDP-derived VPS/SPS/PPS to land. The\n   * H.265 source-RTP path needs to seed its `H265Repacketizer` with\n   * these param sets so the AP codec packet can fire ahead of the\n   * first IDR — without it Chrome's HEVC decoder never initialises on\n   * cameras that only ship parameter sets in the SDP (Reolink high-\n   * profile streams). Sessions created while the broker is still\n   * dialing rfc4571 (battery-cam wake-up) have null `sdpParameterSets`\n   * at create time; this subscription delivers them as soon as\n   * `onVideoTrack` fires. Mirrors `replaySdpParamSetsToEncodedSubscribers`\n   * for the source-RTP path.\n   */\n  private readonly sdpParameterSetsCallbacks = new Set<(ps: ReadonlyArray<Buffer>) => void>()\n\n  /**\n   * Subscribe to SDP-derived VPS/SPS/PPS deliveries. The callback\n   * fires once immediately if the broker already has them (synchronous\n   * delivery for late subscribers), and again every time\n   * `onVideoTrack` repopulates them on a reader rebuild.\n   */\n  onSdpParameterSets(callback: (ps: ReadonlyArray<Buffer>) => void): Unsubscribe {\n    this.sdpParameterSetsCallbacks.add(callback)\n    if (this.sdpParameterSets && this.sdpParameterSets.length > 0) {\n      try {\n        callback(this.sdpParameterSets)\n      } catch (err) {\n        this.logger?.warn('sdp param-set subscriber threw on initial delivery', {\n          meta: { error: errMsg(err) },\n        })\n      }\n    }\n    return () => {\n      this.sdpParameterSetsCallbacks.delete(callback)\n    }\n  }\n\n  /** Fan out the current SDP param sets to every waiting subscriber. */\n  private replaySdpParamSetsToRtpSubscribers(): void {\n    if (!this.sdpParameterSets || this.sdpParameterSetsCallbacks.size === 0) return\n    const ps = this.sdpParameterSets\n    for (const cb of this.sdpParameterSetsCallbacks) {\n      try {\n        cb(ps)\n      } catch (err) {\n        this.logger?.warn('sdp param-set subscriber threw', {\n          meta: { error: errMsg(err) },\n        })\n      }\n    }\n  }\n\n  onVideoRtp(\n    callback: (rtpData: Buffer) => void,\n    attribution: BrokerConsumerAttribution = UNKNOWN_ATTRIBUTION,\n  ): Unsubscribe {\n    this.subscriberIdCounter += 1\n    const entry: EncodedSubscriberEntry = {\n      id: `rtp-${this.deviceId}-${this.subscriberIdCounter}`,\n      channel: 'rtp',\n      attribution,\n      subscribedAt: Date.now(),\n      packetsDelivered: 0,\n    }\n    this.rtpVideoCallbacks.set(callback, entry)\n    if (this.streamingDebug) {\n      this.logger?.info('rtp video subscriber added', {\n        meta: {\n          total: this.rtpVideoCallbacks.size,\n          subscriberId: entry.id,\n          kind: attribution.kind,\n          label: attribution.label,\n        },\n      })\n    }\n    this.checkDemand()\n    return () => {\n      this.rtpVideoCallbacks.delete(callback)\n      if (this.streamingDebug) {\n        this.logger?.info('rtp video subscriber removed', {\n          meta: { total: this.rtpVideoCallbacks.size, subscriberId: entry.id },\n        })\n      }\n      this.checkDemand()\n    }\n  }\n\n  /**\n   * Re-tag every subscriber whose attribution carries `sessionId` with the\n   * supplied patch (e.g. a remote address discovered once ICE connects, or a\n   * User-Agent resolved at the signaling edge). Both the audio + video rows of\n   * a WebRTC session share one `sessionId`, so a single call updates both.\n   *\n   * The stored attribution is never mutated — the entry's pointer is swapped\n   * to a freshly merged object. Returns true when at least one entry matched.\n   */\n  updateConsumerAttributionBySession(\n    sessionId: string,\n    patch: Partial<BrokerConsumerAttribution>,\n  ): boolean {\n    let matched = false\n    for (const entry of this.encodedCallbacks.values()) {\n      if (entry.attribution.sessionId === sessionId) {\n        entry.attribution = { ...entry.attribution, ...patch }\n        matched = true\n      }\n    }\n    for (const entry of this.rtpVideoCallbacks.values()) {\n      if (entry.attribution.sessionId === sessionId) {\n        entry.attribution = { ...entry.attribution, ...patch }\n        matched = true\n      }\n    }\n    return matched\n  }\n\n  getSdpParameterSets(): ReadonlyArray<Buffer> | null {\n    return this.sdpParameterSets\n  }\n\n  /**\n   * Append a source RTP packet to the pre-buffer ring, trimming it to begin\n   * at the most recent keyframe access unit. Access-unit boundaries are\n   * detected via the RTP marker bit (set on the last packet of a frame);\n   * keyframe-ness comes from the depacketizer flags set just before this\n   * call. The ring therefore always holds `[current keyframe AU start .. now]`.\n   */\n  private captureRtpForPreBuffer(rtpData: Buffer): void {\n    // marker bit: RTP header byte 1, bit 7 — last packet of an access unit.\n    const marker = rtpData.length > 1 && (rtpData[1]! & 0x80) !== 0\n    const auStart = this.rtpRingPrevMarker || this.rtpRing.length === 0\n    if (auStart) this.rtpRingCurAuStart = this.rtpRing.length\n    this.rtpRing.push(rtpData)\n    this.rtpRingBytes += rtpData.length\n\n    // This packet's NAL began/contained keyframe data → the current AU is a\n    // keyframe AU. Trim the ring to start at its first packet so a fresh\n    // viewer gets the minimal current GOP.\n    if (this._lastNalParamSet || this._lastNalKeyframe) {\n      if (this.rtpRingCurAuStart > 0) {\n        const dropped = this.rtpRing.splice(0, this.rtpRingCurAuStart)\n        for (const b of dropped) this.rtpRingBytes -= b.length\n        this.rtpRingCurAuStart = 0\n      }\n      this.rtpRingHasKeyframe = true\n    }\n\n    this.rtpRingPrevMarker = marker\n\n    // Memory ceiling: a GOP longer than the cap drops the bootstrap entirely\n    // (late joiners wait for the next keyframe) rather than ballooning memory.\n    if (this.rtpRingBytes > StreamBroker.RTP_RING_MAX_BYTES) {\n      this.resetRtpPreBuffer()\n    }\n  }\n\n  /**\n   * Reset the source-RTP pre-buffer ring to its empty/cold state. Called on\n   * every reader teardown (`resetRtpPipeline`) so a stale GOP from a prior\n   * session can NEVER leak across an on-demand idle window — and on cap\n   * overflow. Without this, a redial that begins mid-GOP (no IDR yet) would\n   * replay the previous session's stale `rtpRingHasKeyframe`-true ring to a\n   * fresh viewer (the `1 packet, sent=0` cold-start symptom): a degenerate,\n   * non-decodable slice instead of waiting for the new dial's first keyframe.\n   */\n  private resetRtpPreBuffer(): void {\n    this.rtpRing = []\n    this.rtpRingBytes = 0\n    this.rtpRingCurAuStart = 0\n    this.rtpRingPrevMarker = true\n    this.rtpRingHasKeyframe = false\n  }\n\n  /**\n   * Snapshot of the source-RTP pre-buffer (the current GOP from its\n   * keyframe) for a late-joining repacketizer-path viewer to replay on\n   * connect — instant decoder start. Empty until a keyframe has been seen\n   * (or after a cap overflow / reader teardown), in which case the viewer\n   * falls back to waiting for the camera's next keyframe.\n   */\n  getRtpPreBuffer(): readonly Buffer[] {\n    return this.rtpRingHasKeyframe ? this.rtpRing.slice() : []\n  }\n\n  getSourceType(): StreamSourceType | null {\n    return this.source?.type ?? null\n  }\n\n  /**\n   * True when the broker emits source RTP packets via `onVideoRtp`.\n   * RTSP demuxes incoming RTP directly; rfc4571 (RTP-over-TCP framed\n   * per RFC 4571) deserializes into the same wire shape and runs\n   * through the same `buildRtpStreamCallbacks.onVideoRtp` path;\n   * `push-rtp` sources synthesize it via `pushVideoRtp()`. Bare\n   * `push` (AnnexB-only) and `rtmp` (no RTP wire) return false.\n   *\n   * Crucial for the WebRTC dispatch decision: H.265 sources marked\n   * as RTP route through `H265Repacketizer` for source-RTP forwarding\n   * (Chrome's HEVC depacketizer requires the original RTP header\n   * layout); non-RTP sources fall back to AnnexB → `writeVideoNals`\n   * with cached-keyframe-on-PLI. Marking rfc4571 as non-RTP put the\n   * Reolink Baichuan H.265 native:main stream on the wrong path —\n   * Chrome got stuck in a PLI loop because each cached re-send\n   * collided with the live decode state.\n   */\n  isRtpSource(): boolean {\n    const t = this.source?.type\n    return t === 'rtsp' || t === 'rfc4571' || t === 'push-rtp'\n  }\n\n  /**\n   * Push a raw RTP packet from an external `push-rtp` source. Captures into\n   * the source-RTP pre-buffer (so a late-joining repacketizer-path viewer\n   * can instant-start from the current GOP) and fans out to every\n   * `onVideoRtp` subscriber.\n   *\n   * `push-rtp` providers deliver AnnexB-equivalents separately via\n   * `pushEncodedPacket()` (see `IStreamBroker` contract) — so this path\n   * does NOT re-emit EncodedPackets; it only derives the per-packet\n   * keyframe/param-set flags the pre-buffer capture needs (via a\n   * flag-only depacketizer that never touches the AnnexB plane), keeping\n   * the AnnexB consumers (decoder, restreamer) single-fed by the provider.\n   *\n   * The pre-buffer is captured even with zero subscribers attached — a\n   * viewer that connects mid-GOP must be able to replay the GOP that\n   * accumulated before it arrived. The fan-out is what no-ops without\n   * subscribers, so providers can still fire-and-forget.\n   */\n  pushVideoRtp(rtpData: Buffer): void {\n    if (this.stopping) return\n\n    // Rolling activity watchdog: every source-RTP packet (push-rtp,\n    // native RTSP, rfc4571) resets the silence timer. Mirrors the\n    // `buildRtpStreamCallbacks.onVideoRtp` arm so a push-rtp source\n    // benefits from the same stall detection even though its expiry\n    // is informational-only (no reader to tear down).\n    this.kickRtpActivityWatchdog()\n\n    // Derive the keyframe/param-set flags the capture reads. Reset\n    // per-packet so each capture sees only this packet's flags.\n    this._lastNalKeyframe = false\n    this._lastNalParamSet = false\n    this.ensurePushRtpFlagDepacketizer().processPacket(rtpData)\n\n    // Capture into the source-RTP pre-buffer (current GOP) BEFORE fan-out\n    // so a subscriber bootstrapping from it sees this packet as the tail —\n    // exactly as the native reader does (`buildRtpStreamCallbacks.onVideoRtp`).\n    this.captureRtpForPreBuffer(rtpData)\n\n    if (this.rtpVideoCallbacks.size === 0) return\n    for (const [cb, entry] of this.rtpVideoCallbacks) {\n      try {\n        cb(rtpData)\n        entry.packetsDelivered += 1\n      } catch (err) {\n        this.logger?.warn('rtp video subscriber threw', {\n          meta: { error: errMsg(err) },\n        })\n      }\n    }\n  }\n\n  /**\n   * Lazily build a flag-only RTP depacketizer for the `push-rtp` source\n   * path. Unlike the native reader's depacketizer (which emits AnnexB\n   * EncodedPackets via `handleDepacketizedNal`), this callback ONLY sets\n   * `_lastNalKeyframe`/`_lastNalParamSet` — the push provider owns the\n   * AnnexB plane through `pushEncodedPacket`, so re-emitting here would\n   * double-feed the decoder/restreamer. push-rtp has no SDP `onVideoTrack`,\n   * so the codec comes from the source declaration (`detectedCodec`, seeded\n   * in `start()`). Dropped on teardown (`resetRtpPipeline`) so a redial\n   * starts with clean FU reassembly state.\n   */\n  private ensurePushRtpFlagDepacketizer(): RtpDepacketizer {\n    if (!this.pushRtpFlagDepacketizer) {\n      const codec = this.detectedCodec === 'h265' ? 'h265' : 'h264'\n      this.pushRtpFlagDepacketizer = new RtpDepacketizer(codec, ({ keyframe, parameterSet }) => {\n        if (keyframe) this._lastNalKeyframe = true\n        if (parameterSet) this._lastNalParamSet = true\n      })\n    }\n    return this.pushRtpFlagDepacketizer\n  }\n\n  /** Toggle the per-device streaming debug flag. */\n  setStreamingDebug(enabled: boolean): void {\n    this.streamingDebug = enabled\n  }\n\n  /**\n   * Test seam exposing the source-RTP activity watchdog state. Production\n   * code MUST NOT read this — it exists so unit tests can assert recovery\n   * behavior without inspecting private timer handles (which can't be\n   * compared by reference, and tracking them externally would race with\n   * arm/cancel paths). Mirrors the `firstVideoWatchdog` behavior but\n   * fires on every-packet activity rather than first-packet only.\n   */\n  getRtpActivityWatchdogStatus(): { armed: boolean; firedCount: number } {\n    return {\n      armed: this.rtpActivityWatchdog !== undefined,\n      firedCount: this.rtpActivityWatchdogFiredCount,\n    }\n  }\n\n  // ── Decoded audio-chunk plane (Phase 5 / D9) ─────────────────────────\n\n  /**\n   * Open a decoded audio-chunk subscription — the poll-based, tRPC-reachable\n   * replacement for the live-object `onDecodedAudioChunk` callback. The\n   * broker registers a per-subscription FIFO queue and returns a\n   * `subscriptionId` the consumer drains via `pullAudioChunks`.\n   */\n  subscribeAudioChunks(options?: { readonly tag?: string }): SubscribeAudioChunksResult {\n    const tag = options?.tag\n    if (!tag) {\n      this.logger?.warn(`subscribeAudioChunks called without tag — listClients will show 'unknown'`)\n    }\n    const result = this.audioChunkPlane.subscribe({ tag })\n    // Audio counts as demand (see hasDemand). Trigger the same\n    // suspend/resume reconcile that video subscribers do — without\n    // this, attaching to a suspended broker leaves it suspended and\n    // no chunks ever arrive.\n    this.checkDemand()\n    return result\n  }\n\n  /** Drain up to `maxCount` `DecodedAudioChunk`s for an audio-chunk subscription. */\n  pullAudioChunks(subscriptionId: string, maxCount: number): readonly DecodedAudioChunk[] {\n    return this.audioChunkPlane.pull(subscriptionId, maxCount)\n  }\n\n  /** Release an audio-chunk subscription. Returns `true` when it existed. */\n  unsubscribeAudioChunks(subscriptionId: string): boolean {\n    const released = this.audioChunkPlane.unsubscribe(subscriptionId)\n    if (released) this.checkDemand()\n    return released\n  }\n\n  // ── Shared-memory frame-handle plane (Phase 5 / D9) ──────────────────\n\n  /**\n   * Resolve the stream descriptor the `FrameHandlePlane` needs to create a\n   * `frameSink: 'shm'` decoder session — codec, numeric device id, log tag,\n   * and (for a pull-mode decoder) the local pipe URL. Returns `null` before\n   * the broker has a source.\n   */\n  private resolveFrameHandleStreamInfo(): FrameHandlePlaneStreamInfo | null {\n    const codec = this.detectedCodec ?? this.source?.videoCodec\n    if (!codec || !this.source) return null\n    const slash = this.deviceId.indexOf('/')\n    const numericDeviceId = slash >= 0\n      ? Number.parseInt(this.deviceId.slice(0, slash), 10)\n      : Number.parseInt(this.deviceId, 10)\n    // Push-mode decoders (the node-av default) consume the broker's\n    // `pushPacket` feed; only a pull-mode decoder needs the local pipe URL.\n    // The plane decides per-session via this value.\n    return {\n      codec,\n      ...(Number.isFinite(numericDeviceId) ? { numericDeviceId } : {}),\n      tag: `broker:${this.deviceId}`,\n      // push-mode is the node-av default today; pull-mode URL wiring is deferred to a later Phase-5 task.\n      pullModeUrl: null,\n    }\n  }\n\n  /** Lazily construct the frame-handle plane, returning `null` with no decoder. */\n  private ensureFrameHandlePlane(): FrameHandlePlane | null {\n    if (this.frameHandlePlane) return this.frameHandlePlane\n    if (!this.decoderApi) {\n      this.logger?.warn('subscribeFrameHandles: no decoder API — frame handles unavailable')\n      return null\n    }\n    this.frameHandlePlane = new FrameHandlePlane({\n      decoderApi: this.decoderApi,\n      logger: this.logger?.child('frame-handle-plane'),\n      resolveStreamInfo: () => this.resolveFrameHandleStreamInfo(),\n    })\n    return this.frameHandlePlane\n  }\n\n  /**\n   * Subscribe to this broker's shared-memory frame-handle stream — the broker's\n   * decoded-frame surface. The broker spins up (or reuses) a `frameSink: 'shm'`\n   * decoder session producing `input.format` and returns a `subscriptionId`\n   * the consumer polls via `pullFrameHandles`. fps throttling is implicit\n   * (latest-wins ring reads); `input.maxFps` is echoed as a cadence hint.\n   */\n  async subscribeFrameHandles(\n    input: Omit<SubscribeFramesInput, 'brokerId'>,\n  ): Promise<SubscribeFramesResult> {\n    const plane = this.ensureFrameHandlePlane()\n    if (!plane) {\n      throw new Error('stream-broker: no decoder available for frame-handle subscription')\n    }\n    this.checkDemand()\n    const { subscriptionId, maxFps } = await plane.subscribe({\n      format: input.format,\n      maxFps: input.maxFps,\n      tag: input.tag,\n    })\n    return { subscriptionId, maxFps }\n  }\n\n  /** Drain up to `maxCount` `FrameHandle`s for a frame-handle subscription. */\n  pullFrameHandles(subscriptionId: string, maxCount: number): readonly FrameHandle[] {\n    return this.frameHandlePlane?.pullHandles(subscriptionId, maxCount) ?? []\n  }\n\n  /** Release a frame-handle subscription. Returns `true` when it existed. */\n  async unsubscribeFrameHandles(subscriptionId: string): Promise<boolean> {\n    const plane = this.frameHandlePlane\n    if (!plane) return false\n    const released = await plane.unsubscribe(subscriptionId)\n    if (released && plane.subscriberCount === 0) {\n      this.checkDemand()\n    }\n    return released\n  }\n\n  getStats(): BrokerStats {\n    // Phase 5 / D9: decoded frames flow through the shm frame plane's\n    // per-format decoder sessions. `decodeFps` is the plane's aggregate\n    // decoded-frame rate; `inputFps` stays from the encoded-packet counter.\n    return {\n      status: this._suspended ? 'idle' : this._status,\n      inputFps: this.encodedInputFps,\n      decodeFps: this.frameHandlePlane?.decodeFps() ?? 0,\n      encodedSubscribers: this.encodedCallbacks.size + this.rtpVideoCallbacks.size,\n      decodedSubscribers: this.frameHandlePlane?.subscriberCount ?? 0,\n      uptimeMs: Date.now() - this.startedAt,\n      bitrateKbps: this.bitrateKbps,\n      idrIntervalMs: this.idrIntervalMs,\n      codec: this.detectedCodec ?? this.source?.videoCodec,\n      totalBytes: this.totalBytes,\n      packetCount: this.packetCount,\n      rtspClients: this.rtspRestreamer.getSessionCount(),\n      pipeClients: this.pipeServer.getClientCount(),\n      preBufferSec: this.preBuffer.getDuration(),\n      preBufferMs: this.preBuffer.getBufferedDurationMs(),\n      preBufferPackets: this.preBuffer.getPacketCount(),\n      decoderNodeId: this.frameHandlePlane?.decoderNodeIds()[0] ?? null,\n      audio: this.audioTrackInfo ?? null,\n    }\n  }\n\n  /**\n   * Per-broker consumer inventory (Phase 10). Returns the live subscriber\n   * list grouped by transport — every RTSP client, decoded-frame\n   * subscriber, and audio-chunk subscriber. Used by the admin UI's\n   * Streams tab and by operators diagnosing \"phantom\" holders keeping a\n   * broker open.\n   */\n  listClients(): {\n    readonly rtsp: readonly {\n      readonly sessionId: string\n      readonly remoteAddr: string\n      readonly playing: boolean\n      readonly muted: boolean\n      readonly connectedAt: number\n      readonly lastRtpAt: number\n      readonly bytesSent: number\n    }[]\n    readonly decoded: readonly {\n      readonly tag: string\n      readonly subscribedAt: number\n      readonly maxFps: number\n      readonly framesDelivered: number\n      readonly framesDropped: number\n    }[]\n    readonly audio: readonly {\n      readonly tag: string\n      readonly subscribedAt: number\n      readonly chunksDelivered: number\n    }[]\n    readonly encoded: readonly BrokerEncodedClient[]\n    readonly pipeClients: number\n    readonly encodedSubscribers: number\n  } {\n    const encoded: BrokerEncodedClient[] = []\n    for (const entry of this.encodedCallbacks.values()) {\n      encoded.push({\n        id: entry.id,\n        channel: entry.channel,\n        attribution: entry.attribution,\n        subscribedAt: entry.subscribedAt,\n        packetsDelivered: entry.packetsDelivered,\n      })\n    }\n    for (const entry of this.rtpVideoCallbacks.values()) {\n      encoded.push({\n        id: entry.id,\n        channel: entry.channel,\n        attribution: entry.attribution,\n        subscribedAt: entry.subscribedAt,\n        packetsDelivered: entry.packetsDelivered,\n      })\n    }\n\n    return {\n      rtsp: this.rtspRestreamer.listSessionInfos(),\n      // Phase 5 / D9: the `decoded` channel is the shm frame-handle plane's\n      // subscriber set. `framesDropped` is always 0 — a slow consumer's\n      // dropped frames are recycled silently at the ring, not counted here.\n      decoded: (this.frameHandlePlane?.listSubscribers() ?? []).map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        maxFps: s.maxFps,\n        framesDelivered: s.framesDelivered,\n        framesDropped: 0,\n      })),\n      audio: this.audioChunkPlane.listSubscribers().map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        chunksDelivered: s.chunksDelivered,\n      })),\n      encoded,\n      pipeClients: this.pipeServer.getClientCount(),\n      encodedSubscribers: this.encodedCallbacks.size + this.rtpVideoCallbacks.size,\n    }\n  }\n\n  /**\n   * Force-disconnect a single consumer. Matches by channel:\n   *   - `rtsp`: by `sessionId` (calls `RtspRestreamer.killSession`).\n   *   - `decoded`: by `tag` — releases every shm frame-handle subscription\n   *     whose tag matches (the plane winds the decoder session down once its\n   *     last subscriber leaves).\n   *   - `audio`: by `tag` — drops every audio subscriber whose tag matches,\n   *     then re-evaluates demand.\n   * Returns `true` when at least one consumer was dropped.\n   */\n  killClient(channel: 'rtsp' | 'decoded' | 'audio', handle: string): boolean {\n    if (channel === 'rtsp') {\n      return this.rtspRestreamer.killSession(handle)\n    }\n    if (channel === 'decoded') {\n      const plane = this.frameHandlePlane\n      if (!plane) return false\n      const matched = plane.listSubscribers().some((s) => s.tag === handle)\n      if (!matched) return false\n      // Plane unsubscribe is async (decoder-session teardown); fire it in the\n      // background — the caller only needs the synchronous \"matched\" verdict.\n      plane.killByTag(handle).then((killed) => {\n        this.logger?.info('frame-handle subscribers force-disconnected', {\n          meta: { handle, killed },\n        })\n        this.checkDemand()\n      }).catch((err: unknown) => {\n        this.logger?.warn('frame-handle killByTag failed', {\n          meta: { handle, error: errMsg(err) },\n        })\n      })\n      return true\n    }\n    const killed = this.audioChunkPlane.killByTag(handle)\n    if (killed === 0) return false\n    this.logger?.info('audio subscribers force-disconnected', {\n      meta: { handle, killed, remaining: this.audioChunkPlane.subscriberCount },\n    })\n    this.checkDemand()\n    return true\n  }\n\n  /**\n   * Returns the local TCP URL that restreamers should connect to\n   * instead of the camera's RTSP URL. This ensures the broker is\n   * the sole reader from the camera.\n   */\n  getLocalStreamUrl(): string {\n    return this.pipeServer.getUrl()\n  }\n\n  /** Get the RTSP restreamer instance (for RtspListenServer registration) */\n  getRtspRestreamer(): RtspRestreamer {\n    return this.rtspRestreamer\n  }\n\n  /** Returns the number of TCP pipe clients currently connected */\n  getPipeClientCount(): number {\n    return this.pipeServer.getClientCount()\n  }\n\n  /**\n   * Get the pre-buffer: the most recent N seconds of encoded video packets,\n   * starting from the last keyframe. Returns packets in chronological order.\n   * Useful for clip creation (get the last N seconds on demand) and\n   * late-joining consumers that need to start from a keyframe.\n   */\n  getPreBuffer(): readonly EncodedPacket[] {\n    return this.preBuffer.getPackets()\n  }\n\n  /** Set the pre-buffer duration in seconds (clamped to 0..MAX_PRE_BUFFER_SEC). */\n  setPreBufferDuration(seconds: number): void {\n    const clamped = Math.max(0, Math.min(seconds, StreamBroker.MAX_PRE_BUFFER_SEC))\n    this.preBuffer.setDuration(clamped)\n    this.logger?.info('Pre-buffer duration set', { meta: { seconds: clamped } })\n  }\n\n  /** Get the current pre-buffer duration in seconds. */\n  getPreBufferDuration(): number {\n    return this.preBuffer.getDuration()\n  }\n\n  private startRtspReader(source: StreamSource): void {\n    this.startNativeRtspReader(source)\n  }\n\n  /**\n   * Native RTMP reader — pulls media from an RTMP URL, demuxes FLV tags\n   * into AnnexB access units, and routes them through `pushEncodedPacket`\n   * (same fan-out as Reolink Baichuan / Frigate push sources).\n   *\n   * Codec support: H.264 standard FLV + H.265 Enhanced FLV (FOURCC\n   * 'hvc1' / 'hev1'). Audio: AAC only — non-AAC tracks are skipped.\n   *\n   * On terminal error: schedules a reconnect with the same exponential\n   * backoff used by the RTSP path.\n   */\n  private startRtmpReader(source: StreamSource): void {\n    this.logger?.info('starting RTMP reader', { meta: { url: maskUrlCredentials(source.url) } })\n\n    this.destroyRtmpReader()\n\n    const reader = new RtmpReader(source.url, this.logger?.child('rtmp'), {\n      onVideoCodec: (codec) => {\n        this.detectedCodec = codec\n        this.logger?.info('rtmp: video codec detected', { meta: { codec } })\n      },\n      onEncodedPacket: (packet) => {\n        if (this.stopping) return\n        this.pushEncodedPacket(packet)\n      },\n      onAudioInfo: (info) => {\n        if (this.stopping) return\n        this.pushAudioInfo(info)\n      },\n      onPlaying: () => {\n        this.logger?.info('rtmp: streaming')\n        this._status = 'streaming'\n        this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n        this.stopErrorPlaceholder()\n      },\n      onError: (error) => {\n        if (this.manualStop) return\n        this.logger?.warn('rtmp error', { meta: { error: error.message } })\n        this.destroyRtmpReader()\n        this._status = 'error'\n        this.scheduleReconnect()\n      },\n      onTeardown: () => {\n        this.rtmpReader = null\n      },\n    })\n\n    this.rtmpReader = reader\n\n    reader.connect().catch((err) => {\n      if (this.manualStop || this._suspended) return\n      this.logger?.warn('rtmp connect failed', {\n        meta: { error: errMsg(err), ...this.sourceMetaForLog() },\n      })\n      this.destroyRtmpReader()\n      this._status = 'error'\n      this.scheduleReconnect()\n    })\n  }\n\n  private destroyRtmpReader(): void {\n    if (this.rtmpReader) {\n      this.rtmpReader.destroy()\n      this.rtmpReader = null\n    }\n  }\n\n  /**\n   * Native RTSP reader: connect directly to the camera, receive raw RTP.\n   *\n   * - Preserves RTP framing (no Annex-B corruption on low-bitrate streams)\n   * - RTP passthrough to RTSP restreamer (zero re-packetization)\n   * - No ffprobe needed (codec detected from SDP DESCRIBE)\n   * - Lower latency (no ffmpeg buffering/probing)\n   * - Native G.711 audio decode (PCMU/PCMA → PCM f32le 16kHz)\n   *\n   * On error: reconnects with exponential backoff, emits placeholder frames.\n   */\n  /**\n   * Shared RTP-stream callback factory. Used by both the native RTSP\n   * reader and the RFC 4571 reader — they ingest RTP from different\n   * transports but the broker's downstream pipeline (depacketize →\n   * AnnexB → prebuffer / pipe / decoder / restream) is identical.\n   *\n   * `label` is the source-tag we attach to log messages so it's clear\n   * which transport surfaced the event. `onTerminate` is the\n   * transport-specific cleanup invoked from `onError` (the native client\n   * runs `destroyNativeClient`; the RFC 4571 reader closes the TCP\n   * socket and clears its handle).\n   */\n  private buildRtpStreamCallbacks(\n    label: string,\n    onTerminate: () => void,\n  ): import('../rtsp/rtsp-client.js').RtspClientCallbacks {\n    return {\n        onVideoTrack: (track, sdpText) => {\n          this.detectedCodec = track.codec\n          this.logger?.info('native: video track detected', {\n            meta: {\n              codec: track.codec,\n              payloadType: track.payloadType,\n              hasSdpParamSets: Boolean(track.codecParams?.parameterSets?.length),\n            },\n          })\n\n          // Capture codec parameter sets from SDP. Replay these to any\n          // already-registered encoded subscribers so WebRTC sessions\n          // that connected before the SDP arrived still get the VPS/\n          // SPS/PPS they need to initialise the HEVC decoder.\n          if (track.codecParams?.parameterSets?.length) {\n            this.sdpParameterSets = track.codecParams.parameterSets.map((p) => Buffer.from(p))\n            this.replaySdpParamSetsToEncodedSubscribers()\n            this.replaySdpParamSetsToRtpSubscribers()\n          }\n\n          // Set up RTP depacketizer for Annex-B output (prebuffer, pipe, decoder)\n          this.rtpDepacketizer = new RtpDepacketizer(track.codec, (depacketized) => {\n            this.handleDepacketizedNal(depacketized)\n          })\n\n          // Arm the watchdog: TCP is up + SDP parsed, so we now expect\n          // RTP. If nothing arrives, force a reconnect from scratch.\n          this.armFirstVideoWatchdog()\n\n          // Switch restreamer to RTP passthrough mode\n          this.rtspRestreamer.setCameraSdp(sdpText, track.codec, track.codecParams)\n        },\n\n        onVideoRtp: (rtpData) => {\n          if (this.stopping) return\n\n          this._status = 'streaming'\n          this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n          this.stopErrorPlaceholder()\n          // First RTP packet of this connection cycle — the watchdog\n          // can stand down. Cheap to call when already cancelled.\n          this.cancelFirstVideoWatchdog()\n          // Rolling activity watchdog: reset on every packet. Wi-Fi\n          // packet-loss stalls on the TCP socket (audio+video share\n          // the interleaved channel) show up here as the per-packet\n          // gap exceeding RTP_ACTIVITY_TIMEOUT_MS; the expiry tears\n          // the reader down so scheduleReconnect rebuilds the source.\n          this.kickRtpActivityWatchdog()\n\n          // Diagnostic milestones: same cadence as `audio: rtp packet\n          // received` so a wake-up cycle that brings audio back but\n          // never video is visible side-by-side in the log. Without\n          // this, on a battery cam wake-up there's no signal between\n          // \"rfc4571: streaming\" and \"decoder push\" — we can't tell\n          // if the camera is silent or only the consumer pipeline is\n          // misconfigured.\n          this.videoRtpSeen++\n          if (this.videoRtpSeen === 1 || this.videoRtpSeen === 50) {\n            this.logger?.info('video: rtp packet received', {\n              meta: { seq: this.videoRtpSeen, bytes: rtpData.length },\n            })\n          }\n\n          // Depacketize RTP → NALs for Annex-B consumers (prebuffer, pipe, decoder)\n          // The depacketizer callback (handleDepacketizedNal) runs synchronously\n          // within processPacket, so _lastNalKeyframe/_lastNalParamSet are set\n          // before we reach the pushRtpPassthrough call below.\n          this._lastNalKeyframe = false\n          this._lastNalParamSet = false\n          this.rtpDepacketizer?.processPacket(rtpData)\n\n          // Forward raw RTP to RTSP restreamer (zero re-packetization)\n          this.rtspRestreamer.pushRtpPassthrough(\n            rtpData,\n            this._lastNalKeyframe,\n            this._lastNalParamSet,\n          )\n\n          // Capture into the source-RTP pre-buffer (current GOP) BEFORE the\n          // fan-out, so a subscriber that bootstraps from it sees the current\n          // packet as the ring's tail. Keyframe flags were set by the\n          // depacketizer above.\n          this.captureRtpForPreBuffer(rtpData)\n\n          // Fan out source RTP to direct-RTP subscribers (WebRTC\n          // H.265 path uses this to feed a packet-level repacketizer).\n          if (this.rtpVideoCallbacks.size > 0) {\n            for (const [cb, entry] of this.rtpVideoCallbacks) {\n              try {\n                cb(rtpData)\n                entry.packetsDelivered += 1\n              } catch (err) {\n                this.logger?.warn('rtp video subscriber threw', {\n                  meta: { error: errMsg(err) },\n                })\n              }\n            }\n          }\n        },\n\n        onAudioTrack: (audioTrack) => {\n          const codecUpper = audioTrack.codec.toUpperCase()\n          this.audioCodec = codecUpper\n          const supportedNative = codecUpper === 'PCMU' || codecUpper === 'PCMA'\n\n          // Diagnostic: surface SDP fmtp keys + audio-codec cap availability\n          // so we can tell at a glance why the broker did or didn't pick the\n          // cap path for this camera.\n          this.logger?.info('audio: track detected', {\n            meta: {\n              codec: audioTrack.codec,\n              codecUpper,\n              clockRate: audioTrack.clockRate,\n              channels: audioTrack.channels,\n              fmtpKeys: Object.keys(audioTrack.fmtp),\n              hasConfigHex: 'config' in audioTrack.fmtp || 'CONFIG' in audioTrack.fmtp,\n              audioCodecApiAvailable: this.audioCodecApi !== null,\n              supportedNative,\n            },\n          })\n\n          let supported = supportedNative\n          let effectiveSampleRate = audioTrack.clockRate\n          let effectiveChannels = audioTrack.channels\n\n          if (supportedNative) {\n            this.audioRtpDecoder = new AudioRtpDecoder(codecUpper as AudioCodec, (chunk) => {\n              this.audioChunkPlane.fanout(chunk)\n            })\n            this.logger?.info('native: audio decoder initialized', {\n              meta: { codec: codecUpper, sampleRate: audioTrack.clockRate, channels: audioTrack.channels },\n            })\n          } else if (this.audioCodecApi) {\n            const capCfg = this.buildAudioCodecSessionConfig(audioTrack)\n            if (capCfg) {\n              effectiveSampleRate = capCfg.sourceSampleRate\n              effectiveChannels = capCfg.sourceChannels\n              supported = true\n              // Tear down any prior session before replacing — without\n              // this an RTSP reconnect leaks the previous AudioCodecSession\n              // (its poll timer keeps ticking and its upstream cap session\n              // lingers until idle reap).\n              if (this.audioCodecSession) {\n                const prior = this.audioCodecSession\n                this.audioCodecSession = null\n                void prior.close().catch((err) => {\n                  this.logger?.debug('audio-codec: prior RTSP session close failed', {\n                    meta: { error: errMsg(err) },\n                  })\n                })\n              }\n              this.audioCodecSession = new AudioCodecSession(\n                this.audioCodecApi,\n                capCfg,\n                this.buildAudioCodecEmitCallback(),\n                this.logger?.child('audio-codec'),\n              )\n              this.logger?.info('audio-codec: decode session armed', {\n                meta: {\n                  codec: capCfg.codec,\n                  sourceSampleRate: capCfg.sourceSampleRate,\n                  sourceChannels: capCfg.sourceChannels,\n                  rfc3640: capCfg.rfc3640 ? 'aac-hbr' : 'one-frame',\n                },\n              })\n            } else {\n              this.logger?.info('audio-codec: codec not handled by cap — audio skipped', {\n                meta: { codec: audioTrack.codec, sampleRate: audioTrack.clockRate, channels: audioTrack.channels },\n              })\n            }\n          } else {\n            this.logger?.info('native: audio codec not supported natively — audio skipped', {\n              meta: { codec: audioTrack.codec, sampleRate: audioTrack.clockRate, channels: audioTrack.channels },\n            })\n          }\n\n          this.audioTrackInfo = {\n            codec: codecUpper,\n            sampleRate: effectiveSampleRate,\n            channels: effectiveChannels,\n            supported,\n          }\n        },\n\n        onAudioRtp: (rtpData) => {\n          if (this.stopping) return\n          // Audio shares the same TCP interleaved socket as video — an\n          // audio packet is equally good proof the source is alive, so\n          // it also resets the rolling activity watchdog.\n          this.kickRtpActivityWatchdog()\n          this.audioRtpSeen++\n          if (this.audioRtpSeen === 1 || this.audioRtpSeen === 50) {\n            this.logger?.info('audio: rtp packet received', {\n              meta: {\n                seq: this.audioRtpSeen,\n                bytes: rtpData.length,\n                hasNativeDecoder: this.audioRtpDecoder !== null,\n                hasCodecSession: this.audioCodecSession !== null,\n              },\n            })\n          }\n          this.audioRtpDecoder?.processPacket(rtpData)\n          this.audioCodecSession?.processPacket(rtpData)\n\n          // Emit raw G.711 audio payload as EncodedPacket so WebRTC and\n          // other onEncodedData consumers can forward it without transcoding.\n          // ONLY for native codecs (PCMU/PCMA) — AAC/Opus raw RTP payloads\n          // are NOT G.711 and must go through audio-codec decode first.\n          const isNativeAudioCodec = this.audioCodec === 'PCMU' || this.audioCodec === 'PCMA'\n          if (isNativeAudioCodec && this.encodedCallbacks.size > 0 && rtpData.length > 12) {\n            const payload = rtpData.subarray(12)\n            if (payload.length > 0) {\n              // Extract RTP timestamp (bytes 4-7, big-endian) for PTS\n              const rtpTimestamp = rtpData.readUInt32BE(4)\n              const pts = Math.floor(rtpTimestamp / 8) // 8kHz → ms\n              const packet: EncodedPacket = {\n                type: 'audio',\n                data: payload,\n                pts,\n                dts: pts,\n                keyframe: false,\n                codec: this.audioCodec?.toLowerCase() ?? 'pcmu',\n              }\n              for (const [cb, entry] of this.encodedCallbacks) {\n                cb(packet)\n                entry.packetsDelivered += 1\n              }\n            }\n          }\n\n          // Forward raw audio RTP to RTSP restreamer for audio passthrough\n          this.rtspRestreamer.pushAudioRtpPassthrough(rtpData)\n        },\n\n        onPlaying: () => {\n          this.logger?.info(`${label}: streaming`)\n          this._status = 'streaming'\n        },\n\n        onError: (error) => {\n          if (this.manualStop) return\n          this.logger?.warn(`${label} error`, {\n            meta: {\n              error: error.message,\n              ...this.sourceMetaForLog(),\n            },\n          })\n          onTerminate()\n          this._status = 'error'\n          // rfc4571 sources are loopback to a publisher-owned TCP\n          // server. A read error after the lib's idle teardown means\n          // the cached URL is stale — ask the publisher to republish\n          // before the next reconnect tick.\n          if (this.source?.type === 'rfc4571') {\n            this.notifySourceRefresh()\n          }\n          this.scheduleReconnect()\n        },\n\n        onTeardown: () => {\n          onTerminate()\n        },\n    }\n  }\n\n  private startNativeRtspReader(source: StreamSource): void {\n    this.logger?.info('starting native RTSP reader')\n\n    const client = new NativeRtspClient(\n      {\n        url: source.url,\n        logger: this.logger?.child('rtsp-native'),\n      },\n      this.buildRtpStreamCallbacks('native RTSP', () => this.destroyNativeClient()),\n    )\n\n    this.nativeClient = client\n\n    client.connect().catch((err) => {\n      if (this.manualStop || this._suspended) return\n      this.logger?.warn('native RTSP connect failed', {\n        meta: { error: errMsg(err), ...this.sourceMetaForLog() },\n      })\n      this.destroyNativeClient()\n      this._status = 'error'\n      this.scheduleReconnect()\n    })\n  }\n\n  /**\n   * Handle a complete NAL from the RTP depacketizer.\n   * Converts to EncodedPacket and feeds all Annex-B consumers\n   * (prebuffer, pipe server, decoder, restreamers).\n   *\n   * Called synchronously from processPacket — sets _lastNalKeyframe and\n   * _lastNalParamSet so the onVideoRtp callback can forward them to the\n   * RTSP restreamer's pushRtpPassthrough.\n   */\n  /**\n   * Pending parameter-set NALs (SPS/PPS for H.264, VPS/SPS/PPS for H.265).\n   * Cameras emit these as separate RTP packets before the IDR/IRAP frame.\n   * We buffer them here and prepend to the next VCL NAL so downstream\n   * consumers (WebRTC, prebuffer, decoder) receive complete access units\n   * with a single `EncodedPacket` that contains SPS+PPS+IDR together.\n   */\n  private pendingParamNals: Buffer[] = []\n\n  /**\n   * handleDepacketizedNal receives individual NALs from the RTP depacketizer\n   * and emits them as EncodedPackets.\n   *\n   * Parameter-set NALs (SPS/PPS type 7/8 for H.264, VPS/SPS/PPS type 32/33/34\n   * for H.265) are buffered and prepended to the next VCL NAL (slice/IDR).\n   * This guarantees every keyframe EncodedPacket is a self-contained access\n   * unit that a decoder can initialise from — critical for WebRTC (Chrome\n   * can't decode an IDR without preceding SPS/PPS).\n   *\n   * The RTSP restreamer handles its own RTP passthrough separately and is\n   * not affected by this merging.\n   */\n  private handleDepacketizedNal(depacketized: DepacketizedNal): void {\n    const { nal, keyframe, parameterSet, timestamp } = depacketized\n    const codec = this.detectedCodec ?? 'h264'\n\n    // Track flags for the parent onVideoRtp call\n    if (keyframe) this._lastNalKeyframe = true\n    if (parameterSet) this._lastNalParamSet = true\n\n    // Detect NAL type\n    const isH264 = codec === 'h264'\n    const nalType = isH264 ? (nal[0]! & 0x1f) : ((nal[0]! & 0x7e) >> 1)\n\n    // H.264 parameter sets: SPS=7, PPS=8\n    // H.265 parameter sets: VPS=32, SPS=33, PPS=34\n    const isParamSet = isH264\n      ? (nalType === 7 || nalType === 8)\n      : (nalType === 32 || nalType === 33 || nalType === 34)\n\n    if (isParamSet) {\n      // Buffer parameter set — don't emit yet\n      const startCode = Buffer.from([0x00, 0x00, 0x00, 0x01])\n      this.pendingParamNals.push(Buffer.concat([startCode, nal]))\n      return\n    }\n\n    // VCL NAL (slice/IDR) — prepend any buffered parameter sets\n    const startCode = Buffer.from([0x00, 0x00, 0x00, 0x01])\n    const hadParamSets = this.pendingParamNals.length > 0\n    let annexBData: Buffer\n    if (hadParamSets) {\n      this.pendingParamNals.push(Buffer.concat([startCode, nal]))\n      annexBData = Buffer.concat(this.pendingParamNals)\n      this.pendingParamNals = []\n    } else {\n      annexBData = Buffer.concat([startCode, nal])\n    }\n\n    const pts = Math.floor(timestamp / 90) // 90kHz → ms\n\n    const packet: EncodedPacket = {\n      type: 'video',\n      data: annexBData,\n      pts,\n      dts: pts,\n      keyframe: keyframe || hadParamSets,\n      codec,\n    }\n\n    // Feed all Annex-B consumers (prebuffer, pipe, decoder).\n    // RTSP restream is handled separately via RTP passthrough in onVideoRtp.\n    this.pushEncodedPacket(packet)\n  }\n\n  /**\n   * Start emitting a pre-encoded H.264 keyframe at 1fps for an\n   * explicit `placeholder`-typed source (i.e. disabled / not-yet-\n   * configured cameras). The default reason is `'disabled'` because\n   * the only path that reaches this is \"operator turned the device\n   * off\" or \"device was created without credentials\"; callers who\n   * arrive here via a different upstream state can adjust reason via\n   * `setPlaceholderReason()` before calling.\n   *\n   * Frame is re-resolved on every tick from `placeholderReason` so\n   * upstream state transitions (`disabled` → `offline`, etc.) propagate\n   * within one second.\n   */\n  private startPlaceholderReader(_source: StreamSource): void {\n    this.logger?.debug('Starting placeholder')\n\n    if (this.placeholderReason === 'reconnecting') {\n      // Default the reason to `'disabled'` for the explicit placeholder\n      // source, but only if the caller hasn't already set something\n      // more specific.\n      this.placeholderReason = 'disabled'\n    }\n    this._status = 'streaming'\n\n    // Emit immediately\n    this.emitPlaceholderFrame()\n\n    // Re-emit at 1fps so consumers see an active stream\n    this.placeholderTimer = setInterval(() => {\n      if (this.manualStop || this.stopping) {\n        if (this.placeholderTimer) {\n          clearInterval(this.placeholderTimer)\n          this.placeholderTimer = undefined\n        }\n        return\n      }\n      this.emitPlaceholderFrame()\n    }, 1000)\n  }\n\n  /**\n   * Resolve the codec the placeholder frame should be encoded as.\n   * Picks the broker's currently-detected codec when a reader has\n   * already parsed the camera's SDP, otherwise falls back to the\n   * source's published codec, otherwise H.264. The choice MUST match\n   * the negotiated WebRTC session codec — sending an H.264 placeholder\n   * over an H.265 RTP track corrupts the decoder state on Chrome and\n   * leaves the viewer frozen on the last placeholder until the\n   * session is recreated. Mirroring the broker codec keeps the\n   * placeholder feed bit-stream-compatible across the same path the\n   * live camera uses.\n   */\n  private placeholderCodec(): PlaceholderCodec {\n    const codec = (this.detectedCodec ?? this.source?.videoCodec ?? '').toLowerCase()\n    if (codec === 'h265' || codec === 'hevc') return 'h265'\n    return 'h264'\n  }\n\n  /**\n   * Push the current placeholder frame (selected by `placeholderReason`\n   * + `placeholderCodec()`) to all encoded subscribers. No-op-safe to\n   * call from any timer.\n   */\n  private emitPlaceholderFrame(): void {\n    const codec: PlaceholderCodec = this.placeholderCodec()\n    const packet: EncodedPacket = {\n      type: 'video',\n      data: getPlaceholderFrame(this.placeholderReason, codec),\n      pts: Date.now(),\n      dts: Date.now(),\n      keyframe: true, // single-frame encode is always a keyframe\n      codec,\n      // Tag so consumers (notably the WebRTC encoded-data subscriber,\n      // which keeps a sticky SPS/PPS to prepend to keyframes lacking\n      // inline param sets) can drop the packet's parameter sets from\n      // any per-stream cache they maintain. Without this tag, a\n      // 1280×720 placeholder SPS gets prepended to the camera's real\n      // (different resolution) IDR and Chrome's decoder freezes on\n      // the last placeholder frame until the user manually restarts.\n      isPlaceholder: true,\n    }\n    this.pushEncodedPacket(packet)\n  }\n\n  // ── First-video watchdog ──────────────────────────────────────────\n\n  /**\n   * Arm the first-video watchdog. Called when the rfc4571 reader\n   * reports a working SDP (`onVideoTrack`), at which point the broker\n   * has a live TCP connection to the lib's loopback server and is\n   * waiting for the upstream Baichuan dedicated session to start\n   * pushing RTP. No-op when already armed (so duplicate `onVideoTrack`\n   * fires don't reset the timer).\n   */\n  private armFirstVideoWatchdog(): void {\n    if (this.firstVideoWatchdog) return\n    if (this.manualStop || this.stopping) return\n    this.firstVideoWatchdog = setTimeout(() => {\n      this.firstVideoWatchdog = undefined\n      if (this.manualStop || this.stopping) return\n      // Race-safe check: a packet may have arrived between the timer\n      // firing and this callback running. Be permissive.\n      if (this.videoRtpSeen > 0) return\n      this.logger?.warn(\n        'first-video watchdog: no video RTP after rfc4571 connect — forcing reconnect',\n        { meta: { timeoutMs: StreamBroker.FIRST_VIDEO_TIMEOUT_MS } },\n      )\n      // Reset videoRtpSeen so the next watchdog cycle gets a clean\n      // measurement, and `video: rtp packet received` re-fires for\n      // the fresh connection.\n      this.videoRtpSeen = 0\n      // Tear down the rfc4571 reader → lib detects TCP close →\n      // releases its dedicated session → next dial triggers a fresh\n      // session. This mirrors the user's \"click restart\" workaround.\n      this.destroyRfc4571Reader()\n      this._status = 'error'\n      // Treat this as a fresh recovery cycle rather than a continuation\n      // of an exponential-backoff sequence — the user noticed the\n      // stuck state after wake-up, the manual restart resets backoff,\n      // and we're emulating that.\n      this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n      this.lastRefreshRequestAt = 0\n      // Provider needs to know its source might be stale (the lib's\n      // server may have idle-torn-down between dial attempts).\n      this.notifySourceRefresh()\n      this.scheduleReconnect()\n    }, StreamBroker.FIRST_VIDEO_TIMEOUT_MS)\n  }\n\n  /** Cancel the first-video watchdog. Idempotent. */\n  private cancelFirstVideoWatchdog(): void {\n    if (!this.firstVideoWatchdog) return\n    clearTimeout(this.firstVideoWatchdog)\n    this.firstVideoWatchdog = undefined\n  }\n\n  // ── Rolling RTP activity watchdog ─────────────────────────────────\n\n  /**\n   * Kick the rolling source-RTP activity watchdog: cancel any prior arm\n   * and schedule a fresh expiry. Called per RTP packet from the native\n   * RTSP reader, the RFC 4571 reader, AND `pushVideoRtp` so push-rtp\n   * sources stay covered by the same arm/cancel lifecycle (the expiry\n   * path is reader-aware — it does nothing when no reader is alive).\n   *\n   * Cheap (clearTimeout + setTimeout); called at ~50-100Hz on a healthy\n   * stream — the overhead is negligible compared to the depacketize and\n   * fan-out work each packet already does.\n   */\n  private kickRtpActivityWatchdog(): void {\n    if (this.manualStop || this.stopping || this._suspended) return\n    if (this.rtpActivityWatchdog) {\n      clearTimeout(this.rtpActivityWatchdog)\n    }\n    this.rtpActivityWatchdog = setTimeout(() => {\n      this.onRtpActivityWatchdogExpired()\n    }, StreamBroker.RTP_ACTIVITY_TIMEOUT_MS)\n  }\n\n  /** Cancel the rolling RTP activity watchdog. Idempotent. */\n  private cancelRtpActivityWatchdog(): void {\n    if (!this.rtpActivityWatchdog) return\n    clearTimeout(this.rtpActivityWatchdog)\n    this.rtpActivityWatchdog = undefined\n  }\n\n  /**\n   * Activity watchdog expiry: source RTP went silent for longer than\n   * `RTP_ACTIVITY_TIMEOUT_MS`. The TCP socket is still open as far as\n   * the OS knows but no data has arrived — typical Wi-Fi packet-loss\n   * stall. Tear the reader down (\"activity watchdog fired: stream\n   * inactivity\") so `scheduleReconnect()` rebuilds the source.\n   *\n   * Only fires for sources that actually have a reader to tear down:\n   * native RTSP and RFC 4571. push / push-rtp / RTMP sources are\n   * provider-driven and don't go through this lifecycle — they have\n   * their own `PUSH_STALL_TIMEOUT_MS` path.\n   */\n  private onRtpActivityWatchdogExpired(): void {\n    this.rtpActivityWatchdog = undefined\n    this.rtpActivityWatchdogFiredCount++\n    if (this.manualStop || this.stopping || this._suspended) return\n\n    const meta = {\n      timeoutMs: StreamBroker.RTP_ACTIVITY_TIMEOUT_MS,\n      ...this.sourceMetaForLog(),\n    }\n    this.logger?.warn(\n      'source-RTP activity watchdog: no packets within window — forcing reconnect',\n      { meta },\n    )\n\n    if (this.nativeClient) {\n      if (this.streamingDebug) {\n        this.logger?.info('activity watchdog: destroying native RTSP reader', { meta })\n      }\n      this.destroyNativeClient()\n      this._status = 'error'\n      this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n      this.scheduleReconnect()\n      return\n    }\n\n    if (this.rfc4571Reader) {\n      if (this.streamingDebug) {\n        this.logger?.info('activity watchdog: destroying rfc4571 reader', { meta })\n      }\n      this.destroyRfc4571Reader()\n      this._status = 'error'\n      this.reconnectDelayMs = INITIAL_RECONNECT_DELAY_MS\n      // The lib's loopback may have idle-rotated — ask the publisher to\n      // refresh before the next dial picks up a stale URL.\n      this.notifySourceRefresh()\n      this.scheduleReconnect()\n      return\n    }\n\n    // No reader to tear down (push / push-rtp / RTMP) — log only.\n    // The provider owns the lifecycle on those paths.\n    if (this.streamingDebug) {\n      this.logger?.info(\n        'activity watchdog: no reader to tear down (push/RTMP source) — log only',\n        { meta },\n      )\n    }\n  }\n\n  // ── Error/idle placeholder ────────────────────────────────────────\n\n  /** Timer emitting error placeholder at 1fps while reconnecting. */\n  private errorPlaceholderTimer: ReturnType<typeof setInterval> | undefined\n\n  /**\n   * Reason currently displayed by the placeholder image. Drives\n   * `getPlaceholderFrame()` lookups so a sleeping cam shows\n   * \"SLEEPING\", an offline cam shows \"OFFLINE\", and so on. Default\n   * `'reconnecting'` matches the legacy single-frame behaviour.\n   *\n   * Mutated via `setPlaceholderReason()` from upstream surfaces (the\n   * Reolink provider transitions on sleep events; the broker manager\n   * may surface explicit disabled/offline states); the running\n   * `errorPlaceholderTimer` re-reads it on every tick so transitions\n   * appear within ~1 frame of the upstream signal.\n   */\n  private placeholderReason: PlaceholderKind = 'reconnecting'\n\n  /**\n   * Update the reason shown by the placeholder image. Idempotent —\n   * a no-op when the reason hasn't changed. Safe to call from\n   * upstream callers regardless of whether the placeholder timer is\n   * currently active; the next tick (or next `startErrorPlaceholder`\n   * invocation) picks up the new reason.\n   */\n  setPlaceholderReason(reason: PlaceholderKind): void {\n    if (this.placeholderReason === reason) return\n    this.placeholderReason = reason\n    // If the placeholder is actively emitting, push a fresh frame so\n    // the operator sees the new label immediately rather than waiting\n    // for the next 1Hz tick. Cheap (~5-9KB per emit) and matches user\n    // expectation when toggling state from the admin UI.\n    if (this.errorPlaceholderTimer) {\n      this.emitPlaceholderFrame()\n    }\n  }\n\n  private stopErrorPlaceholder(): void {\n    if (this.errorPlaceholderTimer) {\n      clearInterval(this.errorPlaceholderTimer)\n      this.errorPlaceholderTimer = undefined\n    }\n  }\n\n  private scheduleReconnect(): void {\n    if (this.manualStop || !this.source) {\n      return\n    }\n    // Hard rule: a broker never holds an open dial loop when no client\n    // is consuming the stream. Without demand, suspend instead of\n    // re-arming the reconnect timer; the next subscriber triggers\n    // `checkDemand()` → `start(this.source)` to dial again.\n    if (!this.hasDemand()) {\n      this._suspended = true\n      this._status = 'idle'\n      this.stopErrorPlaceholder()\n      if (this.reconnectTimer) {\n        clearTimeout(this.reconnectTimer)\n        this.reconnectTimer = undefined\n      }\n      this.logger?.info('Reconnect skipped — no demand, broker suspended')\n      return\n    }\n\n    // Dedup: a single connect failure fans out into two notifications\n    // (`socket.on('error')` → `onError` callback AND the rejected\n    // `client.connect()` promise → `.catch`). Both end up here. When a\n    // reconnect timer is already armed for the same failure window, do\n    // nothing — otherwise every doubled call doubles the live timers,\n    // and a few generations later one connect failure expands into\n    // hundreds of scheduled retries firing in the same tick.\n    if (this.reconnectTimer) {\n      return\n    }\n\n    // Placeholder pump is intentionally NOT started during reconnect.\n    // We tried both H.264 and H.265 cross-codec injection paths and\n    // they confused Chrome's HEVC decoder on the H.265 RTP path\n    // (placeholder SPS conflicting with the repacketizer's seeded\n    // camera SPS). Reconnect/wake-up windows now leave the existing\n    // last-frame on screen — the WebRTC viewer's UI overlay is the\n    // surface for reconnect feedback. Placeholders remain only for\n    // the explicit `placeholder` source type (disabled cameras),\n    // handled by `startPlaceholderReader`.\n\n    this.logger?.info(\n      'Reconnecting RTSP reader',\n      { meta: { delayMs: this.reconnectDelayMs } },\n    )\n\n    this.reconnectTimer = setTimeout(() => {\n      this.reconnectTimer = undefined\n      if (this.manualStop) return\n      // Re-resolve the source on every reconnect so an URL change\n      // landed while we were waiting (e.g. Reolink republished after\n      // an `ensureRfc4571Server` recreate) is observed.\n      const src = this.resolveCurrentSource()\n      if (!src) return\n      this._status = 'connecting'\n      if (src.type === 'rtmp') {\n        this.startRtmpReader(src)\n      } else if (src.type === 'rfc4571') {\n        this.startRfc4571Reader(src)\n      } else {\n        this.startRtspReader(src)\n      }\n    }, this.reconnectDelayMs)\n\n    this.reconnectDelayMs = Math.min(\n      this.reconnectDelayMs * 2,\n      MAX_RECONNECT_DELAY_MS,\n    )\n  }\n\n  /**\n   * Reset the push-mode stall detection timer.\n   * If no packet arrives within PUSH_STALL_TIMEOUT_MS, switch to error\n   * state and emit placeholder frames so RTSP clients don't freeze.\n   *\n   * Sources flagged `allowStall: true` (battery cameras that\n   * intentionally go silent between motion events) bypass the error\n   * placeholder — the broker logs at debug, marks the broker\n   * `suspended`, and waits silently for the next packet. The next\n   * `pushEncodedPacket` clears the suspension and resumes streaming.\n   */\n  private resetPushStallTimer(): void {\n    if (this.pushStallTimer) {\n      clearTimeout(this.pushStallTimer)\n    }\n    this.pushStallTimer = setTimeout(() => {\n      this.pushStallTimer = undefined\n      if (this.stopping || this.manualStop) return\n      if (this.source?.type !== 'push') return\n\n      if (this.source?.allowStall) {\n        this.logger?.debug('push source stall (allowStall=true) — suspending until next packet', {\n          meta: { timeoutMs: PUSH_STALL_TIMEOUT_MS },\n        })\n        this._suspended = true\n        return\n      }\n\n      this.logger?.warn('push source stalled — no data received', {\n        meta: { timeoutMs: PUSH_STALL_TIMEOUT_MS },\n      })\n      this._status = 'error'\n      // Placeholder intentionally not emitted — see scheduleReconnect\n      // for the rationale (cross-codec issues on H.265 RTP path).\n      // `_status='error'` is enough for stats consumers; UI overlay\n      // handles operator-facing feedback.\n    }, PUSH_STALL_TIMEOUT_MS)\n  }\n\n  /**\n   * Moleculer nodeID of a decoder provider hosting one of this broker's shm\n   * frame-plane sessions. `null` when the plane has no session. Used by the\n   * broker-manager's `rotateDecodersOnNode` to drop only the sessions\n   * affected by a per-agent event (e.g. an hwaccel override change).\n   */\n  getDecoderNodeId(): string | null {\n    return this.frameHandlePlane?.decoderNodeIds()[0] ?? null\n  }\n\n  /**\n   * Force this broker's shm frame-plane decoder sessions hosted on\n   * `agentNodeId` to rotate — destroy and rebuild so they re-pull the latest\n   * per-agent preferences (hwaccel backend, …). A no-op when the broker has\n   * no frame plane. Subscriptions survive the rotation.\n   */\n  async rotateDecoderSession(reason: string, agentNodeId?: string): Promise<void> {\n    const plane = this.frameHandlePlane\n    if (!plane) return\n    const target = agentNodeId\n      ?? (plane.decoderNodeIds()[0]?.split('/')[0])\n    if (!target) return\n    this.logger?.info('rotating shm decoder sessions', { meta: { reason, agentNodeId: target } })\n    await plane.rotate(target, reason)\n  }\n\n\n  private destroyNativeClient(): void {\n    if (this.nativeClient) {\n      this.nativeClient.destroy()\n      this.nativeClient = null\n    }\n    this.resetRtpPipeline()\n    // Reader is gone — stop the rolling activity watchdog so it can't\n    // fire against a destroyed client. The next reconnect re-arms it\n    // on the first RTP packet of the fresh stream.\n    this.cancelRtpActivityWatchdog()\n  }\n\n  /**\n   * RFC 4571 reader — pulls framed RTP from a local TCP server (e.g. the\n   * Reolink lib's `createRfc4571TcpServer`). Same downstream pipeline as\n   * the native RTSP reader: SDP arrives out-of-band on the StreamSource,\n   * RTP packets feed the same depacketizer, AnnexB output goes through\n   * the same prebuffer / pipe / decoder paths.\n   *\n   * `source.metadata.sdp` is required and must describe the video\n   * (and optionally audio) track the publisher is multiplexing on the\n   * single TCP connection.\n   */\n  private startRfc4571Reader(source: StreamSource): void {\n    this.logger?.info('starting RFC 4571 reader')\n\n    // Lazy publish — when the URL carries the `lazy:rfc4571:` sentinel\n    // the publisher hasn't materialized the upstream socket yet (it's\n    // deferred to first-consumer demand). Trigger a refresh; the\n    // publisher's `StreamBrokerOnRequestStreamSourceRefresh` listener\n    // is responsible for opening the real rfc4571 server and\n    // re-publishing with a real `tcp://...` URL — the reconnect tick\n    // resolves the fresh source via `sourceProvider`.\n    if (typeof source.url === 'string' && source.url.startsWith('lazy:rfc4571:')) {\n      this.logger?.info('rfc4571: lazy URL — requesting publisher to materialize upstream', {\n        meta: { url: source.url },\n      })\n      this._status = 'error'\n      this.notifySourceRefresh()\n      this.scheduleReconnect()\n      return\n    }\n\n    const sdp = typeof source.metadata?.['sdp'] === 'string' ? source.metadata['sdp'] as string : null\n    if (!sdp) {\n      // No SDP is the same situation as a stale lazy URL — the\n      // publisher needs to repopulate. Mirror the lazy-URL path: emit\n      // `notifySourceRefresh` so the publisher gets a chance to\n      // materialize, instead of failing silently.\n      this.logger?.warn('rfc4571: source missing metadata.sdp — requesting publisher refresh')\n      this._status = 'error'\n      this.notifySourceRefresh()\n      this.scheduleReconnect()\n      return\n    }\n\n    const reader = new Rfc4571Reader(\n      {\n        url: source.url,\n        sdp,\n        ...(this.logger ? { logger: this.logger.child('rfc4571') } : {}),\n      },\n      this.buildRtpStreamCallbacks('rfc4571', () => this.destroyRfc4571Reader()),\n    )\n\n    this.rfc4571Reader = reader\n\n    reader.connect().catch((err) => {\n      if (this.manualStop || this._suspended) return\n      this.logger?.warn('rfc4571 connect failed', {\n        meta: { error: errMsg(err), ...this.sourceMetaForLog() },\n      })\n      this.destroyRfc4571Reader()\n      this._status = 'error'\n      // Ask the publisher to refresh its loopback transport. The lib's\n      // RFC 4571 TCP server idle-tears-down after ~15s with no\n      // clients and may rebind to a different port on recreate, so a\n      // cached URL goes stale. The publisher's response (re-running\n      // `ensureRfc4571Server` + `publishCameraStream`) updates the\n      // cam-stream registry; the next reconnect picks the fresh URL\n      // up via `resolveCurrentSource()`. Deduped against the parallel\n      // onError emit by `notifySourceRefresh`.\n      this.notifySourceRefresh()\n      this.scheduleReconnect()\n    })\n  }\n\n  private destroyRfc4571Reader(): void {\n    if (this.rfc4571Reader) {\n      this.rfc4571Reader.destroy()\n      this.rfc4571Reader = null\n    }\n    this.resetRtpPipeline()\n    // The watchdog only makes sense while a reader is alive — drop it\n    // here so a teardown initiated outside `armFirstVideoWatchdog`'s\n    // own fire path doesn't leave a stale timer that would force a\n    // useless reconnect later.\n    this.cancelFirstVideoWatchdog()\n    // Same reasoning for the rolling activity watchdog: reader gone,\n    // no source to monitor. The next reconnect re-arms on first RTP.\n    this.cancelRtpActivityWatchdog()\n  }\n\n  /** Tear down the shared RTP/audio pipeline state used by both readers. */\n  private resetRtpPipeline(): void {\n    if (this.rtpDepacketizer) {\n      this.rtpDepacketizer.reset()\n      this.rtpDepacketizer = null\n    }\n    if (this.pushRtpFlagDepacketizer) {\n      this.pushRtpFlagDepacketizer.reset()\n      this.pushRtpFlagDepacketizer = null\n    }\n    // Drop the source-RTP pre-buffer with the reader — the GOP it holds\n    // belongs to the connection being torn down. Keeping it across an\n    // on-demand idle/redial leaks a stale GOP into the next session's\n    // instant-start replay (root cause of the device-97 cold-start stutter).\n    this.resetRtpPreBuffer()\n    if (this.audioRtpDecoder) {\n      this.audioRtpDecoder.flush()\n      this.audioRtpDecoder = null\n    }\n    if (this.audioCodecSession) {\n      const session = this.audioCodecSession\n      this.audioCodecSession = null\n      void session.close().catch((err) => {\n        this.logger?.warn('audio-codec: session close failed', {\n          meta: { error: errMsg(err) },\n        })\n      })\n    }\n  }\n\n  /**\n   * Map an SDP audio track to an `AudioCodecSession` config the\n   * audio-codec cap can consume. Returns `null` when the codec is\n   * outside the cap's catalogue (broker logs and skips audio in that\n   * case). Pure — relies only on parsed SDP fields.\n   */\n  /**\n   * Shared emit callback for `AudioCodecSession`. Decoded PCM is fanned\n   * out to audio subscribers and re-encoded as G.711 µ-law for WebRTC\n   * consumers. Used by both the RTSP onAudioTrack path and the push\n   * `pushAudioInfo` path.\n   */\n  private buildAudioCodecEmitCallback(): (chunk: import('@camstack/types').DecodedAudioChunk) => void {\n    return (chunk) => {\n      // The analyzer pipeline relies on the AudioCodecSession's ~500ms\n      // window to give the sound classifier a stable input length, so\n      // we hand the big chunk over to the audio-chunk plane unchanged.\n      this.audioChunkPlane.fanout(chunk)\n      if (this.encodedCallbacks.size > 0) {\n        const f32 = new Float32Array(chunk.data.buffer, chunk.data.byteOffset, chunk.data.byteLength / 4)\n        const pcm16 = new Int16Array(f32.length)\n        for (let i = 0; i < f32.length; i++) {\n          const s = Math.max(-1, Math.min(1, f32[i]!))\n          pcm16[i] = s < 0 ? s * 0x8000 : s * 0x7FFF\n        }\n        const ulaw = pcmToMulaw(pcm16)\n        const ratio = Math.max(1, Math.round(chunk.sampleRate / 8000))\n        const resampled = ratio > 1 ? downsampleUlaw(ulaw, ratio) : ulaw\n\n        // Split the 500ms-window mu-law buffer into RFC 3551-canonical\n        // 20ms RTP packets (160 bytes @ 8kHz µ-law). The analyzer\n        // pipeline can stomach the long window because it integrates\n        // over half-second classifier frames, but WebRTC senders\n        // can't: a single 4000-byte payload exceeds typical UDP MTU\n        // (frags drop on any single-fragment loss) AND arrives at the\n        // browser in 500ms-bursts the audio jitter buffer can't\n        // smooth out — net effect is \"no audio output on the player\".\n        // 20ms per packet is what every Reolink-style RTSP/rfc4571\n        // restream produces on the wire, so the WebRTC sender now\n        // looks identical to a passthrough G.711 source.\n        const PCMU_BYTES_PER_PACKET = 160 // 8kHz × 20ms × 1 byte/sample\n        const baseTs = chunk.timestamp\n        let offsetBytes = 0\n        // Stamp each sub-packet's `pts` with a monotonically advancing\n        // ms timestamp (20ms per slice). The WebRTC session uses its\n        // own RTP timestamp counter (`audioTimestampBase` in\n        // `session.ts:1032`), so this value mostly drives metadata —\n        // but keeping it monotonic lets any future RTP-aware consumer\n        // reconstruct the original timing without surprises.\n        let chunkOffsetMs = 0\n        while (offsetBytes < resampled.length) {\n          const end = Math.min(offsetBytes + PCMU_BYTES_PER_PACKET, resampled.length)\n          const slice = resampled.subarray(offsetBytes, end)\n          const packet: EncodedPacket = {\n            type: 'audio',\n            data: Buffer.from(slice),\n            pts: baseTs + chunkOffsetMs,\n            dts: baseTs + chunkOffsetMs,\n            keyframe: false,\n            codec: 'pcmu',\n          }\n          for (const [cb, entry] of this.encodedCallbacks) {\n            cb(packet)\n            entry.packetsDelivered += 1\n          }\n          offsetBytes = end\n          chunkOffsetMs += 20\n        }\n      }\n    }\n  }\n\n  /**\n   * Build an `AudioCodecSessionConfig` from `PushAudioInfo` declared by\n   * a push-source provider. Mirrors `buildAudioCodecSessionConfig`\n   * (which builds from an SDP audio track) for sources that don't\n   * carry SDP — Reolink Baichuan, Frigate, etc.\n   *\n   * Push sources don't carry rfc3640 framing — they emit one complete\n   * codec frame per packet — so the session is configured without\n   * `rfc3640` and the broker calls `pushRawFrame` instead of\n   * `processPacket`.\n   */\n  private buildAudioCodecSessionConfigFromPush(\n    info: import('@camstack/types').PushAudioInfo,\n  ): import('./audio-codec-session.js').AudioCodecSessionConfig | null {\n    const codec = info.codec.toLowerCase()\n    const tag = `broker:${this.deviceId}`\n    if (codec === 'aac') {\n      return {\n        codec: 'aac',\n        sourceSampleRate: info.sampleRate,\n        sourceChannels: info.channels,\n        ...(info.extraData ? { extraData: info.extraData } : {}),\n        targetSampleRate: 16000,\n        targetChannels: 1,\n        tag,\n      }\n    }\n    if (codec === 'opus') {\n      return {\n        codec: 'opus',\n        sourceSampleRate: info.sampleRate,\n        sourceChannels: info.channels,\n        targetSampleRate: 16000,\n        targetChannels: 1,\n        tag,\n      }\n    }\n    return null\n  }\n\n  private buildAudioCodecSessionConfig(\n    audioTrack: import('../rtsp/sdp-parser.js').SdpAudioTrack,\n  ): import('./audio-codec-session.js').AudioCodecSessionConfig | null {\n    const codecUpper = audioTrack.codec.toUpperCase()\n    const fmtp = audioTrack.fmtp\n    const tag = `broker:${this.deviceId}`\n\n    if (codecUpper === 'MPEG4-GENERIC' || codecUpper === 'AAC') {\n      let extraData: Uint8Array | undefined\n      let sourceSampleRate = audioTrack.clockRate\n      let sourceChannels = audioTrack.channels\n      const configHex = fmtp['config'] ?? fmtp['CONFIG']\n      if (configHex) {\n        try {\n          const asc = parseAudioSpecificConfig(configHex)\n          sourceSampleRate = asc.sampleRate\n          sourceChannels = asc.channelConfig > 0 ? asc.channelConfig : sourceChannels\n          extraData = hexToBytes(configHex)\n        } catch (err) {\n          this.logger?.warn('audio-codec: AudioSpecificConfig parse failed', {\n            meta: { configHex, error: errMsg(err) },\n          })\n        }\n      }\n      const sizeLength = parseFmtpInt(fmtp, 'sizelength', 13)\n      const indexLength = parseFmtpInt(fmtp, 'indexlength', 3)\n      const indexDeltaLength = parseFmtpInt(fmtp, 'indexdeltalength', indexLength)\n      return {\n        codec: 'aac',\n        sourceSampleRate,\n        sourceChannels,\n        ...(extraData ? { extraData } : {}),\n        targetSampleRate: 16000,\n        targetChannels: 1,\n        tag,\n        rfc3640: { sizeLength, indexLength, indexDeltaLength },\n      }\n    }\n\n    if (codecUpper === 'OPUS') {\n      return {\n        codec: 'opus',\n        sourceSampleRate: audioTrack.clockRate,\n        sourceChannels: audioTrack.channels,\n        targetSampleRate: 16000,\n        targetChannels: 1,\n        tag,\n      }\n    }\n\n    return null\n  }\n}\n\nfunction parseFmtpInt(\n  fmtp: Readonly<Record<string, string>>,\n  key: string,\n  fallback: number,\n): number {\n  const raw = fmtp[key] ?? fmtp[key.toLowerCase()] ?? fmtp[key.toUpperCase()]\n  if (raw === undefined) return fallback\n  const n = Number(raw)\n  return Number.isFinite(n) && n > 0 ? n : fallback\n}\n\nfunction hexToBytes(hex: string): Uint8Array {\n  const trimmed = hex.replace(/\\s+/g, '')\n  if (trimmed.length % 2 !== 0) {\n    throw new Error(`audio-codec: hex string has odd length (${trimmed.length})`)\n  }\n  const out = new Uint8Array(trimmed.length / 2)\n  for (let i = 0; i < out.length; i++) {\n    const byte = parseInt(trimmed.slice(i * 2, i * 2 + 2), 16)\n    if (Number.isNaN(byte)) {\n      throw new Error(`audio-codec: invalid hex at offset ${i * 2}`)\n    }\n    out[i] = byte\n  }\n  return out\n}\n","/**\n * Thin tRPC-backed proxy for the `audio-codec` capability — the broker\n * uses this to decode AAC / opus / pcm_alaw / pcm_mulaw audio into PCM\n * for downstream subscribers (audio-analyzer, WebRTC audio leg, …).\n *\n * The cap is a system singleton; one provider serves all brokers on the\n * cluster. Plumbing here is a pass-through to `ctx.api.audioCodec.*` —\n * the broker doesn't know whether the audio decode runs locally or on a\n * remote node, the cap router resolves it.\n */\nimport type {\n  AudioDecodeSessionConfig,\n  AudioPcmChunk,\n  AudioEncodeSessionConfig,\n  AudioEncodedChunk,\n  AudioCodecInfo,\n} from '@camstack/types'\n\n// Session-bound methods accept an optional `nodeId` so callers can pin\n// every call to the node that owns the session. The generated cap router\n// auto-routes any input field literally named `nodeId` to that node's\n// provider — sessions are process-local, so without sticky routing the\n// round-robin across nodes serves \"session not found\" on every other call.\n\nexport interface AudioCodecCapApi {\n  listSupportedCodecs(): Promise<readonly AudioCodecInfo[]>\n  canHandle(input: { codec: string; kind: 'decode' | 'encode' }): Promise<boolean>\n  createDecodeSession(input: AudioDecodeSessionConfig): Promise<{ sessionId: string; nodeId: string }>\n  createEncodeSession(input: AudioEncodeSessionConfig): Promise<{ sessionId: string; nodeId: string }>\n  closeSession(input: { sessionId: string; nodeId?: string }): Promise<void>\n  pushEncodedFrame(input: { sessionId: string; nodeId?: string; data: Uint8Array; pts?: number }): Promise<void>\n  pullPcm(input: { sessionId: string; nodeId?: string; maxCount: number }): Promise<readonly AudioPcmChunk[]>\n  pushPcm(input: { sessionId: string; nodeId?: string; data: Uint8Array; pts?: number }): Promise<void>\n  pullEncoded(input: { sessionId: string; nodeId?: string; maxCount: number }): Promise<readonly AudioEncodedChunk[]>\n  flushEncode(input: { sessionId: string; nodeId?: string }): Promise<readonly AudioEncodedChunk[]>\n}\n\ninterface AudioCodecRouter {\n  listSupportedCodecs: { query: (input?: void) => Promise<readonly AudioCodecInfo[]> }\n  canHandle: { query: (input: { codec: string; kind: 'decode' | 'encode' }) => Promise<boolean> }\n  createDecodeSession: { mutate: (input: AudioDecodeSessionConfig) => Promise<{ sessionId: string; nodeId: string }> }\n  createEncodeSession: { mutate: (input: AudioEncodeSessionConfig) => Promise<{ sessionId: string; nodeId: string }> }\n  closeSession: { mutate: (input: { sessionId: string; nodeId?: string }) => Promise<void> }\n  pushEncodedFrame: { mutate: (input: { sessionId: string; nodeId?: string; data: Uint8Array; pts?: number }) => Promise<void> }\n  pullPcm: { query: (input: { sessionId: string; nodeId?: string; maxCount: number }) => Promise<readonly AudioPcmChunk[]> }\n  pushPcm: { mutate: (input: { sessionId: string; nodeId?: string; data: Uint8Array; pts?: number }) => Promise<void> }\n  pullEncoded: { query: (input: { sessionId: string; nodeId?: string; maxCount: number }) => Promise<readonly AudioEncodedChunk[]> }\n  flushEncode: { mutate: (input: { sessionId: string; nodeId?: string }) => Promise<readonly AudioEncodedChunk[]> }\n}\n\n/**\n * Build a tRPC-backed `AudioCodecCapApi`. Returns `null` when the cap\n * isn't mounted (no audio-codec addon installed) so callers can fall\n * back to native PCMU/PCMA path or skip audio entirely.\n */\nexport function buildAudioCodecCapApi(api: unknown | null): AudioCodecCapApi | null {\n  if (!api) return null\n  const router = (api as Record<string, unknown>)['audioCodec'] as AudioCodecRouter | undefined\n  if (!router) return null\n\n  return {\n    listSupportedCodecs: () => router.listSupportedCodecs.query(),\n    canHandle: (input) => router.canHandle.query(input),\n    createDecodeSession: (input) => router.createDecodeSession.mutate(input),\n    createEncodeSession: (input) => router.createEncodeSession.mutate(input),\n    closeSession: (input) => router.closeSession.mutate(input),\n    pushEncodedFrame: (input) => router.pushEncodedFrame.mutate(input),\n    pullPcm: (input) => router.pullPcm.query(input),\n    pushPcm: (input) => router.pushPcm.mutate(input),\n    pullEncoded: (input) => router.pullEncoded.query(input),\n    flushEncode: (input) => router.flushEncode.mutate(input),\n  }\n}\n","import type { Socket } from 'node:net'\nimport { RTSP_METHODS, INTERLEAVED_MAGIC, MAX_WRITE_BUFFER, type RtpPacket, type RtspRequest, type TrackSetup } from './rtsp-types.js'\nimport { randomUUID } from 'node:crypto'\n\n/**\n * Handles one RTSP client connection.\n * Parses requests, responds to OPTIONS/DESCRIBE/SETUP/PLAY/TEARDOWN,\n * and sends RTP data via TCP interleaved framing.\n */\nexport class RtspSession {\n  private readonly sessionId: string\n  private readonly tracks: TrackSetup[] = []\n  private playing = false\n  private buffer = ''\n  private closed = false\n  private readonly sdp: string\n  private readonly onClose?: () => void\n  /** Epoch ms when the TCP connection was accepted. Used by listClients. */\n  private readonly connectedAt: number = Date.now()\n  /** Epoch ms of the last RTP packet sent successfully. 0 if none yet. */\n  private lastRtpAt = 0\n  /** Total video RTP bytes written to the socket. */\n  private bytesSent = 0\n\n  /** When true, this session receives video-only (no audio RTP). */\n  private readonly muted: boolean\n\n  constructor(\n    private readonly socket: Socket,\n    sdp: string,\n    onClose?: () => void,\n    muted?: boolean,\n  ) {\n    this.sdp = sdp\n    this.onClose = onClose\n    this.muted = muted ?? false\n    this.sessionId = randomUUID().replace(/-/g, '').slice(0, 16)\n\n    socket.on('data', (chunk: Buffer) => {\n      if (this.closed) return\n      this.buffer += chunk.toString('ascii')\n      this.processBuffer()\n    })\n\n    socket.on('close', () => { this.closed = true; this.onClose?.() })\n    socket.on('error', () => { this.closed = true; this.onClose?.() })\n  }\n\n  getSessionId(): string { return this.sessionId }\n  isPlaying(): boolean { return this.playing && !this.closed }\n  isMuted(): boolean { return this.muted }\n\n  /**\n   * Diagnostic snapshot consumed by `StreamBroker.listClients` (Phase 10).\n   * Gives operators the information needed to identify a specific RTSP\n   * session: where it's coming from, whether it's active, when it\n   * connected, and the last RTP activity timestamp.\n   */\n  getInfo(): {\n    readonly sessionId: string\n    readonly remoteAddr: string\n    readonly playing: boolean\n    readonly muted: boolean\n    readonly connectedAt: number\n    readonly lastRtpAt: number\n    readonly bytesSent: number\n  } {\n    const remoteIp = this.socket.remoteAddress ?? 'unknown'\n    const remotePort = this.socket.remotePort ?? 0\n    return {\n      sessionId: this.sessionId,\n      remoteAddr: `${remoteIp}:${remotePort}`,\n      playing: this.isPlaying(),\n      muted: this.muted,\n      connectedAt: this.connectedAt,\n      lastRtpAt: this.lastRtpAt,\n      bytesSent: this.bytesSent,\n    }\n  }\n\n  /** Send a video RTP packet via TCP interleaved framing. */\n  sendRtp(packet: RtpPacket): void {\n    if (!this.playing || this.closed) return\n    if (this.tracks.length === 0) return\n\n    // Drop slow clients\n    if (this.socket.writableLength > MAX_WRITE_BUFFER) {\n      this.close()\n      return\n    }\n\n    const channel = this.tracks[0]!.rtpChannel\n    const frame = Buffer.allocUnsafe(4 + packet.data.length)\n    frame[0] = INTERLEAVED_MAGIC\n    frame[1] = channel\n    frame.writeUInt16BE(packet.data.length, 2)\n    packet.data.copy(frame, 4)\n\n    this.socket.write(frame)\n    this.lastRtpAt = Date.now()\n    this.bytesSent += frame.length\n  }\n\n  /** Send an audio RTP packet via TCP interleaved framing on the audio track channel. */\n  sendAudioRtp(packet: RtpPacket): void {\n    if (!this.playing || this.closed || this.muted) return\n    if (this.tracks.length < 2) return // no audio track set up\n\n    if (this.socket.writableLength > MAX_WRITE_BUFFER) {\n      this.close()\n      return\n    }\n\n    const channel = this.tracks[1]!.rtpChannel\n    const frame = Buffer.allocUnsafe(4 + packet.data.length)\n    frame[0] = INTERLEAVED_MAGIC\n    frame[1] = channel\n    frame.writeUInt16BE(packet.data.length, 2)\n    packet.data.copy(frame, 4)\n\n    this.socket.write(frame)\n    this.lastRtpAt = Date.now()\n    this.bytesSent += frame.length\n  }\n\n  close(): void {\n    if (this.closed) return\n    this.closed = true\n    this.playing = false\n    try { this.socket.destroy() } catch { /* already closed */ }\n    this.onClose?.()\n  }\n\n  private processBuffer(): void {\n    while (true) {\n      // Skip any interleaved data from client (starts with $)\n      if (this.buffer.length > 0 && this.buffer.charCodeAt(0) === INTERLEAVED_MAGIC) {\n        if (this.buffer.length < 4) return\n        const len = (this.buffer.charCodeAt(2) << 8) | this.buffer.charCodeAt(3)\n        if (this.buffer.length < 4 + len) return\n        this.buffer = this.buffer.slice(4 + len)\n        continue\n      }\n\n      const endIdx = this.buffer.indexOf('\\r\\n\\r\\n')\n      if (endIdx < 0) return\n\n      const requestText = this.buffer.slice(0, endIdx)\n      this.buffer = this.buffer.slice(endIdx + 4)\n\n      const request = this.parseRequest(requestText)\n      if (request) this.handleRequest(request)\n    }\n  }\n\n  private parseRequest(text: string): RtspRequest | null {\n    const lines = text.split('\\r\\n')\n    const firstLine = lines[0]\n    if (!firstLine) return null\n\n    const parts = firstLine.split(' ')\n    if (parts.length < 3) return null\n\n    const headers = new Map<string, string>()\n    for (let i = 1; i < lines.length; i++) {\n      const colonIdx = lines[i]!.indexOf(':')\n      if (colonIdx < 0) continue\n      const key = lines[i]!.slice(0, colonIdx).trim().toLowerCase()\n      const value = lines[i]!.slice(colonIdx + 1).trim()\n      headers.set(key, value)\n    }\n\n    return {\n      method: parts[0]!,\n      uri: parts[1]!,\n      cseq: parseInt(headers.get('cseq') ?? '0', 10),\n      headers,\n    }\n  }\n\n  private handleRequest(req: RtspRequest): void {\n    switch (req.method) {\n      case 'OPTIONS': return this.handleOptions(req)\n      case 'DESCRIBE': return this.handleDescribe(req)\n      case 'SETUP': return this.handleSetup(req)\n      case 'PLAY': return this.handlePlay(req)\n      case 'TEARDOWN': return this.handleTeardown(req)\n      case 'GET_PARAMETER': return this.respond(req, 200, 'OK')\n      default: return this.respond(req, 405, 'Method Not Allowed')\n    }\n  }\n\n  private handleOptions(req: RtspRequest): void {\n    this.respond(req, 200, 'OK', {\n      'Public': RTSP_METHODS.join(', '),\n    })\n  }\n\n  private handleDescribe(req: RtspRequest): void {\n    this.respond(req, 200, 'OK', {\n      'Content-Type': 'application/sdp',\n      'Content-Length': String(Buffer.byteLength(this.sdp)),\n    }, this.sdp)\n  }\n\n  private handleSetup(req: RtspRequest): void {\n    const transport = req.headers.get('transport') ?? ''\n    const interleavedMatch = transport.match(/interleaved=(\\d+)-(\\d+)/)\n    const rtpChannel = interleavedMatch ? parseInt(interleavedMatch[1]!, 10) : 0\n    const rtcpChannel = interleavedMatch ? parseInt(interleavedMatch[2]!, 10) : 1\n\n    this.tracks.push({ trackId: this.tracks.length, rtpChannel, rtcpChannel })\n\n    this.respond(req, 200, 'OK', {\n      'Transport': `RTP/AVP/TCP;unicast;interleaved=${rtpChannel}-${rtcpChannel}`,\n      'Session': this.sessionId,\n    })\n  }\n\n  private handlePlay(req: RtspRequest): void {\n    this.playing = true\n    this.respond(req, 200, 'OK', {\n      'Session': this.sessionId,\n      'Range': 'npt=0.000-',\n    })\n  }\n\n  private handleTeardown(req: RtspRequest): void {\n    this.respond(req, 200, 'OK')\n    this.close()\n  }\n\n  private respond(req: RtspRequest, code: number, reason: string, headers?: Record<string, string>, body?: string): void {\n    if (this.closed) return\n\n    let response = `RTSP/1.0 ${code} ${reason}\\r\\nCSeq: ${req.cseq}\\r\\n`\n    if (headers) {\n      for (const [k, v] of Object.entries(headers)) {\n        response += `${k}: ${v}\\r\\n`\n      }\n    }\n    response += '\\r\\n'\n    if (body) response += body\n\n    this.socket.write(response)\n  }\n}\n","import * as net from 'node:net'\nimport { randomBytes } from 'node:crypto'\nimport { RtspSession } from './rtsp-session.js'\nimport { RtspRestreamer } from './rtsp-restreamer.js'\n\n/**\n * Closure that wakes the broker (subscribes a synthetic consumer to its\n * encoded-data fan-out, which trips `checkDemand` → upstream dial) and\n * returns a teardown handle. Released by the listen server when the RTSP\n * session ends, so the broker can idle-tear-down normally when no real\n * consumers remain. Without this, an RTSP client doing DESCRIBE on a\n * cold broker gets back a 200 with empty SDP (broker never dialed\n * because nothing subscribed) — `ffprobe`, recording sidecars, and any\n * external RTSP integration are effectively broken until something else\n * happens to be actively consuming the broker.\n */\nexport type BrokerWarmUp = () => () => void\n\n/**\n * Single TCP server for all RTSP restream connections.\n * Routes clients to the correct RtspRestreamer by a random token path\n * (not the predictable brokerId) for security.\n */\nexport class RtspListenServer {\n  private server: net.Server | null = null\n  private port = 0\n  /** token → restreamer */\n  private readonly restreamers = new Map<string, RtspRestreamer>()\n  /** brokerId → token (for lookup/regeneration) */\n  private readonly brokerTokens = new Map<string, string>()\n  /** token → brokerId (reverse map) */\n  private readonly tokenBrokers = new Map<string, string>()\n  /** brokerId → enabled (default true) */\n  private readonly brokerEnabled = new Map<string, boolean>()\n  /** brokerId → warm-up closure registered alongside the restreamer */\n  private readonly brokerWarmUps = new Map<string, BrokerWarmUp>()\n\n  /** Generate a random token for a stream path */\n  static generateToken(): string {\n    return randomBytes(16).toString('hex')\n  }\n\n  /**\n   * Register a restreamer with a token.\n   * If token is provided, use it (restored from persistence).\n   * If not, generate a new one.\n   * Returns the token used.\n   */\n  registerRestreamer(\n    brokerId: string,\n    restreamer: RtspRestreamer,\n    warmUp?: BrokerWarmUp,\n    existingToken?: string,\n  ): string {\n    // Unregister old token if broker is being re-registered\n    const oldToken = this.brokerTokens.get(brokerId)\n    if (oldToken) {\n      this.restreamers.delete(oldToken)\n      this.tokenBrokers.delete(oldToken)\n    }\n\n    const token = existingToken ?? RtspListenServer.generateToken()\n    this.restreamers.set(token, restreamer)\n    this.brokerTokens.set(brokerId, token)\n    this.tokenBrokers.set(token, brokerId)\n    if (warmUp) this.brokerWarmUps.set(brokerId, warmUp)\n    return token\n  }\n\n  unregisterRestreamer(brokerId: string): void {\n    const token = this.brokerTokens.get(brokerId)\n    if (token) {\n      this.restreamers.delete(token)\n      this.tokenBrokers.delete(token)\n    }\n    this.brokerTokens.delete(brokerId)\n    this.brokerWarmUps.delete(brokerId)\n  }\n\n  /** Regenerate a random token for a broker. Returns the new token. */\n  regenerateToken(brokerId: string): string | null {\n    const oldToken = this.brokerTokens.get(brokerId)\n    if (!oldToken) return null\n\n    const restreamer = this.restreamers.get(oldToken)\n    if (!restreamer) return null\n\n    // Remove old\n    this.restreamers.delete(oldToken)\n    this.tokenBrokers.delete(oldToken)\n\n    // Create new\n    const newToken = RtspListenServer.generateToken()\n    this.restreamers.set(newToken, restreamer)\n    this.brokerTokens.set(brokerId, newToken)\n    this.tokenBrokers.set(newToken, brokerId)\n    return newToken\n  }\n\n  /** Get the current token for a broker. */\n  getToken(brokerId: string): string | undefined {\n    return this.brokerTokens.get(brokerId)\n  }\n\n  /** Get all broker → token mappings. */\n  getAllTokens(): ReadonlyMap<string, string> {\n    return this.brokerTokens\n  }\n\n  /** Set the enabled state for a broker's restream. Default is true. */\n  setEnabled(brokerId: string, enabled: boolean): void {\n    this.brokerEnabled.set(brokerId, enabled)\n  }\n\n  /** Check if a broker's restream is enabled. Default true. */\n  isEnabled(brokerId: string): boolean {\n    return this.brokerEnabled.get(brokerId) ?? true\n  }\n\n  /** Get all broker → enabled mappings (only stores explicit overrides). */\n  getAllEnabled(): ReadonlyMap<string, boolean> {\n    return this.brokerEnabled\n  }\n\n  /** Load persisted enabled states (called at boot). */\n  loadEnabled(states: ReadonlyMap<string, boolean>): void {\n    for (const [k, v] of states) {\n      this.brokerEnabled.set(k, v)\n    }\n  }\n\n  getPort(): number { return this.port }\n\n  async start(port: number): Promise<void> {\n    return new Promise((resolve, reject) => {\n      this.server = net.createServer((socket) => this.handleConnection(socket))\n      this.server.on('error', reject)\n      this.server.listen(port, () => {\n        const addr = this.server!.address()\n        this.port = typeof addr === 'object' && addr ? addr.port : port\n        resolve()\n      })\n    })\n  }\n\n  async stop(): Promise<void> {\n    return new Promise((resolve) => {\n      if (!this.server) { resolve(); return }\n      this.server.close(() => resolve())\n      this.server = null\n    })\n  }\n\n  private handleConnection(socket: net.Socket): void {\n    let requestBuffer = ''\n\n    const onData = (chunk: Buffer) => {\n      requestBuffer += chunk.toString('ascii')\n\n      // Wait for first complete RTSP request to determine the stream\n      const endIdx = requestBuffer.indexOf('\\r\\n\\r\\n')\n      if (endIdx < 0) return\n\n      socket.removeListener('data', onData)\n      void this.dispatch(socket, requestBuffer)\n    }\n\n    socket.on('data', onData)\n    socket.on('error', () => {})\n  }\n\n  /**\n   * Async dispatch path for the first complete RTSP request. Resolves the\n   * restreamer, wakes the broker via its `warmUp` closure, awaits the SDP\n   * (broker only populates `restreamer.sdp` after dialing the upstream and\n   * parsing `onVideoTrack`), and then hands the socket off to a\n   * `RtspSession`. The warm-up handle stays alive for the entire socket\n   * lifetime so `checkDemand` on the broker keeps the dial up; we release\n   * it on `close` / `error`. Without this hop, `DESCRIBE` would race the\n   * cold broker and return 200 with empty SDP.\n   */\n  private async dispatch(socket: net.Socket, requestBuffer: string): Promise<void> {\n    // Parse the URI from the first line to determine stream path\n    const firstLine = requestBuffer.split('\\r\\n')[0] ?? ''\n    const uriMatch = firstLine.match(/\\s(rtsp:\\/\\/[^\\s]+)\\s/)\n    const uri = uriMatch ? uriMatch[1]! : ''\n\n    // Extract token path: rtsp://host:port/{token} → {token}\n    const pathMatch = uri.match(/rtsp:\\/\\/[^/]+(\\/.*?)(?:\\s|$)/)\n    const streamPath = pathMatch ? pathMatch[1]!.replace(/^\\//, '').replace(/\\/trackID=\\d+$/, '') : ''\n\n    // Check for /muted suffix — same token, audio-free variant\n    const isMuted = streamPath.endsWith('/muted')\n    const lookupPath = isMuted ? streamPath.slice(0, -'/muted'.length) : streamPath\n\n    const restreamer = this.restreamers.get(lookupPath)\n    if (!restreamer) {\n      socket.write(`RTSP/1.0 404 Not Found\\r\\nCSeq: 1\\r\\n\\r\\n`)\n      socket.destroy()\n      return\n    }\n\n    // Check if restream is enabled for this broker\n    const brokerId = this.tokenBrokers.get(lookupPath)\n    if (brokerId && !this.isEnabled(brokerId)) {\n      socket.write(`RTSP/1.0 403 Forbidden\\r\\nCSeq: 1\\r\\n\\r\\n`)\n      socket.destroy()\n      return\n    }\n\n    // Acquire the broker warm-up consumer (no-op when the broker is\n    // already streaming — `checkDemand` is monotonic in subscriber count).\n    // The release handle is held until the socket closes so the dial stays\n    // up for the entire RTSP session, never seeing a zero-consumer dip\n    // that would tear down the upstream pull mid-PLAY.\n    const warmUp = brokerId ? this.brokerWarmUps.get(brokerId) : undefined\n    const releaseWarmUp = warmUp ? warmUp() : null\n    let released = false\n    const releaseOnce = (): void => {\n      if (released) return\n      released = true\n      try { releaseWarmUp?.() } catch { /* swallow — listener teardown */ }\n    }\n    socket.once('close', releaseOnce)\n    socket.once('error', releaseOnce)\n\n    // Wait briefly for the broker to dial upstream and parse the SDP.\n    // RTSP DESCRIBE is the operation that fails worst with empty SDP, so\n    // we block here up to ~3 s. Bounded: a dead upstream camera shouldn't\n    // hang the connection forever — we fall through to whatever SDP the\n    // restreamer has (likely empty) and let the client get a normal 200\n    // with whatever it can parse.\n    const sdpReady = await this.waitForSdp(restreamer, isMuted, 3000)\n\n    const sdp = isMuted\n      ? (restreamer.getMutedSdp() ?? restreamer.getSdp() ?? '')\n      : (restreamer.getSdp() ?? '')\n    const session = new RtspSession(socket, sdp, () => {\n      restreamer.removeSession(session.getSessionId())\n      releaseOnce()\n    }, isMuted)\n    restreamer.addSession(session)\n\n    // Diagnostic-only: surface that we waited (or timed out) so an\n    // operator chasing \"ffprobe got empty SDP\" sees the gap.\n    void sdpReady\n\n    // Re-inject the buffered data so the session processes the first request\n    socket.unshift(Buffer.from(requestBuffer, 'ascii'))\n  }\n\n  /**\n   * Poll the restreamer's SDP cache every 50 ms up to `timeoutMs`. The\n   * cache is populated by the broker's `onVideoTrack` callback once the\n   * RTSP DESCRIBE/SETUP/PLAY against the upstream camera completes — for\n   * a cold broker that's typically ~200-1000 ms. Resolves `true` if the\n   * SDP becomes non-empty within the budget, `false` on timeout.\n   */\n  private async waitForSdp(\n    restreamer: RtspRestreamer,\n    isMuted: boolean,\n    timeoutMs: number,\n  ): Promise<boolean> {\n    const start = Date.now()\n    const have = (): boolean => {\n      const s = isMuted ? (restreamer.getMutedSdp() ?? restreamer.getSdp()) : restreamer.getSdp()\n      return !!s && s.length > 0\n    }\n    if (have()) return true\n    return new Promise<boolean>((resolve) => {\n      const interval = setInterval(() => {\n        if (have()) {\n          clearInterval(interval)\n          resolve(true)\n          return\n        }\n        if (Date.now() - start > timeoutMs) {\n          clearInterval(interval)\n          resolve(false)\n        }\n      }, 50)\n    })\n  }\n}\n","import type { IScopedLogger } from '@camstack/types'\nimport { parseProfileBrokerId } from '@camstack/types'\nimport type { RtspListenServer } from './rtsp-listen-server.js'\n\nexport type EnabledPersister = (states: ReadonlyMap<string, boolean>) => void\nexport type TokenPersister = (tokens: ReadonlyMap<string, string>) => void\n\n/**\n * Per-broker RTSP restream entry.\n *\n * Kept local to `addon-stream-broker` after the `rtsp-restream` capability\n * was absorbed into `stream-broker`. The shape is still validated at the\n * `stream-broker.cap.ts` boundary via `RtspRestreamEntrySchema`.\n */\nexport interface RtspRestreamEntry {\n  readonly brokerId: string\n  readonly url: string\n  readonly mutedUrl: string\n  readonly enabled: boolean\n}\n\n/**\n * Internal sub-object of `StreamBrokerManager` that implements the RTSP\n * restream bookkeeping. No longer registered as a separate capability —\n * accessed via passthrough methods on `StreamBrokerManager`.\n */\nexport class RtspRestreamProviderImpl {\n  private tokenPersister: TokenPersister | null = null\n  private enabledPersister: EnabledPersister | null = null\n\n  constructor(\n    private readonly server: RtspListenServer,\n    private readonly logger: IScopedLogger,\n  ) {}\n\n  // ── Persistence ────────────────────────────────────────────────────────\n\n  setTokenPersister(persister: TokenPersister): void {\n    this.tokenPersister = persister\n  }\n\n  setEnabledPersister(persister: EnabledPersister): void {\n    this.enabledPersister = persister\n  }\n\n  loadPersistedEnabled(states: ReadonlyMap<string, boolean>): void {\n    this.server.loadEnabled(states)\n  }\n\n  // ── IRtspRestreamProvider ──────────────────────────────────────────────\n\n  getRtspPort(): number {\n    return this.server.getPort()\n  }\n\n  getAllEntries(hostname?: string): readonly RtspRestreamEntry[] {\n    const port = this.server.getPort()\n    if (port === 0) return []\n    const host = hostname ?? '127.0.0.1'\n    const result: RtspRestreamEntry[] = []\n    for (const [brokerId, token] of this.server.getAllTokens()) {\n      // Skip profile-keyed alias tokens (see getEntriesForDevice).\n      if (parseProfileBrokerId(brokerId)) continue\n      result.push({\n        brokerId,\n        url: `rtsp://${host}:${port}/${token}`,\n        mutedUrl: `rtsp://${host}:${port}/${token}/muted`,\n        enabled: this.server.isEnabled(brokerId),\n      })\n    }\n    return result\n  }\n\n  /**\n   * Server-side filter by device — returns only entries whose\n   * `brokerId` is keyed `${deviceId}/${camStreamId}`. Mirrors\n   * `getAllEntries` but lets device-scoped consumers skip the\n   * cluster-wide scan + client-side filter that was the cost of\n   * routing through `getAllEntries`.\n   */\n  getEntriesForDevice(deviceId: number, hostname?: string): readonly RtspRestreamEntry[] {\n    const port = this.server.getPort()\n    if (port === 0) return []\n    const host = hostname ?? '127.0.0.1'\n    const prefix = `${deviceId}/`\n    const result: RtspRestreamEntry[] = []\n    for (const [brokerId, token] of this.server.getAllTokens()) {\n      if (!brokerId.startsWith(prefix)) continue\n      // Skip profile-keyed alias tokens (`${deviceId}/high|mid|low`) — they\n      // are an internal exporter convergence mechanism, NOT raw camStream\n      // restreams. The raw/live-view path must surface source camStreams only;\n      // profile entries are served via getProfileRtspEntries / getEntry.\n      if (parseProfileBrokerId(brokerId)) continue\n      result.push({\n        brokerId,\n        url: `rtsp://${host}:${port}/${token}`,\n        mutedUrl: `rtsp://${host}:${port}/${token}/muted`,\n        enabled: this.server.isEnabled(brokerId),\n      })\n    }\n    return result\n  }\n\n  getEntry(brokerId: string, hostname?: string): RtspRestreamEntry | null {\n    const token = this.server.getToken(brokerId)\n    if (!token) return null\n    const port = this.server.getPort()\n    const host = hostname ?? '127.0.0.1'\n    return {\n      brokerId,\n      url: `rtsp://${host}:${port}/${token}`,\n      mutedUrl: `rtsp://${host}:${port}/${token}/muted`,\n      enabled: this.server.isEnabled(brokerId),\n    }\n  }\n\n  regenerateToken(brokerId: string): string | null {\n    const token = this.server.regenerateToken(brokerId)\n    if (token) {\n      this.persistTokens()\n      this.logger.info('RTSP token regenerated', {\n        meta: { brokerId, tokenPrefix: token.slice(0, 8) },\n      })\n    }\n    return token\n  }\n\n  setEnabled(brokerId: string, enabled: boolean): void {\n    this.server.setEnabled(brokerId, enabled)\n    this.persistEnabled()\n    this.logger.info('RTSP restream state changed', { meta: { brokerId, enabled } })\n  }\n\n  isEnabled(brokerId: string): boolean {\n    return this.server.isEnabled(brokerId)\n  }\n\n  // ── Internal (called by StreamBrokerManager) ──────────────────────────\n\n  /** Persist tokens after registration/regeneration triggered by broker manager */\n  persistTokens(): void {\n    if (this.tokenPersister) {\n      this.tokenPersister(this.server.getAllTokens())\n    }\n  }\n\n  private persistEnabled(): void {\n    if (this.enabledPersister) {\n      this.enabledPersister(this.server.getAllEnabled())\n    }\n  }\n}\n","/**\n * Low-level ffmpeg argument fragments shared, byte-for-byte, across the\n * broker's ffmpeg sites (transcode egress, the WebRTC transcode feed, the\n * mjpeg decoder session). These are the pieces that were genuinely identical;\n * the higher-level builders differ (RTSP vs stdin input, hw vs sw encoders,\n * scale-filter shapes) and stay separate. Pure + side-effect-free.\n */\n\n/** Audio encoders addressable by both the preset model and the EncodeProfile model. */\nexport type FfmpegAudioCodec = 'opus' | 'aac' | 'pcmu' | 'pcma' | 'copy'\n\nexport interface AudioEncoderArgsOptions {\n  readonly bitrateKbps?: number\n  readonly sampleRateHz?: number\n  readonly channels?: number\n}\n\nconst AUDIO_ENCODER_BY_CODEC: Record<FfmpegAudioCodec, string> = {\n  opus: 'libopus',\n  aac: 'aac',\n  pcmu: 'pcm_mulaw',\n  pcma: 'pcm_alaw',\n  copy: 'copy',\n}\n\n/**\n * Build the audio-encode args `-c:a <enc> [-b:a Nk] [-ar N] [-ac N]`. Each knob\n * is emitted only when provided, in this fixed order — matching what every\n * broker site already emitted by hand.\n */\nexport function audioEncoderArgs(codec: FfmpegAudioCodec, opts: AudioEncoderArgsOptions = {}): string[] {\n  const args = ['-c:a', AUDIO_ENCODER_BY_CODEC[codec]]\n  if (opts.bitrateKbps !== undefined) args.push('-b:a', `${opts.bitrateKbps}k`)\n  if (opts.sampleRateHz !== undefined) args.push('-ar', String(opts.sampleRateHz))\n  if (opts.channels !== undefined) args.push('-ac', String(opts.channels))\n  return args\n}\n\n/** The `-hide_banner -loglevel <level>` preamble every broker ffmpeg site opens with. */\nexport function logBannerArgs(level: 'error' | 'warning' | 'info'): string[] {\n  return ['-hide_banner', '-loglevel', level]\n}\n","/**\n * Shared, side-effect-free ffmpeg invocation builder for the stream-broker.\n *\n * One place that embeds hardware acceleration (decode `-hwaccel` from the\n * platform probe + encoder selection), scale, thread count, audio, consumer\n * `outputArgs` and a discriminated output sink. Every broker ffmpeg site builds\n * its argument list here so hw-accel and the system-configured ffmpeg behaviour\n * stay consistent across the transcode pipeline (today) and the webrtc-feed /\n * decoder-session paths (follow-up).\n *\n * The builder returns the ARGUMENT list only — the binary path comes from the\n * `system-config` `ffmpeg.binaryPath` section and is passed to `spawn`\n * separately, keeping this function pure.\n */\nimport type { HardwareEncoders, HardwareEncoderId, AudioCodecTarget } from '@camstack/types'\nimport { audioEncoderArgs, logBannerArgs } from './ffmpeg-args-common.js'\n\n/** The `system-config` `ffmpeg.hwAccel` user preference. */\nexport type FfmpegHwAccelSetting =\n  | 'auto'\n  | 'none'\n  | 'videotoolbox'\n  | 'vaapi'\n  | 'qsv'\n  | 'cuda'\n\n/** The `prefer` value passed to `platformProbe.resolveHwAccel` (no `'auto'`). */\nexport type FfmpegHwAccelPrefer = Exclude<FfmpegHwAccelSetting, 'auto'>\n\n/** Resolved `system-config` `ffmpeg` section the broker spawns against. */\nexport interface FfmpegSystemConfig {\n  /** `ffmpeg.binaryPath` — passed to `spawn`, never into the arg list. */\n  readonly binaryPath: string\n  /** `ffmpeg.hwAccel` — drives decode `prefer` + encoder forcing. */\n  readonly hwAccel: FfmpegHwAccelSetting\n  /** `ffmpeg.threadCount` — `0` = auto (let ffmpeg decide). */\n  readonly threadCount: number\n}\n\n/** Where the transcoded output goes. */\nexport type FfmpegSink =\n  | { readonly kind: 'rtsp-listen'; readonly url: string }\n  | { readonly kind: 'stdout'; readonly container: 'h264' | 'hevc' | 'mpegts' }\n\n/** Audio codecs that can be muxed into an RTP sidecar output. */\nexport type AudioSidecarCodec = Extract<AudioCodecTarget, 'aac' | 'opus' | 'pcmu'>\n\n/** Narrow an audio target to a sidecar-capable codec, or `null` (copy/none). */\nexport function toAudioSidecarCodec(audio: AudioCodecTarget): AudioSidecarCodec | null {\n  return audio === 'aac' || audio === 'opus' || audio === 'pcmu' ? audio : null\n}\n\n/**\n * Optional second ffmpeg output that maps the source audio to its own RTP\n * stream (over UDP) alongside an elementary video stdout sink. ffmpeg writes\n * the authoritative SDP for this stream to `sdpFile` (`-sdp_file`), which the\n * egress grafts onto the restreamer's video-only SDP.\n */\nexport interface FfmpegAudioSidecar {\n  readonly codec: AudioSidecarCodec\n  /** `rtp://127.0.0.1:<port>` the egress listens on. */\n  readonly rtpUrl: string\n  /** Path ffmpeg writes the RTP stream's SDP to. */\n  readonly sdpFile: string\n}\n\nexport interface FfmpegTranscodeInvocation {\n  readonly sourceUrl: string\n  readonly targetCodec: 'H264' | 'H265'\n  readonly encoder: HardwareEncoderId\n  /** Decode hw-accel backend; `null`/`'none'` ⇒ software decode (no `-hwaccel`). */\n  readonly decodeHwAccel: string | null\n  readonly scale: { readonly width: number; readonly height: number } | null\n  readonly audio: AudioCodecTarget\n  /** `0` = auto (omit `-threads`). */\n  readonly threadCount: number\n  readonly outputArgs: readonly string[]\n  readonly sink: FfmpegSink\n  /**\n   * When set, append a second RTP audio output. Only meaningful with an\n   * elementary video sink (which is `-an`); the audio rides the sidecar.\n   */\n  readonly audioSidecar?: FfmpegAudioSidecar | null\n}\n\n/**\n * Map the `system-config` `ffmpeg.hwAccel` preference to the `prefer` argument\n * of `platformProbe.resolveHwAccel`: `'auto'` lets the resolver rank backends,\n * `'none'` forces software, a specific backend pins it.\n */\nexport function hwAccelSettingToPrefer(setting: FfmpegHwAccelSetting): FfmpegHwAccelPrefer | null {\n  if (setting === 'auto') return null\n  return setting\n}\n\n/**\n * Pick the decode `-hwaccel` backend, gated by what the configured ffmpeg\n * binary actually supports (`platformProbe.getHardwareDecodeAccels`).\n *\n * `preferred` entries are literal `-hwaccel` names (kernel-resolved), so they\n * intersect directly with the probed method list. Conservative by design:\n * - empty `preferred` → `null` (software decode);\n * - empty `supportedMethods` (probe miss / build reports none) → keep the top\n *   preference and let the egress's per-stream software fallback cover failures;\n * - otherwise return the first preference the build supports, or `null` when it\n *   supports none of them (positive evidence to skip a doomed hw-decode spawn).\n */\nexport function pickDecodeHwAccel(\n  preferred: readonly string[],\n  supportedMethods: readonly string[],\n): string | null {\n  if (preferred.length === 0) return null\n  if (supportedMethods.length === 0) return preferred[0] ?? null\n  const supported = new Set(supportedMethods.map((m) => m.toLowerCase()))\n  return preferred.find((name) => supported.has(name.toLowerCase())) ?? null\n}\n\n/**\n * Pick the video encoder, honouring the `hwAccel` preference: `'none'` forces\n * the software encoder; otherwise the probe's default (hw when available) is\n * used, falling back to software when no probe result is present.\n */\nexport function pickVideoEncoder(\n  target: 'H264' | 'H265',\n  encoders: HardwareEncoders | null,\n  hwAccel: FfmpegHwAccelSetting,\n): HardwareEncoderId {\n  const software: HardwareEncoderId = target === 'H264' ? 'libx264' : 'libx265'\n  if (hwAccel === 'none' || !encoders) return software\n  return target === 'H264' ? encoders.defaultH264 : encoders.defaultH265\n}\n\n/** Audio encoder args for a transcode target (preset values over the shared low-level builder). */\nexport function pickAudioEncoderArgs(audio: AudioCodecTarget): readonly string[] {\n  switch (audio) {\n    case 'aac':\n      return audioEncoderArgs('aac', { bitrateKbps: 128, sampleRateHz: 48000, channels: 2 })\n    case 'opus':\n      return audioEncoderArgs('opus', { bitrateKbps: 64, sampleRateHz: 48000, channels: 2 })\n    case 'pcmu':\n      return audioEncoderArgs('pcmu', { sampleRateHz: 8000, channels: 1 })\n    case 'copy':\n      return audioEncoderArgs('copy')\n    case 'none':\n      return ['-an']\n    default:\n      return audioEncoderArgs('aac', { bitrateKbps: 128 })\n  }\n}\n\n/**\n * Build the `-vf` filter args. A consumer `-vf` in `outputArgs` always wins (we\n * never emit a duplicate `-vf`); otherwise an explicit scale is applied;\n * otherwise no filter (native source resolution).\n */\nfunction buildVideoFilterArgs(\n  scale: { readonly width: number; readonly height: number } | null,\n  outputArgs: readonly string[],\n): string[] {\n  if (outputArgs.some((a) => a === '-vf')) return []\n  if (scale) return ['-vf', `scale=${scale.width}:${scale.height}`]\n  return []\n}\n\n/** Build the output-sink args (terminal — the output target is last). */\nfunction buildSinkArgs(sink: FfmpegSink): string[] {\n  if (sink.kind === 'stdout') {\n    return ['-f', sink.container, 'pipe:1']\n  }\n  return ['-f', 'rtsp', '-rtsp_transport', 'tcp', '-rtsp_flags', 'listen', sink.url]\n}\n\n/**\n * Build a second output that maps the source audio to RTP-over-UDP. `0:a:0?`\n * makes the audio stream optional — a source with no audio just skips it\n * instead of failing the whole ffmpeg invocation.\n */\nfunction buildAudioSidecarArgs(sidecar: FfmpegAudioSidecar): string[] {\n  return [\n    '-map', '0:a:0?',\n    ...pickAudioEncoderArgs(sidecar.codec),\n    '-f', 'rtp',\n    '-sdp_file', sidecar.sdpFile,\n    sidecar.rtpUrl,\n  ]\n}\n\n/**\n * Assemble the full ffmpeg argument list for a transcode invocation. Layout:\n *\n *   -hide_banner -loglevel warning\n *   [-hwaccel <decode>]            (before -i, when set and not 'none')\n *   -rtsp_transport tcp -i <src>\n *   [-vf scale=WxH]               (unless a consumer -vf is present)\n *   -c:v <encoder>\n *   [-threads <N>]                (when N > 0)\n *   <audio args>\n *   <consumer outputArgs verbatim>\n *   <sink args>                   (terminal: output target last)\n */\nexport function buildFfmpegArgs(inv: FfmpegTranscodeInvocation): string[] {\n  const decodeArgs =\n    inv.decodeHwAccel && inv.decodeHwAccel !== 'none' ? ['-hwaccel', inv.decodeHwAccel] : []\n  const threadArgs = inv.threadCount > 0 ? ['-threads', String(inv.threadCount)] : []\n\n  // An elementary stdout sink (raw h264/hevc) cannot carry audio — force `-an`\n  // so ffmpeg doesn't waste cycles encoding an audio stream it can't mux.\n  const isElementaryVideoSink =\n    inv.sink.kind === 'stdout' && (inv.sink.container === 'h264' || inv.sink.container === 'hevc')\n  const audioArgs = isElementaryVideoSink ? ['-an'] : pickAudioEncoderArgs(inv.audio)\n\n  return [\n    ...logBannerArgs('warning'),\n    ...decodeArgs,\n    '-rtsp_transport', 'tcp',\n    '-i', inv.sourceUrl,\n    ...buildVideoFilterArgs(inv.scale, inv.outputArgs),\n    '-c:v', inv.encoder,\n    ...threadArgs,\n    ...audioArgs,\n    ...inv.outputArgs,\n    ...buildSinkArgs(inv.sink),\n    ...(inv.audioSidecar ? buildAudioSidecarArgs(inv.audioSidecar) : []),\n  ]\n}\n","import { createSocket, type Socket } from 'node:dgram'\n\n/**\n * Receives RTP audio datagrams ffmpeg pushes to a local UDP port (the\n * transcode-egress audio sidecar). The egress forwards each datagram verbatim\n * to the restreamer's `pushAudioRtpPassthrough`.\n */\nexport interface AudioRtpReceiver {\n  /** Bind a UDP socket; resolves with the `rtp://` URL ffmpeg should target. */\n  start(onPacket: (rtp: Buffer) => void): Promise<{ rtpUrl: string }>\n  stop(): void\n}\n\nexport class UdpAudioRtpReceiver implements AudioRtpReceiver {\n  private readonly socket: Socket = createSocket('udp4')\n  private stopped = false\n\n  start(onPacket: (rtp: Buffer) => void): Promise<{ rtpUrl: string }> {\n    return new Promise<{ rtpUrl: string }>((resolve, reject) => {\n      this.socket.once('error', reject)\n      this.socket.on('message', (msg) => {\n        if (!this.stopped) onPacket(msg)\n      })\n      this.socket.bind(0, '127.0.0.1', () => {\n        const addr = this.socket.address()\n        resolve({ rtpUrl: `rtp://127.0.0.1:${addr.port}` })\n      })\n    })\n  }\n\n  stop(): void {\n    if (this.stopped) return\n    this.stopped = true\n    try {\n      this.socket.close()\n    } catch {\n      /* already closed */\n    }\n  }\n}\n","/**\n * TranscodeEgress — the egress half of `getStreamWithCodec`.\n *\n * The broker's RTSP listen server is restream-only (no ANNOUNCE/RECORD), so an\n * ffmpeg that *publishes* to it 404s. ffmpeg's `-rtsp_flags listen` server mode\n * is unreliable across builds (it attempts to connect, not listen). So we use\n * the pattern already proven in this codebase (the WebRTC transcode feed):\n * ffmpeg writes the transcoded **Annex-B elementary stream to stdout**, and we\n * push those bytes into a `RtspRestreamer` via `pushPacket` (which deframes +\n * repacketizes). The consumer gets back a normal broker token URL.\n *\n * VIDEO-ONLY: `pushPacket` handles video; audio in the restreamer travels a\n * separate RTP path, so the transcode egress drops audio (matching the WebRTC\n * transcode feed). Audio-in-transcode is a future enhancement.\n *\n * Decode hw-accel is tried first (big CPU win) but is fragile for some HEVC\n * streams (VideoToolbox can fail to decode). If ffmpeg dies before producing any\n * output and a decode backend was set, we respawn once with software decode.\n *\n * Every collaborator is injected so the wiring is unit-testable without real\n * ffmpeg or sockets.\n */\nimport type { ChildProcess, spawn as nodeSpawn } from 'node:child_process'\nimport { readFile } from 'node:fs/promises'\nimport { tmpdir } from 'node:os'\nimport { join } from 'node:path'\nimport { randomUUID } from 'node:crypto'\nimport { errMsg } from '@camstack/types'\nimport type { IScopedLogger, EncodedPacket } from '@camstack/types'\nimport {\n  buildFfmpegArgs,\n  type FfmpegTranscodeInvocation,\n  type FfmpegAudioSidecar,\n  type AudioSidecarCodec,\n} from './ffmpeg-invocation.js'\nimport { extractAudioMediaBlock } from '../rtsp/egress-audio-sdp.js'\nimport { UdpAudioRtpReceiver, type AudioRtpReceiver } from '../rtsp/udp-audio-rtp-receiver.js'\n\n/** Minimal surface of RtspRestreamer the egress feeds (Annex-B chunks + optional audio). */\nexport interface RtspRestreamerLike {\n  pushPacket(packet: EncodedPacket): void\n  /** Forward an RTP audio datagram (transcode-egress audio sidecar). */\n  pushAudioRtpPassthrough?(rtp: Buffer): void\n  /** Graft the egress audio track onto the push-mode SDP. */\n  setEgressAudioBlock?(block: string | null): void\n}\n\n/**\n * A registered egress channel: a restreamer already registered on the broker's\n * main listen server, its shareable consumer URL, and a release hook that\n * unregisters + destroys it. Created by production code that holds the concrete\n * `RtspRestreamer` type, so the egress only ever sees the structural `*Like`.\n */\nexport interface EgressChannel {\n  readonly restreamer: RtspRestreamerLike\n  readonly consumerUrl: string\n  release(): void\n}\n\nexport interface TranscodeEgressDeps {\n  readonly logger: IScopedLogger\n  readonly ffmpegBinaryPath: string\n  readonly spawnFn: typeof nodeSpawn\n  /** Create + register a restreamer for this brokerId on the main listen server. */\n  readonly acquireChannel: (brokerId: string) => EgressChannel\n  /** How long to wait for ffmpeg's first output byte before declaring it dead. */\n  readonly firstDataTimeoutMs?: number\n  /** Audio-sidecar collaborators — injected so the wiring is testable. */\n  readonly audioReceiverFactory?: () => AudioRtpReceiver\n  readonly readSdpFile?: (path: string) => Promise<string>\n  readonly makeSdpFilePath?: () => string\n}\n\nexport interface TranscodeEgressArgs {\n  readonly brokerId: string\n  /** Invocation minus the sink + audioSidecar — the egress fills those in. */\n  readonly invocation: Omit<FfmpegTranscodeInvocation, 'sink' | 'audioSidecar'>\n  /**\n   * Opt-in audio sidecar. When set, the egress maps the source audio to an RTP\n   * output (UDP) and forwards it to the restreamer, grafting ffmpeg's audio SDP\n   * onto the push-mode video SDP. Off by default — the default egress is the\n   * live-verified video-only path. NOT live-verified end-to-end (no A+V\n   * consumer drives it yet).\n   */\n  readonly audio?: { readonly codec: AudioSidecarCodec }\n}\n\nconst DEFAULT_FIRST_DATA_TIMEOUT_MS = 8_000\n\nexport class TranscodeEgress {\n  private child: ChildProcess | null = null\n  private channel: EgressChannel | null = null\n  private stopped = false\n  private startedAtMs = 0\n  private audioReceiver: AudioRtpReceiver | null = null\n  private audioSidecar: FfmpegAudioSidecar | null = null\n  private audioGrafted = false\n\n  constructor(\n    private readonly deps: TranscodeEgressDeps,\n    private readonly args: TranscodeEgressArgs,\n  ) {}\n\n  /** Spawn ffmpeg, stream its stdout into the restreamer. Returns the consumer URL. */\n  async start(): Promise<string> {\n    const channel = this.deps.acquireChannel(this.args.brokerId)\n    this.channel = channel\n    try {\n      // Only the audio path needs the async UDP bind before spawning; the\n      // default video-only egress spawns synchronously.\n      if (this.args.audio) {\n        await this.setupAudioSidecar(channel.restreamer)\n      }\n      await this.spawnWithDecodeFallback(channel.restreamer)\n    } catch (err) {\n      // ffmpeg never produced output — don't leak the process or channel.\n      await this.stop()\n      throw err\n    }\n    this.deps.logger.info('transcode egress ready', { meta: { brokerId: this.args.brokerId } })\n    return channel.consumerUrl\n  }\n\n  /** Tear down ffmpeg and release the channel registration. */\n  async stop(): Promise<void> {\n    if (this.stopped) return\n    this.stopped = true\n    this.killChild()\n    if (this.audioReceiver) {\n      this.audioReceiver.stop()\n      this.audioReceiver = null\n    }\n    if (this.channel) {\n      this.channel.release()\n      this.channel = null\n    }\n  }\n\n  /**\n   * Set up the audio sidecar (opt-in): bind a UDP RTP receiver that forwards\n   * each datagram to the restreamer, and allocate the `-sdp_file` path. ffmpeg\n   * is later told to map source audio to `rtpUrl`. No-op when audio isn't\n   * requested. NOT live-verified end-to-end (no A+V consumer drives it yet).\n   */\n  private async setupAudioSidecar(restreamer: RtspRestreamerLike): Promise<void> {\n    const audio = this.args.audio\n    if (!audio) return\n    const factory = this.deps.audioReceiverFactory ?? ((): AudioRtpReceiver => new UdpAudioRtpReceiver())\n    const makePath = this.deps.makeSdpFilePath\n      ?? ((): string => join(tmpdir(), `camstack-egress-${randomUUID()}.sdp`))\n    const receiver = factory()\n    this.audioReceiver = receiver\n    const { rtpUrl } = await receiver.start((rtp) => restreamer.pushAudioRtpPassthrough?.(rtp))\n    this.audioSidecar = { codec: audio.codec, rtpUrl, sdpFile: makePath() }\n    this.deps.logger.info('transcode egress: audio sidecar listening', {\n      meta: { brokerId: this.args.brokerId, rtpUrl, codec: audio.codec },\n    })\n  }\n\n  /**\n   * Read ffmpeg's `-sdp_file` (written when the RTP audio output opens), extract\n   * the `m=audio` block and graft it onto the restreamer SDP. Best-effort: a\n   * miss just means the consumer negotiates video only.\n   */\n  private async graftAudioSdp(restreamer: RtspRestreamerLike, sdpFile: string): Promise<void> {\n    const read = this.deps.readSdpFile ?? ((p: string): Promise<string> => readFile(p, 'utf8'))\n    try {\n      const sdpText = await read(sdpFile)\n      const block = extractAudioMediaBlock(sdpText)\n      if (block) {\n        restreamer.setEgressAudioBlock?.(block)\n        this.deps.logger.info('transcode egress: audio track grafted onto SDP', {\n          meta: { brokerId: this.args.brokerId },\n        })\n      } else {\n        this.deps.logger.warn('transcode egress: ffmpeg SDP had no audio media block', {\n          meta: { brokerId: this.args.brokerId },\n        })\n      }\n    } catch (err) {\n      this.deps.logger.warn('transcode egress: failed to graft audio SDP', {\n        meta: { brokerId: this.args.brokerId, error: errMsg(err) },\n      })\n    }\n  }\n\n  /**\n   * Try decode hw-accel first; if ffmpeg dies before producing output and a\n   * decode backend was requested, respawn once with software decode.\n   */\n  private async spawnWithDecodeFallback(restreamer: RtspRestreamerLike): Promise<void> {\n    const decode = this.args.invocation.decodeHwAccel\n    try {\n      await this.spawnAttempt(restreamer, decode)\n    } catch (err) {\n      if (this.stopped) throw err\n      if (decode && decode !== 'none') {\n        this.deps.logger.warn('transcode egress: hw decode produced no output — retrying software decode', {\n          meta: { brokerId: this.args.brokerId, decodeHwAccel: decode, error: errMsg(err) },\n        })\n        await this.spawnAttempt(restreamer, null)\n        return\n      }\n      throw err\n    }\n  }\n\n  /**\n   * Spawn ffmpeg writing the transcoded Annex-B stream to stdout, piping each\n   * chunk into the restreamer. Resolves on the FIRST output byte (the pipeline\n   * is live); rejects if ffmpeg exits / errors / is silent before then.\n   */\n  private spawnAttempt(restreamer: RtspRestreamerLike, decodeHwAccel: string | null): Promise<void> {\n    const targetCodec = this.args.invocation.targetCodec\n    const container: 'h264' | 'hevc' = targetCodec === 'H264' ? 'h264' : 'hevc'\n    const packetCodec = targetCodec === 'H264' ? 'h264' : 'h265'\n    const args = buildFfmpegArgs({\n      ...this.args.invocation,\n      decodeHwAccel,\n      sink: { kind: 'stdout', container },\n      audioSidecar: this.audioSidecar,\n    })\n\n    return new Promise<void>((resolve, reject) => {\n      const child = this.deps.spawnFn(this.deps.ffmpegBinaryPath, args, {\n        stdio: ['ignore', 'pipe', 'pipe'],\n      })\n      this.child = child\n      this.startedAtMs = Date.now()\n      let settled = false\n\n      const timeoutMs = this.deps.firstDataTimeoutMs ?? DEFAULT_FIRST_DATA_TIMEOUT_MS\n      const timer = setTimeout(() => {\n        if (settled) return\n        settled = true\n        this.killChild()\n        reject(new Error(`transcode egress: no output within ${timeoutMs}ms`))\n      }, timeoutMs)\n\n      child.stdout?.on('data', (chunk: Buffer) => {\n        if (!settled) {\n          settled = true\n          clearTimeout(timer)\n          resolve()\n          // Pipeline is live: graft the audio SDP once (ffmpeg has opened the\n          // RTP output and written -sdp_file by now).\n          if (this.audioSidecar && !this.audioGrafted) {\n            this.audioGrafted = true\n            void this.graftAudioSdp(restreamer, this.audioSidecar.sdpFile)\n          }\n        }\n        const ptsMs = Date.now() - this.startedAtMs\n        restreamer.pushPacket({\n          type: 'video',\n          data: chunk,\n          pts: ptsMs,\n          dts: ptsMs,\n          keyframe: false, // the restreamer's deframer detects real keyframes\n          codec: packetCodec,\n        })\n      })\n\n      child.stderr?.setEncoding('utf8')\n      child.stderr?.on('data', (line: string) => {\n        this.deps.logger.debug('transcode egress ffmpeg', {\n          meta: { brokerId: this.args.brokerId, line: line.trim() },\n        })\n      })\n\n      child.once('error', (err: Error) => {\n        if (settled) return\n        settled = true\n        clearTimeout(timer)\n        reject(err)\n      })\n\n      child.once('exit', (code, signal) => {\n        if (settled) return // exited after going live → data just stops\n        settled = true\n        clearTimeout(timer)\n        reject(new Error(`transcode egress: ffmpeg exited before output (code=${code} signal=${signal})`))\n      })\n    })\n  }\n\n  private killChild(): void {\n    const child = this.child\n    this.child = null\n    if (child && !child.killed) {\n      try {\n        child.kill('SIGTERM')\n      } catch (err) {\n        this.deps.logger.warn('transcode egress ffmpeg kill error', {\n          meta: { brokerId: this.args.brokerId, error: errMsg(err) },\n        })\n      }\n    }\n  }\n}\n","/**\n * Transcode-pipeline manager — backs `streamBroker.getStreamWithCodec`.\n *\n * Responsibilities:\n *   1. Source-select: profile-driven only. With `profile` the assigned\n *      camStreamId IS the source; without it the first published stream.\n *   2. Passthrough (no ffmpeg, `-c copy`): when `video:'copy'`, OR when the\n *      source codec already matches the requested codec AND `audio:'copy'`\n *      AND no `outputArgs`. Returns the broker restream URL.\n *   3. Spawn + refcount transcode pipelines otherwise: one `TranscodeEgress`\n *      per unique (deviceId, source, videoCodec, audioCodec, resolution,\n *      outputArgs) tuple, multi-consumer via refcount. Stops the egress N\n *      seconds after the refcount drops to 0 (`releaseGraceMs`).\n *   4. Pick the right encoder + decode hw-accel: consults\n *      `platform-probe.getHardwareEncoders` / `resolveHwAccel` (typed, cached)\n *      and the system `ffmpeg` config (binaryPath / hwAccel / threadCount),\n *      falling back to `libx264` / `libx265` and software decode.\n *\n * Egress: ffmpeg runs as an RTSP *server* (`-rtsp_flags listen`) on an internal\n * The transcoded Annex-B stream is read from ffmpeg's stdout and pushed into a\n * restreamer registered under a consumer token (see `TranscodeEgress`). The\n * broker's RTSP server is restream-only, so the old \"ffmpeg publishes to the\n * broker\" path 404'd, and ffmpeg's RTSP listen mode is unreliable — the stdout\n * pipe is the pattern proven by the WebRTC transcode feed.\n */\nimport { spawn } from 'node:child_process'\nimport { randomUUID } from 'node:crypto'\nimport { errMsg, makeSourceBrokerId } from '@camstack/types'\nimport type {\n  CameraStream,\n  CamProfile,\n  IScopedLogger,\n  AddonApi,\n  GetStreamWithCodecInput,\n  RtpSource,\n  AudioCodecTarget,\n  HardwareEncoders,\n} from '@camstack/types'\nimport {\n  pickVideoEncoder,\n  pickDecodeHwAccel,\n  toAudioSidecarCodec,\n  hwAccelSettingToPrefer,\n  type FfmpegSystemConfig,\n  type FfmpegHwAccelPrefer,\n} from './ffmpeg-invocation.js'\nimport {\n  TranscodeEgress,\n  type EgressChannel,\n} from './transcode-egress.js'\n\ntype CameraStreamLookup = (deviceId: number) => readonly CameraStream[]\ntype RtspEntryLookup = (brokerId: string) => { url: string } | null\n/** Resolve the camStreamId assigned to a device's profile slot, or null. */\ntype ProfileSourceLookup = (deviceId: number, profile: CamProfile) => string | null\n/** Create + register an egress channel (restreamer + token URL) for a brokerId. */\ntype AcquireChannelFn = (brokerId: string) => EgressChannel\n\ninterface PipelineEntry {\n  readonly key: string\n  readonly source: RtpSource\n  readonly egress: TranscodeEgress | null\n  refcount: number\n  releaseTimer: NodeJS.Timeout | null\n}\n\nconst DEFAULT_FFMPEG_CONFIG: FfmpegSystemConfig = {\n  binaryPath: 'ffmpeg',\n  hwAccel: 'auto',\n  threadCount: 0,\n}\n\nexport interface TranscodePipelineManagerOptions {\n  readonly logger: IScopedLogger\n  readonly api: AddonApi | null\n  readonly cameraStreamLookup: CameraStreamLookup\n  readonly rtspEntryLookup: RtspEntryLookup\n  /** Resolve a device profile to its assigned source camStreamId. When\n   *  omitted, `getStreamWithCodec` cannot honour a `profile` target. */\n  readonly resolveProfileSource?: ProfileSourceLookup\n  readonly releaseGraceMs?: number\n  /** Live accessor for the system `ffmpeg` config (binaryPath / hwAccel / threadCount). */\n  readonly ffmpegConfig?: () => FfmpegSystemConfig\n  /** Injectable for tests; defaults to `node:child_process` spawn. */\n  readonly spawnFn?: typeof import('node:child_process').spawn\n  /** Create + register the egress restreamer on the broker's main listen\n   *  server. Required for the transcode path (the manager is decoupled from\n   *  the listen server); when omitted, a transcode request throws. */\n  readonly acquireChannel?: AcquireChannelFn\n}\n\nfunction normaliseCodec(codec: string | undefined): 'H264' | 'H265' | null {\n  if (!codec) return null\n  const c = codec.toLowerCase()\n  if (c === 'h264' || c === 'avc' || c === 'avc1') return 'H264'\n  if (c === 'h265' || c === 'hevc' || c === 'hvc1' || c === 'hev1') return 'H265'\n  return null\n}\n\nfunction videoTargetToCodec(v: 'h264' | 'h265'): 'H264' | 'H265' {\n  return v === 'h264' ? 'H264' : 'H265'\n}\n\nfunction pipelineKeyFor(\n  deviceId: number,\n  camStreamId: string,\n  videoCodec: 'H264' | 'H265' | 'copy',\n  audioCodec: AudioCodecTarget,\n  outResolution: { width: number; height: number },\n  outputArgs: readonly string[],\n  audioEgress: boolean,\n): string {\n  const ae = audioEgress ? '|ae=1' : ''\n  const base = `${deviceId}|src=${camStreamId}|v=${videoCodec}|a=${audioCodec}|r=${outResolution.width}x${outResolution.height}${ae}`\n  return outputArgs.length > 0 ? `${base}|o=${JSON.stringify(outputArgs)}` : base\n}\n\nexport class TranscodePipelineManager {\n  private readonly entries = new Map<string, PipelineEntry>()\n  /** In-flight transcode starts, keyed by pipeline key — single-flight the\n   *  async egress start so concurrent acquires share one egress. */\n  private readonly pending = new Map<string, Promise<RtpSource>>()\n  private readonly logger: IScopedLogger\n  private readonly api: AddonApi | null\n  private readonly cameraStreamLookup: CameraStreamLookup\n  private readonly rtspEntryLookup: RtspEntryLookup\n  private readonly resolveProfileSource: ProfileSourceLookup | null\n  private readonly releaseGraceMs: number\n  private readonly ffmpegConfig: () => FfmpegSystemConfig\n  private readonly spawnFn: typeof import('node:child_process').spawn\n  private readonly acquireChannel: AcquireChannelFn | null\n\n  constructor(opts: TranscodePipelineManagerOptions) {\n    this.logger = opts.logger\n    this.api = opts.api\n    this.cameraStreamLookup = opts.cameraStreamLookup\n    this.rtspEntryLookup = opts.rtspEntryLookup\n    this.resolveProfileSource = opts.resolveProfileSource ?? null\n    this.releaseGraceMs = opts.releaseGraceMs ?? 5_000\n    this.ffmpegConfig = opts.ffmpegConfig ?? (() => DEFAULT_FFMPEG_CONFIG)\n    this.spawnFn = opts.spawnFn ?? spawn\n    this.acquireChannel = opts.acquireChannel ?? null\n  }\n\n  async acquire(input: GetStreamWithCodecInput): Promise<RtpSource> {\n    const wantedAudio: AudioCodecTarget = input.audio ?? 'aac'\n    const outputArgs = input.outputArgs ?? []\n    const audioEgress = input.audioEgress === true\n\n    // Source selection precedence:\n    //   - `profile` → the profile's assigned camStreamId IS the source.\n    //   - else `targetResolution` → the published source closest by area.\n    //   - else → the first published stream.\n    const sourceStream = input.profile !== undefined\n      ? this.resolveTargetedStream(input.deviceId, input.profile)\n      : input.targetResolution !== undefined\n        ? (this.pickClosestSource(input.deviceId, input.targetResolution) ?? this.firstPublished(input.deviceId))\n        : this.firstPublished(input.deviceId)\n    if (input.profile === undefined && !sourceStream) {\n      throw new Error(`getStreamWithCodec: no published stream for device ${input.deviceId}`)\n    }\n    if (!sourceStream || !sourceStream.url) {\n      throw new Error(`getStreamWithCodec: no assigned source for device ${input.deviceId}`)\n    }\n\n    const sourceCodec = normaliseCodec(sourceStream.codec)\n    // Effective output resolution: explicit target wins, else native source.\n    const outResolution = input.targetResolution ?? sourceStream.resolution ?? { width: 1280, height: 720 }\n\n    // A `copy` cannot be rescaled, so `targetResolution` is ignored for copy\n    // SCALING (still honoured for source selection above). The codec-match\n    // passthrough branch additionally requires that no rescale is needed.\n    const sourceRes = sourceStream.resolution\n    const resMatchesTarget =\n      input.targetResolution === undefined ||\n      (sourceRes !== undefined &&\n        sourceRes.width === input.targetResolution.width &&\n        sourceRes.height === input.targetResolution.height)\n\n    // Passthrough (no ffmpeg): explicit `copy`, OR the source codec already\n    // matches the requested codec AND audio is copy AND no extra output args\n    // AND no rescale is needed.\n    const wantsPassthrough =\n      input.video === 'copy' ||\n      (sourceCodec !== null &&\n        videoTargetToCodec(input.video) === sourceCodec &&\n        wantedAudio === 'copy' &&\n        outputArgs.length === 0 &&\n        resMatchesTarget)\n    if (wantsPassthrough) {\n      if (sourceCodec === null) {\n        this.logger.warn('copy-passthrough requested but source codec is unknown — reporting H264', {\n          meta: { deviceId: input.deviceId, camStreamId: sourceStream.camStreamId },\n        })\n      }\n      const passVideoCodec: 'H264' | 'H265' = sourceCodec ?? 'H264'\n      const url = this.resolveSourceUrl(input.deviceId, sourceStream.camStreamId)\n      const key = pipelineKeyFor(\n        input.deviceId,\n        sourceStream.camStreamId,\n        'copy',\n        wantedAudio,\n        outResolution,\n        outputArgs,\n        false,\n      )\n      return this.shareOrCreatePassthrough(key, () => ({\n        url,\n        videoCodec: passVideoCodec,\n        audioCodec: 'passthrough',\n        resolution: outResolution,\n        transcoded: false,\n        encoder: 'copy',\n        pipelineKey: key,\n      }))\n    }\n\n    const targetCodec: 'H264' | 'H265' = videoTargetToCodec(\n      input.video === 'copy' ? 'h264' : input.video,\n    )\n    const key = pipelineKeyFor(\n      input.deviceId,\n      sourceStream.camStreamId,\n      targetCodec,\n      wantedAudio,\n      outResolution,\n      outputArgs,\n      audioEgress,\n    )\n\n    const existing = this.entries.get(key)\n    if (existing) {\n      return this.shareEntry(existing).source\n    }\n\n    // Single-flight: a concurrent acquire for the same key shares the in-flight\n    // egress start rather than spawning a second ffmpeg.\n    const inflight = this.pending.get(key)\n    if (inflight) {\n      await inflight\n      const after = this.entries.get(key)\n      if (after) return this.shareEntry(after).source\n      // creation failed — fall through and retry below.\n    }\n\n    const start = this.startTranscode({\n      key,\n      deviceId: input.deviceId,\n      camStreamId: sourceStream.camStreamId,\n      targetCodec,\n      audio: wantedAudio,\n      audioEgress,\n      scale: input.targetResolution ?? null,\n      outResolution,\n      outputArgs,\n    })\n    this.pending.set(key, start)\n    try {\n      return await start\n    } finally {\n      this.pending.delete(key)\n    }\n  }\n\n  private async startTranscode(opts: {\n    key: string\n    deviceId: number\n    camStreamId: string\n    targetCodec: 'H264' | 'H265'\n    audio: AudioCodecTarget\n    audioEgress: boolean\n    scale: { width: number; height: number } | null\n    outResolution: { width: number; height: number }\n    outputArgs: readonly string[]\n  }): Promise<RtpSource> {\n    if (!this.acquireChannel) {\n      throw new Error('getStreamWithCodec: transcode egress not wired (acquireChannel missing)')\n    }\n    const cfg = this.ffmpegConfig()\n    const encoders = await this.resolveHardwareEncoders()\n    const encoder = pickVideoEncoder(opts.targetCodec, encoders, cfg.hwAccel)\n    const decodeHwAccel = await this.resolveDecodeHwAccel(hwAccelSettingToPrefer(cfg.hwAccel))\n    const sourceUrl = this.resolveSourceUrl(opts.deviceId, opts.camStreamId)\n    const brokerId = `transcode:${opts.deviceId}:${randomUUID().slice(0, 8)}`\n\n    // Opt-in audio sidecar: only for a re-encoded, sidecar-capable audio target.\n    const sidecarCodec = opts.audioEgress ? toAudioSidecarCodec(opts.audio) : null\n\n    const egress = new TranscodeEgress(\n      {\n        logger: this.logger.child('egress'),\n        ffmpegBinaryPath: cfg.binaryPath,\n        spawnFn: this.spawnFn,\n        acquireChannel: this.acquireChannel,\n      },\n      {\n        brokerId,\n        invocation: {\n          sourceUrl,\n          targetCodec: opts.targetCodec,\n          encoder,\n          decodeHwAccel,\n          scale: opts.scale,\n          audio: opts.audio,\n          threadCount: cfg.threadCount,\n          outputArgs: opts.outputArgs,\n        },\n        audio: sidecarCodec ? { codec: sidecarCodec } : undefined,\n      },\n    )\n\n    const url = await egress.start()\n\n    const source: RtpSource = {\n      url,\n      videoCodec: opts.targetCodec,\n      audioCodec: opts.audio,\n      resolution: opts.outResolution,\n      transcoded: true,\n      encoder,\n      pipelineKey: opts.key,\n    }\n    const entry: PipelineEntry = { key: opts.key, source, egress, refcount: 1, releaseTimer: null }\n    this.entries.set(opts.key, entry)\n    this.logger.info('Transcode pipeline started', {\n      meta: { key: opts.key, encoder, decodeHwAccel, sourceUrl, url },\n    })\n    return source\n  }\n\n  private firstPublished(deviceId: number): CameraStream | null {\n    return this.cameraStreamLookup(deviceId).find((s) => !!s.url) ?? null\n  }\n\n  /**\n   * Pick the published source whose pixel area is closest to `target`. Only\n   * considers streams that have a `url`. Returns null when none qualify (the\n   * caller falls back to `firstPublished`).\n   */\n  private pickClosestSource(deviceId: number, target: { width: number; height: number }): CameraStream | null {\n    const targetArea = target.width * target.height\n    let best: CameraStream | null = null\n    let bestDelta = Number.MAX_SAFE_INTEGER\n    for (const s of this.cameraStreamLookup(deviceId)) {\n      if (!s.url) continue\n      const w = s.resolution?.width ?? 1280\n      const h = s.resolution?.height ?? 720\n      const delta = Math.abs(w * h - targetArea)\n      if (delta < bestDelta) { bestDelta = delta; best = s }\n    }\n    return best\n  }\n\n  release(pipelineKey: string): { released: boolean; refcount: number } {\n    const entry = this.entries.get(pipelineKey)\n    if (!entry) return { released: false, refcount: 0 }\n    entry.refcount = Math.max(0, entry.refcount - 1)\n    if (entry.refcount > 0) {\n      return { released: true, refcount: entry.refcount }\n    }\n    if (entry.releaseTimer) {\n      clearTimeout(entry.releaseTimer)\n    }\n    entry.releaseTimer = setTimeout(() => {\n      const cur = this.entries.get(pipelineKey)\n      if (!cur || cur.refcount > 0) return\n      void this.shutdownEntry(cur)\n    }, this.releaseGraceMs)\n    return { released: true, refcount: 0 }\n  }\n\n  shutdownAll(): void {\n    for (const entry of this.entries.values()) {\n      void this.shutdownEntry(entry)\n    }\n    this.entries.clear()\n  }\n\n  /**\n   * Resolve a source stream to its broker restream URL\n   * (`${deviceId}/${camStreamId}`), so every consumer rides the broker's one\n   * source dial. Throws when no restream entry exists — a targeted source\n   * must be dialed before it can be consumed.\n   */\n  private resolveSourceUrl(deviceId: number, camStreamId: string): string {\n    const brokerId = makeSourceBrokerId(deviceId, camStreamId)\n    const entry = this.rtspEntryLookup(brokerId)\n    if (entry?.url) return entry.url\n    throw new Error(`getStreamWithCodec: no broker restream for ${brokerId} (source not dialed)`)\n  }\n\n  /**\n   * Resolve a profile target to its assigned, published source stream.\n   * Throws when the profile has no assignment or the assigned source is not\n   * currently published — a targeted request must not silently fall back to\n   * a different source.\n   */\n  private resolveTargetedStream(deviceId: number, profile: CamProfile): CameraStream {\n    const camStreamId = this.resolveProfileSource?.(deviceId, profile) ?? null\n    if (!camStreamId) {\n      throw new Error(\n        `getStreamWithCodec: profile \"${profile}\" has no assigned source for device ${deviceId}`,\n      )\n    }\n    const stream = this.cameraStreamLookup(deviceId).find(\n      (s) => s.camStreamId === camStreamId && !!s.url,\n    )\n    if (!stream) {\n      throw new Error(\n        `getStreamWithCodec: profile \"${profile}\" source \"${camStreamId}\" not published for device ${deviceId}`,\n      )\n    }\n    return stream\n  }\n\n  /** Query the typed, cached hardware-encoder probe; null on any failure. */\n  private async resolveHardwareEncoders(): Promise<HardwareEncoders | null> {\n    if (!this.api) return null\n    try {\n      return await this.api.platformProbe.getHardwareEncoders.query()\n    } catch (err) {\n      this.logger.warn('platformProbe.getHardwareEncoders failed — software encode fallback', {\n        meta: { error: errMsg(err) },\n      })\n      return null\n    }\n  }\n\n  /**\n   * Resolve the preferred decode hw-accel backend; null = software decode.\n   * Gated by what the configured ffmpeg binary can actually decode\n   * (`getHardwareDecodeAccels`), so we never spawn a doomed hw-decode attempt\n   * on a backend this build lacks. Per-stream fragility (e.g. VideoToolbox\n   * HEVC) is still caught by the egress's software fallback.\n   */\n  private async resolveDecodeHwAccel(prefer: FfmpegHwAccelPrefer | null): Promise<string | null> {\n    if (!this.api) return null\n    try {\n      const res = await this.api.platformProbe.resolveHwAccel.query({ prefer })\n      const supported = await this.resolveDecodeAccelMethods()\n      const picked = pickDecodeHwAccel(res.preferred, supported)\n      if (picked === null && res.preferred.length > 0) {\n        this.logger.info('decode hw-accel: configured ffmpeg supports none of the preferred backends — software decode', {\n          meta: { preferred: res.preferred, supported },\n        })\n      }\n      return picked\n    } catch (err) {\n      this.logger.warn('platformProbe.resolveHwAccel failed — software decode fallback', {\n        meta: { error: errMsg(err) },\n      })\n      return null\n    }\n  }\n\n  /**\n   * The `-hwaccel` methods the configured ffmpeg supports; `[]` on any failure\n   * (treated as \"unknown\" by {@link pickDecodeHwAccel}, which then keeps the\n   * top preference rather than filtering it out).\n   */\n  private async resolveDecodeAccelMethods(): Promise<readonly string[]> {\n    if (!this.api) return []\n    try {\n      const res = await this.api.platformProbe.getHardwareDecodeAccels.query()\n      return res.methods\n    } catch (err) {\n      this.logger.warn('platformProbe.getHardwareDecodeAccels failed — decode gate skipped', {\n        meta: { error: errMsg(err) },\n      })\n      return []\n    }\n  }\n\n  private shareEntry(entry: PipelineEntry): PipelineEntry {\n    entry.refcount += 1\n    if (entry.releaseTimer) {\n      clearTimeout(entry.releaseTimer)\n      entry.releaseTimer = null\n    }\n    return entry\n  }\n\n  private shareOrCreatePassthrough(key: string, factory: () => RtpSource): RtpSource {\n    const existing = this.entries.get(key)\n    if (existing) {\n      this.shareEntry(existing)\n      return existing.source\n    }\n    const source = factory()\n    const entry: PipelineEntry = { key, source, egress: null, refcount: 1, releaseTimer: null }\n    this.entries.set(key, entry)\n    return source\n  }\n\n  private async shutdownEntry(entry: PipelineEntry): Promise<void> {\n    if (entry.releaseTimer) {\n      clearTimeout(entry.releaseTimer)\n      entry.releaseTimer = null\n    }\n    this.entries.delete(entry.key)\n    if (entry.egress) {\n      try {\n        await entry.egress.stop()\n      } catch (err) {\n        this.logger.warn('transcode egress shutdown error', {\n          meta: { key: entry.key, error: errMsg(err) },\n        })\n      }\n    }\n  }\n}\n","import {\n  CAM_PROFILE_ORDER,\n  makeProfileBrokerId,\n  makeSourceBrokerId,\n  type CamProfile,\n  type ProfileRtspEntry,\n} from '@camstack/types'\n\nexport interface ProfileRestreamerRegistration {\n  readonly profileBrokerId: string\n  readonly camStreamId: string\n}\n\nexport interface ProfileRestreamerActions {\n  readonly register: readonly ProfileRestreamerRegistration[]\n  readonly unregister: readonly string[]\n}\n\nexport interface ProfileRestreamerActionsParams {\n  readonly deviceId: number\n  readonly assignmentMap: Partial<Record<CamProfile, string>>\n  readonly sourceBrokerExists: (camStreamId: string) => boolean\n  readonly currentBindings: ReadonlyMap<string, string>\n}\n\n/**\n * Diff the desired profile→source aliasing against the current\n * bindings. Pure: the manager supplies `sourceBrokerExists` (does a\n * live `StreamBroker` exist for this camStream?) and the current\n * `profileBrokerId → sourceBrokerId` bindings. Only emits work when a\n * profile's backing source actually changed, so an unchanged read\n * leaves the existing RTSP token (and any in-flight HAP session) alone.\n */\nexport function computeProfileRestreamerActions(\n  params: ProfileRestreamerActionsParams,\n): ProfileRestreamerActions {\n  const { deviceId, assignmentMap, sourceBrokerExists, currentBindings } = params\n  const register: ProfileRestreamerRegistration[] = []\n  const unregister: string[] = []\n  for (const profile of CAM_PROFILE_ORDER) {\n    const profileBrokerId = makeProfileBrokerId(deviceId, profile)\n    const currentSource = currentBindings.get(profileBrokerId) ?? null\n    const camStreamId = assignmentMap[profile]\n    if (camStreamId === undefined) {\n      // Unassigned profile — drop any token that was aliasing it.\n      if (currentSource !== null) unregister.push(profileBrokerId)\n      continue\n    }\n    // Past this guard `camStreamId` narrows to `string`, so the register\n    // push needs no extra defined-check or cast.\n    const desiredSource = sourceBrokerExists(camStreamId)\n      ? makeSourceBrokerId(deviceId, camStreamId)\n      : null\n    if (desiredSource === currentSource) continue\n    if (currentSource !== null) unregister.push(profileBrokerId)\n    if (desiredSource !== null) register.push({ profileBrokerId, camStreamId })\n  }\n  return { register, unregister }\n}\n\n/** Minted restream URL pair for a profile-keyed brokerId. */\nexport interface ProfileRestreamUrl {\n  readonly url: string\n  readonly mutedUrl: string\n  readonly enabled: boolean\n}\n\n/** Source-camStream codec/resolution used to enrich a profile entry. */\nexport interface ProfileSourceMeta {\n  readonly codec?: string\n  readonly resolution?: { readonly width: number; readonly height: number }\n}\n\nexport interface BuildProfileRtspEntriesParams {\n  readonly deviceId: number\n  readonly assignmentMap: Partial<Record<CamProfile, string>>\n  readonly resolveEntry: (profileBrokerId: string) => ProfileRestreamUrl | null\n  readonly resolveMeta: (camStreamId: string) => ProfileSourceMeta\n}\n\n/**\n * Map a device's assigned profiles → `ProfileRtspEntry[]`. Pure: the\n * manager supplies `resolveEntry` (profileBrokerId → minted restream\n * URL or null) and `resolveMeta` (camStreamId → codec/resolution).\n * Profiles with no assignment or no mintable entry are skipped.\n */\nexport function buildProfileRtspEntries(\n  params: BuildProfileRtspEntriesParams,\n): ProfileRtspEntry[] {\n  const { deviceId, assignmentMap, resolveEntry, resolveMeta } = params\n  const result: ProfileRtspEntry[] = []\n  for (const profile of CAM_PROFILE_ORDER) {\n    const camStreamId = assignmentMap[profile]\n    if (camStreamId === undefined) continue\n    const profileBrokerId = makeProfileBrokerId(deviceId, profile)\n    const entry = resolveEntry(profileBrokerId)\n    if (!entry) continue\n    const meta = resolveMeta(camStreamId)\n    result.push({\n      profile,\n      brokerId: profileBrokerId,\n      url: entry.url,\n      mutedUrl: entry.mutedUrl,\n      enabled: entry.enabled,\n      ...(meta.codec !== undefined ? { codec: meta.codec } : {}),\n      ...(meta.resolution !== undefined ? { resolution: meta.resolution } : {}),\n    })\n  }\n  return result\n}\n","import type {\n  StreamSource, IRestreamer, IDecoderProvider, IScopedLogger,\n  StreamFormat, ConfigUISchemaWithValues, ConfigFieldWithValue,\n  ConfigSectionWithValues, ConfigSubTabDefinitionWithValue,\n  CameraStream, CamProfile, CamStreamKind, ProfileSlot, ProfileSlotStatus,\n  IEventBus, SystemEvent, IStreamBroker, BrokerStats, AddonApi,\n  PlaceholderReason,\n  GetStreamWithCodecInput, RtpSource,\n  SubscribeFramesInput, SubscribeFramesResult, FrameHandle,\n  SubscribeAudioChunksInput, SubscribeAudioChunksResult, DecodedAudioChunk,\n  CamStreamDescriptor,\n} from '@camstack/types'\nimport { StreamBroker } from './stream-broker'\nimport type { DecoderCapApi } from './decoder-session-proxy.js'\nimport { buildAudioCodecCapApi } from './audio-codec-proxy.js'\nimport { RtspListenServer } from '../rtsp/rtsp-listen-server.js'\nimport { RtspRestreamer } from '../rtsp/rtsp-restreamer.js'\nimport { RtspRestreamProviderImpl, type RtspRestreamEntry } from '../rtsp/rtsp-restream-provider.js'\nimport { TranscodePipelineManager } from './transcode-pipeline.js'\nimport type { EgressChannel } from './transcode-egress.js'\nimport type { FfmpegSystemConfig } from './ffmpeg-invocation.js'\nimport { CAM_PROFILE_ORDER, DeviceFeature, DeviceType, EventCategory, errMsg, RingBuffer, EncodeProfileSchema, makeSourceBrokerId, makeProfileBrokerId, type EncodeProfile, type VideoEncode, type ProfileRtspEntry } from '@camstack/types'\nimport {\n  computeProfileRestreamerActions,\n  buildProfileRtspEntries,\n} from './profile-restreamers.js'\nimport { z } from 'zod'\n\n/**\n * Runtime capability accessor injected by the addon on initialization.\n * Mirrors the pattern used by other addons — the server extends\n * AddonContext at runtime with this shape so the broker can consume\n * other capabilities (decoder, restreamer, webrtc) without a backend\n * dependency.\n */\nexport interface CapabilitiesAccess {\n  getCollection?<T = unknown>(name: string): readonly T[] | undefined\n  getCollectionEntries?<T = unknown>(name: string): readonly (readonly [string, T])[] | undefined\n  get?<T = unknown>(name: string): T | undefined\n}\n\ninterface StreamPreBuffer {\n  readonly enabled: boolean\n  readonly seconds: number\n}\n\nexport interface DeviceOverride {\n  /** Legacy single-value override — kept for migration compat. */\n  preBufferSecOverride?: number\n  /**\n   * Per-stream pre-buffer settings keyed by `camStreamId` (e.g.\n   * `native:main`, `rtsp:sub`). Each stream sourceconfigures its own\n   * buffer independently; profile assignment is decoupled — a profile\n   * always inherits the pre-buffer of the cam stream currently mapped\n   * to it via `assignments[deviceId].map[profile]`.\n   */\n  preBuffer?: Record<string, StreamPreBuffer>\n  /**\n   * Per-device debug flag for streaming subsystem (broker + WebRTC\n   * session + repacketizer). When true, verbose info-level logs about\n   * SDP param sets, RTP forwarding counts, repacketizer codecInfo\n   * seeding, and source-RTP subscriber lifecycle become enabled.\n   * Off by default to keep production logs quiet.\n   */\n  streamingDebug?: boolean\n  /**\n   * Per-device \"Advanced WebRTC debug logging\" flag. When true, EVERY\n   * WebRTC `AdaptiveSession` created for this device gets `debug = true`,\n   * which turns on the full per-step lifecycle logging in `session.ts`\n   * (local answer ICE candidates + types, remote offer candidates, DTLS\n   * state transitions, ICE nomination, codec negotiation, first\n   * video/audio RTP, audio codec/clock). Distinct from `streamingDebug`\n   * (broker/repacketizer verbosity) so an operator can isolate the\n   * WebRTC negotiation path alone. Off by default.\n   */\n  webrtcDebug?: boolean\n  /**\n   * Operator-defined derived streams. Each entry is materialised\n   * into a live broker by `loadPersistedDeviceOverrides` on hub\n   * boot via publishCameraStream({kind:'derived', …}).\n   */\n  derivedStreams?: Record<string, DerivedStreamEntry>\n}\n\ninterface DerivedStreamEntry {\n  readonly sourceCamStreamId: string\n  readonly encodeProfile: EncodeProfile\n  readonly preBuffer?: StreamPreBuffer\n  readonly rtspEnabled?: boolean\n}\n\nconst StreamPreBufferSchema = z.object({\n  enabled: z.boolean(),\n  seconds: z.number(),\n})\n\nconst DerivedStreamSchema = z.object({\n  // Empty string = an unconfigured placeholder: the UI creates a derived\n  // stream (name + encode params) BEFORE the operator picks a source. That\n  // state must persist so the half-built stream survives a restart — a\n  // `min(1)` here rejected the whole device-overrides blob on save (one\n  // placeholder blocked EVERY device). Runtime materialization skips empty\n  // sources, so an unconfigured entry is harmless until a source is set.\n  sourceCamStreamId: z.string(),\n  encodeProfile: EncodeProfileSchema,\n  preBuffer: StreamPreBufferSchema.optional(),\n  rtspEnabled: z.boolean().optional(),\n})\n\nconst DeviceOverrideSchema = z.object({\n  preBufferSecOverride: z.number().optional(),\n  preBuffer: z.record(z.string(), StreamPreBufferSchema).optional(),\n  streamingDebug: z.boolean().optional(),\n  webrtcDebug: z.boolean().optional(),\n  derivedStreams: z.record(z.string(), DerivedStreamSchema).optional(),\n})\n\nexport function serializeDeviceOverride(o: DeviceOverride): string {\n  return JSON.stringify(o)\n}\n\nexport function parseDeviceOverride(s: string): DeviceOverride {\n  return DeviceOverrideSchema.parse(JSON.parse(s))\n}\n\n/** Whole-blob schema for the persisted `deviceOverrides` store key:\n *  numeric-string deviceId → DeviceOverride (includes `derivedStreams`). */\nexport const DeviceOverridesMapSchema = z.record(z.string(), DeviceOverrideSchema)\nexport type DeviceOverridesMap = z.infer<typeof DeviceOverridesMapSchema>\n\n/** Persisted store-key schemas for the broker's other durable blobs.\n *  `rtspTokens`: camStreamKey → token hex. `rtspEnabled`: brokerId →\n *  enabled. `profileMap`: numeric-string deviceId → {map, auto}. */\nexport const RtspTokensSchema = z.record(z.string(), z.string())\nexport const RtspEnabledSchema = z.record(z.string(), z.boolean())\nexport const ProfileMapSchema = z.record(\n  z.string(),\n  z.object({\n    map: z.record(z.string(), z.string()),\n    auto: z.boolean(),\n  }),\n)\nexport type RtspTokensBlob = z.infer<typeof RtspTokensSchema>\nexport type RtspEnabledBlob = z.infer<typeof RtspEnabledSchema>\nexport type ProfileMapBlob = z.infer<typeof ProfileMapSchema>\n\n/**\n * Per-device assignment state persisted via `setProfileMapPersister`.\n * `auto` is true until the operator makes a manual assignment — used\n * so providers can keep publishing streams and have the broker\n * re-run `computeInitialAssignment` without overwriting a user's\n * chosen mapping.\n */\ninterface DeviceAssignment {\n  readonly map: Partial<Record<CamProfile, string>>\n  readonly auto: boolean\n}\n\nexport type ProfileMapPersister = (\n  assignments: ReadonlyMap<number, DeviceAssignment>,\n) => void\n\ntype CameraStreamInternal = CameraStream & {\n  /** Insertion order within (deviceId) — drives UI dropdown ordering. */\n  readonly order: number\n  readonly deviceFeatures: readonly string[]\n  /**\n   * Whether this stream is eligible for the broker's automatic profile\n   * assignment. `false` streams stay published (visible + manually\n   * assignable), but `computeInitialAssignment` skips them. Defaults\n   * `true` for back-compat with publishers that don't set the flag.\n   */\n  readonly autoEligible: boolean\n}\n\nconst PUSH_KINDS: ReadonlySet<CamStreamKind> = new Set(['push-annexb'])\n\n/** Walk a profile map and return the slot that points at `camStreamId`, or null. */\nfunction findProfileForStream(\n  map: Partial<Record<CamProfile, string>>,\n  camStreamId: string,\n): CamProfile | null {\n  for (const profile of CAM_PROFILE_ORDER) {\n    if (map[profile] === camStreamId) return profile\n  }\n  return null\n}\n\n\n/**\n * Watchdog parameters for the stream-health emitter. Brokers that have\n * received zero video packets for STREAM_STALE_TIMEOUT_MS are reported\n * as `stream.offline`; the next packet flips them back to `stream.online`.\n */\nconst STREAM_STALE_TIMEOUT_MS = 120_000\nconst STREAM_HEALTH_POLL_MS = 15_000\n\n/** Compose a SOURCE broker id from a `(deviceId, camStreamId)` tuple. Thin\n *  wrapper over the canonical `makeSourceBrokerId` — single source of truth. */\nfunction brokerIdFor(deviceId: number, camStreamId: string): string {\n  return makeSourceBrokerId(deviceId, camStreamId)\n}\n\n/**\n * Inverse of `brokerIdFor`: split a `<deviceId>/<camStreamId>` broker id back\n * into its tuple. Splits on the FIRST `/` (the deviceId is a leading integer;\n * the camStreamId may itself contain no `/` today, but be liberal). Returns\n * null when the leading segment isn't a non-negative integer.\n */\nfunction parseBrokerId(brokerId: string): { deviceId: number; camStreamId: string } | null {\n  const slash = brokerId.indexOf('/')\n  if (slash <= 0) return null\n  const deviceId = Number(brokerId.slice(0, slash))\n  const camStreamId = brokerId.slice(slash + 1)\n  if (!Number.isInteger(deviceId) || deviceId < 0 || camStreamId.length === 0) return null\n  return { deviceId, camStreamId }\n}\n\nexport class StreamBrokerManager {\n  /**\n   * brokers keyed by brokerId = `${deviceId}/${camStreamId}`.\n   *\n   * Single source of truth: one StreamBroker per cam stream source.\n   * Profiles are an alias layer over this map — the profile slot\n   * `(deviceId, profile)` resolves to the broker via\n   * `assignments[deviceId].map[profile] -> camStreamId`.\n   *\n   * Lifecycle is tied 1:1 to publish/retract: a broker exists for as\n   * long as the cam stream is published. Pre-buffer defaults OFF —\n   * idle cam streams cost roughly nothing (the upstream readers stay\n   * dormant until something subscribes), but we keep the broker\n   * registered so callers (WebRTC, RTSP restream) can latch on at any\n   * time without an activation handshake.\n   */\n  private readonly brokers = new Map<string, StreamBroker>()\n  /** profileBrokerId (`${deviceId}/${profile}`) → sourceBrokerId currently aliased. */\n  private readonly profileRestreamerBindings = new Map<string, string>()\n  /** Per-device published cam-stream registry. */\n  private readonly cameraStreams = new Map<number, Map<string, CameraStreamInternal>>()\n  /** Monotonic insertion counter per device — feeds `CameraStreamInternal.order`. */\n  private readonly streamOrderSeq = new Map<number, number>()\n  /** Persisted operator intent: which cam stream serves which profile. */\n  private readonly assignments = new Map<number, DeviceAssignment>()\n  /** Per-device overrides (pre-buffer + legacy). */\n  private readonly deviceOverrides = new Map<number, DeviceOverride>()\n  private deviceOverridePersister: ((overrides: ReadonlyMap<number, DeviceOverride>) => void) | null = null\n  private profileMapPersister: ProfileMapPersister | null = null\n  private eventBus: IEventBus | null = null\n  /**\n   * Per-broker stream health flag set by the watchdog. `undefined`\n   * before the first transition — emit `stream.online` on first packet\n   * regardless of prior state.\n   */\n  private readonly streamHealthByBroker = new Map<string, boolean>()\n  private streamHealthTimer: ReturnType<typeof setInterval> | undefined\n  /**\n   * Routes a frame-handle `subscriptionId` (Phase 5 / D9) back to its owning\n   * brokerId so `pullFrameHandles` / `unsubscribeFrames` resolve in O(1)\n   * without scanning every broker.\n   */\n  private readonly frameSubscriptionBroker = new Map<string, string>()\n  /**\n   * Routes an audio-chunk `subscriptionId` (Phase 5 / D9) back to its owning\n   * brokerId so `pullAudioChunks` / `unsubscribeAudioChunks` resolve in O(1)\n   * without scanning every broker. Mirrors `frameSubscriptionBroker`.\n   */\n  private readonly audioSubscriptionBroker = new Map<string, string>()\n  /**\n   * Derived streams that could not be materialised at boot because their\n   * source broker had not yet been registered. Keyed by the source brokerId\n   * (`${deviceId}/${sourceCamStreamId}`). Drained by `publishCameraStream`\n   * when the matching source is registered.\n   */\n  private readonly pendingDerivedStreams = new Map<string, Array<{ deviceId: number; name: string; def: NonNullable<DeviceOverride['derivedStreams']>[string] }>>()\n  // Legacy static restreamers list — production reads via capabilities.\n  private restreamers: readonly IRestreamer[] = []\n  private readonly staticDecoders: readonly IDecoderProvider[]\n  private readonly logger: IScopedLogger\n  /** Public accessor for same-process callers (e.g. `CameraStreamsProvider.pickStream`)\n   *  that want to emit diagnostics through the manager's logger. Internal callers\n   *  use `this.logger` directly. */\n  getLogger(): IScopedLogger { return this.logger }\n  private readonly rtspServer = new RtspListenServer()\n  private readonly rtspProvider: RtspRestreamProviderImpl\n  private persistedTokens = new Map<string, string>()\n  private capabilities: CapabilitiesAccess | null = null\n  /**\n   * tRPC api proxy injected from the host addon. Used to resolve the\n   * `decoder` capability (and any future cross-cap consumers) via the\n   * canonical API surface — `localProviderLink` short-circuits when\n   * the provider lives in the same broker / group, otherwise the call\n   * goes over `brokerTransportLink`. Replaces the previous direct\n   * `CapabilityRegistry` lookup, which only worked on the hub.\n   */\n  private api: AddonApi | null = null\n  private defaultPreBufferSec = 10\n  private webrtcServer: import('../webrtc/broker-webrtc-server.js').BrokerWebrtcServer | null = null\n  /**\n   * Transcode-pipeline manager — backs `getStreamWithCodec`. Lazily\n   * created on first acquire, lives until `destroyAll`.\n   */\n  private transcodeManager: TranscodePipelineManager | null = null\n  /**\n   * Snapshot of the system `ffmpeg` config section (binaryPath / hwAccel /\n   * threadCount). Refreshed by the host addon via `setFfmpegConfig`; the\n   * transcode manager reads it live per spawn. Defaults are safe (PATH ffmpeg,\n   * auto hw-accel, auto threads) until the addon loads the section.\n   */\n  private ffmpegConfig: FfmpegSystemConfig = { binaryPath: 'ffmpeg', hwAccel: 'auto', threadCount: 0 }\n\n  constructor(\n    staticDecoders: readonly IDecoderProvider[] | undefined,\n    logger: IScopedLogger,\n  ) {\n    this.staticDecoders = staticDecoders ?? []\n    this.logger = logger\n    this.rtspProvider = new RtspRestreamProviderImpl(this.rtspServer, logger.child('rtsp-restream'))\n  }\n\n  /**\n   * Lazily create the transcode manager wired with closures rather than\n   * direct refs so it only sees the surface it needs.\n   */\n  private getTranscodeManager(): TranscodePipelineManager {\n    if (!this.transcodeManager) {\n      this.transcodeManager = new TranscodePipelineManager({\n        logger: this.logger.child('transcode'),\n        api: this.api,\n        cameraStreamLookup: (deviceId) => this.getCameraStreamsForDevice(deviceId),\n        rtspEntryLookup: (brokerId) => {\n          const entry = this.rtspProvider.getEntry(brokerId)\n          return entry ? { url: entry.url } : null\n        },\n        resolveProfileSource: (deviceId, profile) =>\n          this.assignments.get(deviceId)?.map[profile] ?? null,\n        ffmpegConfig: () => this.ffmpegConfig,\n        // Create + register the egress restreamer on the main listen server.\n        // Holds the concrete `RtspRestreamer` here so no cast is needed at the\n        // listen-server boundary; the egress only sees the structural `*Like`.\n        acquireChannel: (brokerId): EgressChannel => {\n          const restreamer = new RtspRestreamer(brokerId)\n          const token = this.rtspServer.registerRestreamer(brokerId, restreamer)\n          return {\n            restreamer,\n            consumerUrl: `rtsp://127.0.0.1:${this.rtspProvider.getRtspPort()}/${token}`,\n            release: () => {\n              this.rtspServer.unregisterRestreamer(brokerId)\n              restreamer.destroy()\n            },\n          }\n        },\n      })\n    }\n    return this.transcodeManager\n  }\n\n  // ── Cap methods: getStreamWithCodec / releaseStreamWithCodec ─────────\n\n  async getStreamWithCodec(input: GetStreamWithCodecInput): Promise<RtpSource> {\n    return this.getTranscodeManager().acquire(input)\n  }\n\n  async releaseStreamWithCodec(input: { pipelineKey: string }): Promise<{ released: boolean; refcount: number }> {\n    return this.getTranscodeManager().release(input.pipelineKey)\n  }\n\n  // ── Decoder resolution ───────────────────────────────────────────────\n  //\n  // Routes every decoder call through `ctx.api.decoder.*` (tRPC). The\n  // proxy is wired by the host addon via `setApiAccess(ctx.api)`. When\n  // the decoder addon lives in the same broker (e.g. inside the\n  // `pipeline` group on the hub) the call is short-circuited by\n  // `localProviderLink` with no serialization. When the decoder lives\n  // on a different node, the call falls through to\n  // `brokerTransportLink` which dispatches the matching Moleculer\n  // action. Either way, the broker stays unaware of the addon's\n  // physical placement.\n\n  private buildDecoderCapApi(_deviceId: number | null): DecoderCapApi | null {\n    const localApi = this.buildLocalDecoderCapApi()\n    const apiDecoder = (): {\n      supportsCodec: { query: (input: { codec: string }) => Promise<boolean> }\n      getInfo: { query: (input?: void) => Promise<{ id: string; name: string; isPullMode?: boolean; priority?: number }> }\n      createSession: { query: (input: import('@camstack/types').DecoderSessionConfig) => Promise<{ sessionId: string; nodeId: string }> }\n      pushPacket: { query: (input: { sessionId: string; packet: import('@camstack/types').EncodedPacket }) => Promise<void> }\n      pullFrames: { query: (input: { sessionId: string; maxCount: number }) => Promise<readonly import('@camstack/types').DecodedFrame[]> }\n      pullHandles: { query: (input: { sessionId: string; maxCount: number }) => Promise<readonly import('@camstack/types').FrameHandle[]> }\n      destroySession: { query: (input: { sessionId: string }) => Promise<void> }\n      openStream: { query: (input: { sessionId: string; url: string }) => Promise<void> }\n      updateConfig: { query: (input: { sessionId: string; config: Partial<import('@camstack/types').DecoderSessionConfig> }) => Promise<void> }\n      getStats: { query: (input: { sessionId: string }) => Promise<import('@camstack/types').DecoderStats> }\n    } | null => {\n      if (!this.api) return null\n      const decoder = (this.api as Record<string, unknown>)['decoder']\n      return decoder ? (decoder as ReturnType<typeof apiDecoder> extends infer R ? R extends null ? never : R : never) : null\n    }\n\n    return {\n      supportsCodec: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.supportsCodec.query(input)\n        if (!localApi) return false\n        return localApi.supportsCodec(input)\n      },\n      getInfo: async () => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.getInfo.query()\n        if (!localApi) return { id: 'none', name: 'No decoder' }\n        return localApi.getInfo()\n      },\n      createSession: async (config) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.createSession.query(config)\n        if (!localApi) throw new Error('No decoder provider available')\n        return localApi.createSession(config)\n      },\n      pushPacket: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.pushPacket.query(input)\n        if (!localApi) return\n        return localApi.pushPacket(input)\n      },\n      pullFrames: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.pullFrames.query(input)\n        if (!localApi) return []\n        return localApi.pullFrames(input)\n      },\n      pullHandles: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.pullHandles.query(input)\n        if (!localApi) return []\n        return localApi.pullHandles(input)\n      },\n      destroySession: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.destroySession.query(input)\n        if (!localApi) return\n        return localApi.destroySession(input)\n      },\n      openStream: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.openStream.query(input)\n        if (!localApi) return\n        return localApi.openStream(input)\n      },\n      updateConfig: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.updateConfig.query(input)\n        if (!localApi) return\n        return localApi.updateConfig(input)\n      },\n      getStats: async (input) => {\n        const proxy = apiDecoder()\n        if (proxy) return proxy.getStats.query(input)\n        if (!localApi) return { inputFps: 0, outputFps: 0, avgDecodeTimeMs: 0, droppedFrames: 0 }\n        return localApi.getStats(input)\n      },\n    }\n  }\n\n  private buildLocalDecoderCapApi(): DecoderCapApi | null {\n    if (this.staticDecoders.length === 0) return null\n    // Bounded frame buffer per session — when `pullFrames` falls behind the\n    // decoder's frame rate, old frames get evicted rather than accumulating\n    // unboundedly. Mirrors the capacity used by the nodeav addon.\n    const FRAME_BUFFER_CAPACITY = 32\n    type DecodedFrame = import('@camstack/types').DecodedFrame\n    const frameBuffers = new Map<string, RingBuffer<DecodedFrame>>()\n    const sessions = new Map<string, import('@camstack/types').IDecoderSession>()\n    const unsubscribers = new Map<string, () => void>()\n    let nextId = 0\n\n    const resolveDecoder = async (codec: string): Promise<IDecoderProvider | null> => {\n      for (const p of this.staticDecoders) {\n        if (await p.supportsCodec({ codec })) return p\n      }\n      return null\n    }\n\n    return {\n      supportsCodec: async (input) => (await resolveDecoder(input.codec)) !== null,\n      getInfo: async () => {\n        const provider = this.staticDecoders[0]\n        if (!provider) return { id: 'none', name: 'No decoder' }\n        return { id: provider.id, name: provider.name, isPullMode: provider.isPullMode, priority: provider.priority }\n      },\n      createSession: async (config) => {\n        const provider = await resolveDecoder(config.codec)\n        if (!provider) throw new Error(`No decoder provider for codec \"${config.codec}\"`)\n        const session = await provider.createSession(config)\n        const sessionId = `dec-${++nextId}`\n        sessions.set(sessionId, session)\n        const buffer = new RingBuffer<DecodedFrame>(FRAME_BUFFER_CAPACITY)\n        frameBuffers.set(sessionId, buffer)\n        const unsub = session.onFrame((frame) => { buffer.push(frame) })\n        unsubscribers.set(sessionId, unsub)\n        return { sessionId, nodeId: 'local' }\n      },\n      pushPacket: async (input) => {\n        const session = sessions.get(input.sessionId)\n        if (session) session.pushPacket(input.packet)\n      },\n      pullFrames: async (input) => {\n        const buffer = frameBuffers.get(input.sessionId)\n        if (!buffer) return []\n        return buffer.drain(input.maxCount)\n      },\n      // The in-process `staticDecoders` fallback has no shm sink — its\n      // `IDecoderSession` exposes only `onFrame` (pixels), not\n      // `onFrameHandle`. The shared-memory frame plane (Phase 5 / D9) runs\n      // through the real `decoder` cap (the node-av addon, which owns the\n      // `frameSink: 'shm'` ring writer). `pullHandles` therefore returns\n      // nothing here — a `frameSink: 'shm'` request degrades to no frames\n      // rather than crashing the broker.\n      pullHandles: async () => [],\n      destroySession: async (input) => {\n        const session = sessions.get(input.sessionId)\n        if (session) {\n          const unsub = unsubscribers.get(input.sessionId)\n          if (unsub) unsub()\n          sessions.delete(input.sessionId)\n          frameBuffers.delete(input.sessionId)\n          unsubscribers.delete(input.sessionId)\n          await session.destroy()\n        }\n      },\n      openStream: async (input) => {\n        const session = sessions.get(input.sessionId)\n        if (session?.openStream) await session.openStream(input.url)\n      },\n      updateConfig: async (input) => {\n        const session = sessions.get(input.sessionId)\n        if (session) session.updateConfig(input.config)\n      },\n      getStats: async (input) => {\n        const session = sessions.get(input.sessionId)\n        if (!session) return { inputFps: 0, outputFps: 0, avgDecodeTimeMs: 0, droppedFrames: 0 }\n        return session.getStats()\n      },\n    }\n  }\n\n  // ── Setters (wiring from the addon) ──────────────────────────────────\n\n  setCapabilitiesAccess(access: CapabilitiesAccess): void {\n    this.capabilities = access\n  }\n\n  /**\n   * Wire the addon's `ctx.api` proxy. Required for the broker to\n   * resolve the `decoder` capability via tRPC — see\n   * `buildDecoderCapApi` for the full rationale.\n   */\n  setApiAccess(api: AddonApi): void {\n    this.api = api\n  }\n\n  /** Update the system `ffmpeg` config snapshot used by the transcode egress. */\n  setFfmpegConfig(cfg: FfmpegSystemConfig): void {\n    this.ffmpegConfig = cfg\n  }\n\n  setDefaultPreBufferSec(sec: number): void {\n    this.defaultPreBufferSec = sec\n  }\n\n  setEventBus(bus: IEventBus): void {\n    this.eventBus = bus\n    this.startStreamHealthWatchdog()\n    this.subscribePlaceholderStateSources(bus)\n  }\n\n  /**\n   * Subscribe to the cap-scoped bus events that drive the per-broker\n   * placeholder image. Lives in the manager (not the broker) because\n   * a single device can host multiple brokers (`high`/`mid`/`low`)\n   * and one listener fans out to all of them. Lives at the broker\n   * layer (not the provider) because providers must stay oblivious\n   * to the placeholder pipeline — they only mutate their runtime-\n   * state slices.\n   *\n   * Each cap defines its own state-change event (`BatteryStatus`\n   * surfaces `battery.onStatusChanged`), wired automatically by the\n   * `subscribeCap('battery')` bridge in the provider's\n   * `registerBatteryIfSupported`. We subscribe to the cap event\n   * directly rather than the catch-all `device.state-changed`\n   * firehose so we don't filter the entire device-state stream\n   * client-side and we get a typed payload.\n   */\n  private subscribePlaceholderStateSources(bus: IEventBus): void {\n    bus.subscribe({ category: EventCategory.BatteryOnStatusChanged }, (event: SystemEvent) => {\n      const data = event.data as {\n        deviceId?: unknown\n        status?: { sleeping?: unknown } | null\n      }\n      const deviceId = typeof data.deviceId === 'number' ? data.deviceId : null\n      if (deviceId === null) return\n      // Slice may be `undefined` (cap slice cleared) or `null` (slice\n      // dropped); both map to \"no longer sleeping\" — fall back to the\n      // generic reconnecting placeholder.\n      const sleeping = data.status?.sleeping === true\n      this.applyPlaceholderReasonToDevice(deviceId, sleeping ? 'sleeping' : 'reconnecting')\n    })\n\n    // Wake-up window: between the Baichuan wake-up issue and the\n    // camera's first dialed-back RTP packet. Surfaces as a labelled\n    // \"WAKING UP\" tile (`'waking'` placeholder reason) on encoded-path\n    // subscribers (H.264 sessions, derived-broker pipelines). The\n    // H.265 RTP repacketizer path still drops placeholders to avoid\n    // freezing Chrome's HEVC decoder — see `BrokerWebrtcServer`'s\n    // useRtpRepacketizer guard. The reason snaps back via the\n    // subsequent `BatteryOnStatusChanged { sleeping: false }` event,\n    // or implicitly once the first real packet flips the broker's\n    // status to `'streaming'`.\n    bus.subscribe({ category: EventCategory.BatteryOnWakeStarted }, (event: SystemEvent) => {\n      const data = event.data as { deviceId?: unknown }\n      const deviceId = typeof data.deviceId === 'number' ? data.deviceId : null\n      if (deviceId === null) return\n      this.applyPlaceholderReasonToDevice(deviceId, 'waking')\n    })\n\n    const handlerForReason = (reason: PlaceholderReason) =>\n      (event: SystemEvent) => {\n        const data = event.data as { deviceId?: unknown }\n        const deviceId = typeof data.deviceId === 'number' ? data.deviceId : null\n        if (deviceId === null) return\n        this.applyPlaceholderReasonToDevice(deviceId, reason)\n      }\n    bus.subscribe({ category: EventCategory.DeviceOffline }, handlerForReason('offline'))\n    bus.subscribe({ category: EventCategory.DeviceDisabled }, handlerForReason('disabled'))\n  }\n\n  setWebrtcServer(server: import('../webrtc/broker-webrtc-server.js').BrokerWebrtcServer): void {\n    this.webrtcServer = server\n    // Inject the assignments-aware tier resolver so the WebRTC adaptive\n    // bitrate logic can map cam-stream-keyed brokers back to their\n    // current profile tier ('high'/'mid'/'low'). Without this, all\n    // candidates appear \"untiered\" and `prefersTier` hints fall through\n    // to the order-based pick.\n    server.setProfileTierResolver((brokerId) => {\n      const parsed = this.parseBrokerId(brokerId)\n      if (!parsed) return null\n      return findProfileForStream(this.assignments.get(parsed.deviceId)?.map ?? {}, parsed.camStreamId)\n    })\n    // Back-register every broker created BEFORE the server was wired. Addon\n    // activation runs `reconcileAllCatalogsAtStartup()` (which creates brokers\n    // via ensureBroker → fanOutRegistration) BEFORE this call, so those\n    // registrations hit the `if (this.webrtcServer)` guard while it was still\n    // undefined and were silently dropped. They live only in `this.brokers`;\n    // ensureBroker's `this.brokers.has` short-circuit means no later reconcile\n    // ever repairs them — so the WebRTC server would never learn about them and\n    // every session for those devices fails with \"No broker for stream …\".\n    // Re-registering here also makes a server swap self-healing.\n    for (const [brokerId, broker] of this.brokers) {\n      server.registerBroker(brokerId, broker)\n    }\n  }\n\n  setProfileMapPersister(persister: ProfileMapPersister): void {\n    this.profileMapPersister = persister\n  }\n\n  loadPersistedProfileMap(entries: ReadonlyMap<number, DeviceAssignment>): void {\n    this.assignments.clear()\n    for (const [k, v] of entries) this.assignments.set(k, v)\n  }\n\n  // ── Legacy restreamer accessors (kept for tests) ────────────────────\n\n  private getLiveRestreamers(): readonly IRestreamer[] {\n    const live = this.capabilities?.getCollection?.<IRestreamer>('restreamer')\n    if (live && live.length > 0) return live\n    return this.restreamers\n  }\n\n  setRestreamers(restreamers: readonly IRestreamer[]): void {\n    this.restreamers = restreamers\n    for (const broker of this.brokers.values()) broker.setRestreamers(restreamers)\n    this.logger.info('Updated restreamers', { meta: { count: restreamers.length } })\n  }\n\n  getRestreamers(): readonly IRestreamer[] {\n    return this.restreamers\n  }\n\n  getRtspRestreamProvider(): RtspRestreamProviderImpl {\n    return this.rtspProvider\n  }\n\n  loadPersistedTokens(tokens: ReadonlyMap<string, string>): void {\n    this.persistedTokens = new Map(tokens)\n  }\n\n  // ── Cap methods: cam stream lifecycle ────────────────────────────────\n\n  /** Upsert a single SOURCE (non-derived) cam-stream into the registry\n   *  WITHOUT ranking or broker creation. Returns the prior entry (if any)\n   *  so callers can detect url refresh / already-existed. */\n  private upsertSourceCameraStream(\n    deviceId: number,\n    input: {\n      camStreamId: string\n      kind: CamStreamKind\n      url?: string\n      codec?: string\n      resolution?: { width: number; height: number }\n      fps?: number\n      label?: string\n      deviceFeatures?: readonly string[]\n      autoEligible?: boolean\n      metadata?: Readonly<Record<string, unknown>>\n    },\n  ): CameraStreamInternal | undefined {\n    const { camStreamId, kind, url, codec, resolution, fps, label, metadata } = input\n    let streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) {\n      streamMap = new Map()\n      this.cameraStreams.set(deviceId, streamMap)\n    }\n    const existing = streamMap.get(camStreamId)\n    const order = existing?.order ?? this.nextOrder(deviceId)\n    const deviceFeatures: string[] = input.deviceFeatures\n      ? [...input.deviceFeatures]\n      : (existing?.deviceFeatures ? [...existing.deviceFeatures] : [])\n    const autoEligible = input.autoEligible ?? existing?.autoEligible ?? true\n    const cam: CameraStreamInternal = {\n      camStreamId,\n      deviceId,\n      kind,\n      ...(url !== undefined ? { url } : {}),\n      ...(codec !== undefined ? { codec } : {}),\n      ...(resolution !== undefined ? { resolution } : {}),\n      ...(fps !== undefined ? { fps } : {}),\n      ...(label !== undefined ? { label } : {}),\n      ...(metadata !== undefined ? { metadata } : (existing?.metadata !== undefined ? { metadata: existing.metadata } : {})),\n      order,\n      deviceFeatures,\n      autoEligible,\n    }\n    streamMap.set(camStreamId, cam)\n    return existing\n  }\n\n  async publishCameraStream(input: {\n    deviceId: number\n    camStreamId: string\n    kind: CamStreamKind\n    url?: string\n    codec?: string\n    resolution?: { width: number; height: number }\n    fps?: number\n    label?: string\n    /**\n     * Device-level features the publisher advertised — single source of\n     * truth for downstream policy (battery → relaxed stall watchdog,\n     * prebuffer off, snapshot rate-limit longer). Re-publishes overwrite\n     * the prior list so device feature changes (e.g. live battery\n     * detection) propagate.\n     */\n    deviceFeatures?: readonly string[]\n    /**\n     * Whether this stream is eligible for automatic profile assignment.\n     * Re-publishes overwrite — flipping a stream from auto-eligible to\n     * not (or vice versa) takes effect on the next assignment recompute.\n     * Default `true`.\n     */\n    autoEligible?: boolean\n    /**\n     * Transport-specific opaque metadata. Stored alongside the stream\n     * record and forwarded to the source reader via\n     * `StreamSource.metadata`. `pull-rfc4571` publishers put the SDP\n     * here.\n     */\n    metadata?: Readonly<Record<string, unknown>>\n    /**\n     * Required when `kind === 'derived'`. The `camStreamId` of the source\n     * stream (within the same device) whose encoded plane this derived\n     * stream re-encodes.\n     */\n    sourceCamStreamId?: string\n    /**\n     * Required when `kind === 'derived'`. The encode parameters that the\n     * broker-spawned transcode pipeline will apply to the source plane.\n     */\n    encodeProfile?: EncodeProfile\n  }): Promise<{ success: true }> {\n    const { deviceId, camStreamId, kind, codec, resolution, label } = input\n\n    // ── derived kind: register metadata + validate source, no broker spawn ──\n    if (kind === 'derived') {\n      const { sourceCamStreamId, encodeProfile } = input\n      if (!sourceCamStreamId || !encodeProfile) {\n        throw new Error(\"kind: 'derived' requires sourceCamStreamId + encodeProfile\")\n      }\n\n      // Single-flight: if a derived entry with this name already exists,\n      // return immediately — callers should use retractCameraStream +\n      // republish to change encode parameters.\n      const existingDerived = this.cameraStreams.get(deviceId)?.get(camStreamId)\n      if (existingDerived) {\n        return { success: true as const }\n      }\n\n      // Validate that the source stream is registered for this device.\n      const sourceBrokerId = brokerIdFor(deviceId, sourceCamStreamId)\n      if (!this.brokers.has(sourceBrokerId)) {\n        throw new Error(`SOURCE_NOT_FOUND: ${sourceBrokerId}`)\n      }\n\n      // Register the derived stream metadata so getCameraStreamsForDevice\n      // surfaces it. The encode profile fields are mapped onto the standard\n      // CameraStreamInternal shape for UI / assignment purposes.\n      let derivedStreamMap = this.cameraStreams.get(deviceId)\n      if (!derivedStreamMap) {\n        derivedStreamMap = new Map()\n        this.cameraStreams.set(deviceId, derivedStreamMap)\n      }\n      const order = this.nextOrder(deviceId)\n      const deviceFeaturesArr: string[] = input.deviceFeatures ? [...input.deviceFeatures] : []\n      const derivedCam: CameraStreamInternal = {\n        camStreamId,\n        deviceId,\n        kind: 'derived',\n        ...(label !== undefined ? { label } : {}),\n        ...(codec !== undefined ? { codec } : encodeProfile.video.codec !== 'copy' ? { codec: encodeProfile.video.codec } : {}),\n        ...(resolution !== undefined\n          ? { resolution }\n          : encodeProfile.video.width !== undefined && encodeProfile.video.height !== undefined\n            ? { resolution: { width: encodeProfile.video.width, height: encodeProfile.video.height } }\n            : {}),\n        ...(encodeProfile.video.fps !== undefined ? { fps: encodeProfile.video.fps } : {}),\n        order,\n        deviceFeatures: deviceFeaturesArr,\n        autoEligible: input.autoEligible ?? false,\n      }\n      derivedStreamMap.set(camStreamId, derivedCam)\n      this.emitCamStreamsChanged(deviceId)\n      // TODO(Task N): spawn/attach ffmpeg transcode pipeline via\n      // buildTranscodeFfmpegArgs(encodeProfile) and wire source broker's\n      // onEncodedData → ffmpeg stdin → derived broker's pushEncodedPacket.\n      return { success: true as const }\n    }\n\n    // Detect a refreshed loopback transport: the publisher just re-ran\n    // its publish pipeline (typically in response to an\n    // `OnRequestStreamSourceRefresh` round-trip after a wake-up or an\n    // idle-tear-down) and bound a new URL. Capture it now so we can\n    // kick the broker's reconnect timer once the registry has\n    // absorbed the new entry — without this nudge the broker stays\n    // on its exponential-backoff schedule and may dial the stale URL\n    // for several more seconds before observing the change. The upsert\n    // overwrites the feature snapshot too — battery detection can happen\n    // post-creation (live probe), so the latest signal wins.\n    const existing = this.upsertSourceCameraStream(deviceId, input)\n    const urlRefreshed = existing !== undefined\n      && input.url !== undefined\n      && existing.url !== input.url\n\n    // Capture push-demand state BEFORE the rank so a re-published,\n    // already-assigned push stream re-triggers demand afterward.\n    const prePublishProfile = PUSH_KINDS.has(kind)\n      ? findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, camStreamId)\n      : null\n    const wasAlreadyAssigned = prePublishProfile !== null && existing !== undefined\n\n    await this.reconcileAutoAssignment(deviceId, 'publish')\n\n    if (wasAlreadyAssigned) {\n      const finalProfile = findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, camStreamId)\n      if (finalProfile !== null) {\n        this.emitCamStreamDemand(deviceId, camStreamId, finalProfile)\n      }\n    }\n\n    // Every published cam stream gets a live broker. Pre-buffer is\n    // OFF by default so idle streams don't pin RAM or wake battery\n    // cameras; the broker just sits ready for the first subscriber\n    // (WebRTC viewer, RTSP restream client, detection pipeline) to\n    // attach.\n    await this.ensureBroker(deviceId, camStreamId)\n\n    // The broker that survived this re-publish may have been waiting\n    // on its exponential-backoff timer — kick it so the next dial\n    // uses the fresh URL right away instead of after several more\n    // wasted retries against the stale loopback port.\n    if (urlRefreshed) {\n      const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n      broker?.kickReconnect()\n    }\n\n    // Drain any derived streams that were queued in pendingDerivedStreams\n    // because their source wasn't registered when loadPersistedDeviceOverrides\n    // ran. Now that the source broker has landed, materialise each pending\n    // derived in order. Only non-derived publishes (i.e. real source streams)\n    // can unblock pending deriveds — derived-on-derived is not supported.\n    const justRegisteredBrokerId = brokerIdFor(deviceId, camStreamId)\n    const pending = this.pendingDerivedStreams.get(justRegisteredBrokerId)\n    if (pending) {\n      this.pendingDerivedStreams.delete(justRegisteredBrokerId)\n      for (const p of pending) {\n        await this.publishCameraStream({\n          deviceId: p.deviceId,\n          camStreamId: `derived:${p.name}`,\n          kind: 'derived',\n          sourceCamStreamId: p.def.sourceCamStreamId,\n          encodeProfile: p.def.encodeProfile,\n        })\n      }\n    }\n\n    this.emitCamStreamsChanged(deviceId)\n    return { success: true as const }\n  }\n\n  /**\n   * Publish a device's FULL stream catalog atomically: upsert every\n   * source descriptor, rank the profiles ONCE over the complete set,\n   * then create each broker. Deterministic counterpart to looping\n   * `publishCameraStream` per descriptor (which persisted + emitted an\n   * intermediate, wrong assignment between each publish → the transient\n   * high/mid/low reshuffle during catalog republish). Derived descriptors\n   * (which need their source broker to already exist) fall back to the\n   * per-stream path AFTER the sources are live.\n   */\n  private async publishCatalog(\n    deviceId: number,\n    descriptors: readonly CamStreamDescriptor[],\n  ): Promise<void> {\n    const sources = descriptors.filter((d) => d.kind !== 'derived')\n    const derived = descriptors.filter((d) => d.kind === 'derived')\n\n    const streamMapBefore = this.cameraStreams.get(deviceId)\n    const reEmitDemandFor: string[] = []\n    for (const d of sources) {\n      const existed = streamMapBefore?.get(d.camStreamId) !== undefined\n      const wasAssigned = PUSH_KINDS.has(d.kind)\n        && findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, d.camStreamId) !== null\n      if (existed && wasAssigned) reEmitDemandFor.push(d.camStreamId)\n    }\n    const urlBefore = new Map<string, string | undefined>()\n    for (const d of sources) urlBefore.set(d.camStreamId, streamMapBefore?.get(d.camStreamId)?.url)\n\n    // (1) Upsert every source descriptor (no rank, no broker yet).\n    for (const d of sources) {\n      this.upsertSourceCameraStream(deviceId, d)\n    }\n\n    // (2) Rank ONCE over the complete set.\n    await this.reconcileAutoAssignment(deviceId, 'publish')\n\n    // (3) Re-emit demand for already-assigned push streams.\n    for (const camStreamId of reEmitDemandFor) {\n      const profile = findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, camStreamId)\n      if (profile !== null) this.emitCamStreamDemand(deviceId, camStreamId, profile)\n    }\n\n    // (4) Ensure a broker per source (+ url-refresh reconnect).\n    for (const d of sources) {\n      await this.ensureBroker(deviceId, d.camStreamId)\n      const prevUrl = urlBefore.get(d.camStreamId)\n      if (d.url !== undefined && prevUrl !== undefined && prevUrl !== d.url) {\n        this.brokers.get(brokerIdFor(deviceId, d.camStreamId))?.kickReconnect()\n      }\n    }\n\n    // (5) Derived descriptors: per-stream (source brokers now exist).\n    for (const d of derived) {\n      await this.publishCameraStream({ deviceId, ...d })\n    }\n\n    this.emitCamStreamsChanged(deviceId)\n  }\n\n  async retractCameraStream(input: {\n    deviceId: number\n    camStreamId: string\n  }): Promise<{ success: true }> {\n    const { deviceId, camStreamId } = input\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap?.has(camStreamId)) return { success: true as const }\n\n    const current = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    let newMap = { ...current.map }\n    for (const profile of CAM_PROFILE_ORDER) {\n      if (newMap[profile] === camStreamId) delete newMap[profile]\n    }\n    if (current.auto) {\n      const remaining = [...streamMap.values()]\n        .filter((s) => s.camStreamId !== camStreamId)\n        .sort((a, b) => a.order - b.order)\n      newMap = this.computeInitialAssignment(remaining)\n    }\n\n    // Apply the assignment BEFORE removing from streamMap so\n    // `countPushConsumers` can still identify the retracted stream as\n    // push-kind and emit the N→0 idle event.\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: current.auto, reason: 'retract' })\n\n    // Stream lifecycle is tied 1:1 to publish/retract — destroy the\n    // broker now that its source is gone.\n    await this.releaseBroker(deviceId, camStreamId)\n\n    streamMap.delete(camStreamId)\n    if (streamMap.size === 0) this.cameraStreams.delete(deviceId)\n\n    this.emitCamStreamsChanged(deviceId)\n    return { success: true as const }\n  }\n\n  // ── Catalog reconcile (PULL) ─────────────────────────────────────────\n  //\n  // The broker is the authority for its own cam-stream registry: it PULLS\n  // each camera's `stream-catalog` (via DeviceProxy) and reconciles, rather\n  // than relying on providers to push. This makes the broker self-heal after\n  // a restart — it re-derives the full registry from the providers. Driven by\n  // the host addon on start + a configurable poll + device / stream-params\n  // events.\n\n  /**\n   * Race a catalog pull against a timeout so a wedged provider handler\n   * can't leave the reconcile pending forever. Rejects (caught upstream →\n   * retry next tick) if the provider hasn't answered in time.\n   */\n  private async withCatalogTimeout<T>(promise: Promise<T>, deviceId: number): Promise<T> {\n    let timer: ReturnType<typeof setTimeout> | undefined\n    const timeout = new Promise<never>((_resolve, reject) => {\n      timer = setTimeout(\n        () => reject(new Error(`stream-catalog.getCatalog timed out for device ${String(deviceId)}`)),\n        12_000,\n      )\n    })\n    try {\n      return await Promise.race([promise, timeout])\n    } finally {\n      if (timer !== undefined) clearTimeout(timer)\n    }\n  }\n\n  /**\n   * Pull one camera's `stream-catalog` and reconcile its cam-streams: upsert\n   * every descriptor, retract any cam-stream no longer in the catalog. A pull\n   * that throws (provider mid-restart, no `stream-catalog` cap) is skipped —\n   * the poll retries. An EMPTY catalog is also skipped rather than tearing\n   * down (a provider probe can transiently return none); genuine removal flows\n   * through the device-unregistered path.\n   */\n  async reconcileDeviceCatalog(deviceId: number): Promise<void> {\n    if (!this.api) return\n    let descriptors: readonly CamStreamDescriptor[]\n    try {\n      // Device-scoped cap called from the broker (system context): pass the\n      // deviceId explicitly — the SDK link chain routes to the owning\n      // provider, hub-local or cross-node, transparently.\n      //\n      // Bounded with a timeout: a provider whose getCatalog handler wedges\n      // (e.g. a session-exhausted Reolink camera that never answers the\n      // Baichuan stream-options query) would otherwise leave this await\n      // pending forever, permanently stalling the reconcile and leaving the\n      // device's cam-stream registry — and its device-settings tab — empty.\n      // On timeout we fall into the catch and retry on the next poll tick.\n      descriptors = (await this.withCatalogTimeout(\n        this.api.streamCatalog.getCatalog.query({ deviceId }),\n        deviceId,\n      )) ?? []\n    } catch (err) {\n      this.logger.debug('reconcileDeviceCatalog: stream-catalog pull failed — will retry', {\n        tags: { deviceId }, meta: { error: errMsg(err) },\n      })\n      return\n    }\n    if (descriptors.length === 0) return\n\n    const keep = new Set(descriptors.map((d) => d.camStreamId))\n    // Atomic: upsert all + rank ONCE (no transient reshuffle). Replaces the\n    // per-descriptor publishCameraStream loop, which persisted an\n    // intermediate assignment between each stream.\n    await this.publishCatalog(deviceId, descriptors)\n    const existing = this.cameraStreams.get(deviceId)\n    if (existing) {\n      for (const camStreamId of [...existing.keys()]) {\n        if (!keep.has(camStreamId)) {\n          await this.retractCameraStream({ deviceId, camStreamId })\n        }\n      }\n    }\n  }\n\n  /**\n   * Enumerate every camera and reconcile its catalog. The reconcile backstop\n   * (poll cadence) and the start-up fetch. Per-device failures are isolated so\n   * one unreachable camera can't block the rest.\n   */\n  async reconcileAllCatalogs(): Promise<void> {\n    if (!this.api) return\n    let devices: readonly { id: number; type: string; isCamera: boolean; disabled: boolean }[]\n    try {\n      devices = await this.api.deviceManager.listAll.query({})\n    } catch (err) {\n      this.logger.debug('reconcileAllCatalogs: deviceManager.listAll failed — will retry', {\n        meta: { error: errMsg(err) },\n      })\n      return\n    }\n    const cameras = devices.filter((d) => d.isCamera || d.type === DeviceType.Camera)\n    await Promise.allSettled(cameras.map((d) => this.reconcileDeviceCatalog(d.id)))\n  }\n\n  /**\n   * Startup catalog pull: run a full reconcile, then a bounded number of extra\n   * passes while any camera still has an empty registry (a cold/slow provider\n   * may answer on a later attempt). Bounded so a genuinely-offline camera can't\n   * loop forever — the 30s poll remains the ongoing backstop.\n   */\n  async reconcileAllCatalogsAtStartup(maxPasses = 3, gapMs = 1500): Promise<void> {\n    for (let pass = 0; pass < maxPasses; pass++) {\n      await this.reconcileAllCatalogs()\n      if (!(await this.hasAnyCameraWithEmptyCatalog())) return\n      if (pass < maxPasses - 1) await new Promise((r) => setTimeout(r, gapMs))\n    }\n  }\n\n  private async hasAnyCameraWithEmptyCatalog(): Promise<boolean> {\n    if (!this.api) return false\n    let devices: readonly { id: number; type: string; isCamera: boolean; disabled: boolean }[]\n    try {\n      devices = await this.api.deviceManager.listAll.query({})\n    } catch {\n      return false\n    }\n    for (const d of devices) {\n      if ((d.isCamera || d.type === DeviceType.Camera) && !d.disabled) {\n        const streams = this.cameraStreams.get(d.id)\n        if (!streams || streams.size === 0) return true\n      }\n    }\n    return false\n  }\n\n  /**\n   * Drop a device's cam-streams entirely — fired on `device.unregistered`.\n   */\n  async retractDevice(deviceId: number): Promise<void> {\n    const existing = this.cameraStreams.get(deviceId)\n    if (!existing) return\n    for (const camStreamId of [...existing.keys()]) {\n      await this.retractCameraStream({ deviceId, camStreamId })\n    }\n  }\n\n  /**\n   * Lazily (re)create the broker instance for a `brokerId` if its cam-stream\n   * is known. The audio/frame subscribe paths call this so a consumer's\n   * retry RESURRECTS the broker after a restart — the video path already\n   * `ensureBroker`s, but audio/frame subscribers previously assumed an\n   * existing instance and failed forever once the process respawned.\n   * No-op when the brokerId is malformed or its cam-stream isn't (yet)\n   * reconciled into the registry (`ensureBroker` itself guards on that).\n   */\n  private async ensureBrokerForId(brokerId: string): Promise<void> {\n    if (this.brokers.has(brokerId)) return\n    const parsed = parseBrokerId(brokerId)\n    if (!parsed) return\n    await this.ensureBroker(parsed.deviceId, parsed.camStreamId)\n  }\n\n  // ── Cap methods: profile assignment ─────────────────────────────────\n\n  async assignProfile(input: {\n    deviceId: number\n    profile: CamProfile\n    camStreamId: string\n  }): Promise<{ success: true }> {\n    const { deviceId, profile, camStreamId } = input\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap || !streamMap.has(camStreamId)) {\n      throw new Error(\n        `assignProfile: camStreamId \"${camStreamId}\" is not published for device ${deviceId}`,\n      )\n    }\n    const current = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    const newMap = { ...current.map, [profile]: camStreamId }\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: false, reason: 'assign' })\n    return { success: true as const }\n  }\n\n  async unassignProfile(input: {\n    deviceId: number\n    profile: CamProfile\n  }): Promise<{ success: true }> {\n    const { deviceId, profile } = input\n    const current = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    if (current.map[profile] === undefined) {\n      // Still mark manual so subsequent publishes don't auto-reassign.\n      if (current.auto) {\n        await this.applyAssignmentUpdate(deviceId, current.map, { auto: false, reason: 'unassign' })\n      }\n      return { success: true as const }\n    }\n    const newMap = { ...current.map }\n    delete newMap[profile]\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: false, reason: 'unassign' })\n    return { success: true as const }\n  }\n\n  async restartProfile(input: {\n    deviceId: number\n    profile: CamProfile\n  }): Promise<{ success: boolean }> {\n    const { deviceId, profile } = input\n    const camStreamId = this.assignments.get(deviceId)?.map[profile]\n    if (!camStreamId) return { success: false }\n    await this.restartBroker(deviceId, camStreamId)\n    this.emitProfileSlotsChanged(deviceId)\n    return { success: true }\n  }\n\n  /**\n   * Recreate the broker for a cam stream — destroy the existing\n   * instance then re-create it from the same source. Used by\n   * `restartProfile`; lifecycle stays tied to publish/retract so the\n   * broker comes back unconditionally as long as the source is still\n   * published.\n   */\n  private async restartBroker(deviceId: number, camStreamId: string): Promise<void> {\n    if (!this.cameraStreams.get(deviceId)?.has(camStreamId)) return\n    await this.releaseBroker(deviceId, camStreamId)\n    await this.ensureBroker(deviceId, camStreamId)\n  }\n\n  // ── Cap methods: system-wide views ──────────────────────────────────\n\n  async listAllCameraStreams(): Promise<readonly CameraStream[]> {\n    const out: CameraStream[] = []\n    for (const streamMap of this.cameraStreams.values()) {\n      const sorted = [...streamMap.values()].sort((a, b) => a.order - b.order)\n      for (const s of sorted) out.push(this.toCameraStream(s))\n    }\n    return out\n  }\n\n  async listAllProfileSlots(): Promise<readonly ProfileSlot[]> {\n    const out: ProfileSlot[] = []\n    for (const deviceId of this.assignments.keys()) {\n      for (const slot of this.snapshotProfileSlots(deviceId)) out.push(slot)\n    }\n    return out\n  }\n\n  // ── Device-scoped facades (for camera-streams cap) ───────────────────\n\n  getCameraStreamsForDevice(deviceId: number): readonly CameraStream[] {\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) return []\n    return [...streamMap.values()]\n      .sort((a, b) => a.order - b.order)\n      .map((s) => this.toCameraStream(s))\n  }\n\n  getProfileSlotsForDevice(deviceId: number): readonly ProfileSlot[] {\n    // Return empty for devices the broker has never touched — avoids\n    // phantom `unassigned` slots for every numeric id in the universe.\n    if (!this.cameraStreams.has(deviceId) && !this.assignments.has(deviceId)) return []\n    return this.snapshotProfileSlots(deviceId)\n  }\n\n  // ── Internal: live broker lookup (not a cap method) ─────────────────\n\n  /**\n   * Resolve the live `StreamBroker` instance for a broker id. In-process\n   * only — a `StreamBroker` is not tRPC-serialisable, so this is reachable\n   * solely from same-process callers (device-scoped providers, the WebRTC\n   * server). Cross-process frame access goes through the shm `subscribeFrames`\n   * surface; cross-process stats through `getBrokerStats` / `listClients`.\n   */\n  async getBroker(input: { brokerId: string }): Promise<IStreamBroker | null> {\n    return this.brokers.get(input.brokerId) ?? null\n  }\n\n  // ── Cap methods: broker runtime (stats + client inventory) ──────────\n\n  // ── Cap methods: shared-memory frame-handle plane (Phase 5 / D9) ─────\n\n  /**\n   * Open a `FrameHandle` subscription on a broker — the handle-based\n   * replacement for the live-object `onDecodedFrame` callback. Routed over\n   * tRPC, so a consumer in a different process can subscribe; the consumer\n   * then polls `pullFrameHandles` and feeds each handle to a\n   * `FrameRingReader`.\n   */\n  async subscribeFrames(input: SubscribeFramesInput): Promise<SubscribeFramesResult> {\n    await this.ensureBrokerForId(input.brokerId)\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) {\n      throw new Error(`stream-broker: no broker for \"${input.brokerId}\"`)\n    }\n    const result = await broker.subscribeFrameHandles({\n      format: input.format,\n      maxFps: input.maxFps,\n      tag: input.tag,\n    })\n    this.frameSubscriptionBroker.set(result.subscriptionId, input.brokerId)\n    return result\n  }\n\n  /** Drain `FrameHandle`s for a subscription opened via `subscribeFrames`. */\n  async pullFrameHandles(input: { subscriptionId: string; maxCount: number }): Promise<readonly FrameHandle[]> {\n    const brokerId = this.frameSubscriptionBroker.get(input.subscriptionId)\n    if (!brokerId) return []\n    const broker = this.brokers.get(brokerId)\n    if (!broker) return []\n    return broker.pullFrameHandles(input.subscriptionId, input.maxCount)\n  }\n\n  /**\n   * Drop every `frameSubscriptionBroker` + `audioSubscriptionBroker` entry\n   * pointing at `brokerId`. Called on broker teardown so a destroyed broker\n   * leaves no dangling `subscriptionId → brokerId` rows (a slow consumer that\n   * never calls `unsubscribeFrames` / `unsubscribeAudioChunks` would otherwise\n   * leak a map entry per stream).\n   */\n  private sweepFrameSubscriptions(brokerId: string): void {\n    for (const [subscriptionId, mappedBrokerId] of this.frameSubscriptionBroker) {\n      if (mappedBrokerId === brokerId) {\n        this.frameSubscriptionBroker.delete(subscriptionId)\n      }\n    }\n    for (const [subscriptionId, mappedBrokerId] of this.audioSubscriptionBroker) {\n      if (mappedBrokerId === brokerId) {\n        this.audioSubscriptionBroker.delete(subscriptionId)\n      }\n    }\n  }\n\n  /** Release a frame-handle subscription. */\n  async unsubscribeFrames(input: { subscriptionId: string }): Promise<{ released: boolean }> {\n    const brokerId = this.frameSubscriptionBroker.get(input.subscriptionId)\n    if (!brokerId) return { released: false }\n    this.frameSubscriptionBroker.delete(input.subscriptionId)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) return { released: false }\n    const released = await broker.unsubscribeFrameHandles(input.subscriptionId)\n    return { released }\n  }\n\n  // ── Cap methods: decoded audio-chunk plane (Phase 5 / D9) ────────────\n\n  /**\n   * Open a decoded audio-chunk subscription on a broker — the poll-based,\n   * tRPC-reachable replacement for the live-object `onDecodedAudioChunk`\n   * callback. The consumer then polls `pullAudioChunks` and feeds each\n   * `DecodedAudioChunk` to its downstream audio logic. Audio chunks are tiny\n   * so their bytes travel inline on the RPC wire — no shared memory.\n   */\n  async subscribeAudioChunks(input: SubscribeAudioChunksInput): Promise<SubscribeAudioChunksResult> {\n    await this.ensureBrokerForId(input.brokerId)\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) {\n      throw new Error(`stream-broker: no broker for \"${input.brokerId}\"`)\n    }\n    const result = broker.subscribeAudioChunks({ tag: input.tag })\n    this.audioSubscriptionBroker.set(result.subscriptionId, input.brokerId)\n    return result\n  }\n\n  /** Drain `DecodedAudioChunk`s for a subscription opened via `subscribeAudioChunks`. */\n  async pullAudioChunks(\n    input: { subscriptionId: string; maxCount: number },\n  ): Promise<readonly DecodedAudioChunk[]> {\n    const brokerId = this.audioSubscriptionBroker.get(input.subscriptionId)\n    if (!brokerId) return []\n    const broker = this.brokers.get(brokerId)\n    if (!broker) return []\n    return broker.pullAudioChunks(input.subscriptionId, input.maxCount)\n  }\n\n  /** Release an audio-chunk subscription. */\n  async unsubscribeAudioChunks(input: { subscriptionId: string }): Promise<{ released: boolean }> {\n    const brokerId = this.audioSubscriptionBroker.get(input.subscriptionId)\n    if (!brokerId) return { released: false }\n    this.audioSubscriptionBroker.delete(input.subscriptionId)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) return { released: false }\n    const released = broker.unsubscribeAudioChunks(input.subscriptionId)\n    return { released }\n  }\n\n  async getBrokerStats(input: { brokerId: string }): Promise<BrokerStats> {\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) {\n      return {\n        status: 'stopped',\n        inputFps: 0,\n        decodeFps: 0,\n        encodedSubscribers: 0,\n        decodedSubscribers: 0,\n        uptimeMs: 0,\n        bitrateKbps: 0,\n        idrIntervalMs: 0,\n        totalBytes: 0,\n        packetCount: 0,\n        rtspClients: 0,\n        pipeClients: 0,\n        preBufferSec: 0,\n        preBufferMs: 0,\n        preBufferPackets: 0,\n        decoderNodeId: null,\n        audio: null,\n      }\n    }\n    return broker.getStats()\n  }\n\n  async listClients(input: { brokerId: string }): Promise<import('@camstack/types').BrokerClients> {\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) {\n      return { rtsp: [], decoded: [], audio: [], encoded: [], pipeClients: 0, encodedSubscribers: 0 }\n    }\n    return broker.listClients()\n  }\n\n  async killClient(input: {\n    brokerId: string\n    channel: 'rtsp' | 'decoded' | 'audio'\n    handle: string\n  }): Promise<{ killed: boolean }> {\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) return { killed: false }\n    return { killed: broker.killClient(input.channel, input.handle) }\n  }\n\n  async rotateDecodersOnNode(agentNodeId: string, reason: string): Promise<number> {\n    let rotated = 0\n    for (const broker of this.brokers.values()) {\n      // Only touch brokers with a shm frame-plane decoder session on this\n      // agent — the broker rotates every matching per-format session itself.\n      const onNode = broker.getDecoderNodeId()?.split('/')[0] === agentNodeId\n      if (!onNode) continue\n      try {\n        await broker.rotateDecoderSession(reason, agentNodeId)\n        rotated++\n      } catch (err) {\n        this.logger.warn('decoder rotation failed', {\n          meta: { brokerId: broker.deviceId, agentNodeId, reason, error: errMsg(err) },\n        })\n      }\n    }\n    return rotated\n  }\n\n  /** Raw broker instances — internal use only. */\n  getBrokerInstances(): readonly StreamBroker[] {\n    return [...this.brokers.values()]\n  }\n\n  async destroyAll(): Promise<void> {\n    if (this.streamHealthTimer) {\n      clearInterval(this.streamHealthTimer)\n      this.streamHealthTimer = undefined\n    }\n    if (this.transcodeManager) {\n      this.transcodeManager.shutdownAll()\n      this.transcodeManager = null\n    }\n    const stopPromises = [...this.brokers.values()].map((broker) => broker.stop())\n    await Promise.all(stopPromises)\n    this.brokers.clear()\n    this.streamHealthByBroker.clear()\n    this.frameSubscriptionBroker.clear()\n    this.audioSubscriptionBroker.clear()\n    await this.rtspServer.stop()\n  }\n\n  // ── Stream health watchdog ──────────────────────────────────────────\n  //\n  // Polls every active broker every STREAM_HEALTH_POLL_MS. A broker is\n  // considered stale when its last video packet is older than\n  // STREAM_STALE_TIMEOUT_MS. State transitions emit `stream.online` /\n  // `stream.offline` on the event bus; payload includes the camStreamId\n  // currently bound to the profile slot (the operator-visible \"key\").\n  //\n  // Brokers whose slot is unassigned (no camStreamId) skip emission —\n  // there is no published stream to report on.\n\n  private startStreamHealthWatchdog(): void {\n    if (this.streamHealthTimer) return\n    this.streamHealthTimer = setInterval(() => this.evaluateStreamHealth(), STREAM_HEALTH_POLL_MS)\n  }\n\n  private evaluateStreamHealth(): void {\n    if (!this.eventBus) return\n    const now = Date.now()\n    for (const [brokerId, broker] of this.brokers) {\n      const parsed = this.parseBrokerId(brokerId)\n      if (!parsed) continue\n      const { deviceId, camStreamId } = parsed\n      const lastPacketAt = broker.getLastPacketAt()\n      const wasHealthy = this.streamHealthByBroker.get(brokerId)\n      const isHealthy = lastPacketAt > 0 && now - lastPacketAt <= STREAM_STALE_TIMEOUT_MS\n      if (wasHealthy === isHealthy) continue\n      this.streamHealthByBroker.set(brokerId, isHealthy)\n      this.emitStreamHealth(isHealthy, {\n        deviceId,\n        camStreamId,\n        profile: findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, camStreamId),\n        brokerId,\n        sourceType: broker.getActiveSourceType(),\n        lastPacketAt,\n        reason: isHealthy ? (wasHealthy === undefined ? 'first-packet' : 'recovered') : 'stale-timeout',\n      })\n    }\n  }\n\n  private emitStreamHealth(\n    online: boolean,\n    payload: {\n      deviceId: number\n      camStreamId: string\n      profile: CamProfile | null\n      brokerId: string\n      sourceType: string\n      lastPacketAt: number\n      reason: string\n    },\n  ): void {\n    const bus = this.eventBus\n    if (!bus) return\n    const category = online ? 'stream.online' : 'stream.offline'\n    bus.emit({\n      id: `${category}-${payload.brokerId}-${Date.now()}`,\n      timestamp: new Date(),\n      category,\n      source: { type: 'stream-broker', id: payload.brokerId },\n      data: payload,\n    })\n    // Slot status flipped — re-snapshot and propagate to the device's\n    // `camera-streams` runtime-state slice so consumers (RTSP autonomous\n    // online tracking, UI dashboards) get push semantics without having\n    // to subscribe to the per-stream events themselves. SLICE-ONLY:\n    // we deliberately do NOT fire `onProfileSlotsChanged` here — the\n    // orchestrator listens to that event to restart detection, but a\n    // health flip (lazy-dial suspend / on-demand resume) is not a slot\n    // mutation and must not tear down the runner attachment. Without\n    // this guard each `suspend()`/`start()` cycle would race a\n    // `stopDetection`+`startDetection` pair, surfacing as\n    // `PipelineCameraUnassigned`+`PipelineCameraAssigned` pairs in the\n    // UI events panel.\n    void this.writeCameraStreamsSlice(payload.deviceId, this.snapshotProfileSlots(payload.deviceId))\n  }\n\n  /**\n   * Decompose a brokerId `${deviceId}/${camStreamId}` back into its\n   * components. `camStreamId` may contain `:` (e.g. `native:main`), so\n   * we split on the FIRST `/` only — everything after is the cam stream\n   * id verbatim.\n   */\n  private parseBrokerId(brokerId: string): { deviceId: number; camStreamId: string } | null {\n    const idx = brokerId.indexOf('/')\n    if (idx <= 0) return null\n    const deviceId = Number(brokerId.slice(0, idx))\n    const camStreamId = brokerId.slice(idx + 1)\n    if (!Number.isFinite(deviceId) || camStreamId.length === 0) return null\n    return { deviceId, camStreamId }\n  }\n\n  async startRtspServer(port: number = 8554): Promise<void> {\n    await this.rtspServer.start(port)\n    this.logger.info('RTSP restream server listening', { meta: { port: this.rtspServer.getPort() } })\n  }\n\n  // ── Cap methods: pre-buffer ─────────────────────────────────────────\n\n  async setPreBufferDuration(input: { brokerId: string; seconds: number }): Promise<void> {\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) return\n    broker.setPreBufferDuration(input.seconds)\n  }\n\n  async getPreBufferInfo(input: {\n    brokerId: string\n  }): Promise<{ configuredSec: number; bufferedMs: number; packetCount: number }> {\n    const broker = this.brokers.get(input.brokerId)\n    if (!broker) return { configuredSec: 0, bufferedMs: 0, packetCount: 0 }\n    const stats = broker.getStats()\n    return {\n      configuredSec: stats.preBufferSec,\n      bufferedMs: stats.preBufferMs,\n      packetCount: stats.preBufferPackets,\n    }\n  }\n\n  // ── Cap methods: stream URLs ────────────────────────────────────────\n\n  async getStreamUrl(input: { streamId: string; format: StreamFormat }): Promise<{ url: string }> {\n    const prefix = input.streamId.split('/')[0]\n    if (!prefix) return { url: '' }\n    const deviceId = Number(prefix)\n    if (!Number.isFinite(deviceId)) return { url: '' }\n    for (const restreamer of this.getLiveRestreamers()) {\n      const resources = restreamer.getExposedResources(deviceId)\n      const match = resources.find((r) => r.format === input.format)\n      if (match) return { url: match.value }\n    }\n    return { url: '' }\n  }\n\n  // ── Cap methods: RTSP restream ──────────────────────────────────────\n\n  async getRtspPort(): Promise<number> {\n    return this.rtspProvider.getRtspPort()\n  }\n\n  async getAllRtspEntries(input?: { hostname?: string }): Promise<readonly RtspRestreamEntry[]> {\n    return this.rtspProvider.getAllEntries(input?.hostname).map((e) => this.enrichRtspEntry(e))\n  }\n\n  /** Device-scoped RTSP entry fetch — backs `cameraStreams.getRtspEntries`. */\n  getRtspEntriesForDevice(deviceId: number, hostname?: string): readonly RtspRestreamEntry[] {\n    return this.rtspProvider.getEntriesForDevice(deviceId, hostname).map((e) => this.enrichRtspEntry(e))\n  }\n\n  /** Device-scoped PROFILE RTSP entry fetch — backs `cameraStreams.getProfileRtspEntries`. */\n  getProfileRtspEntriesForDevice(deviceId: number, hostname?: string): readonly ProfileRtspEntry[] {\n    if (!this.assignments.has(deviceId)) return []\n    // Ensure the profile→source aliasing is current before reading URLs.\n    this.syncProfileRestreamers(deviceId)\n    const assignmentMap = this.assignments.get(deviceId)?.map ?? {}\n    const streamMap = this.cameraStreams.get(deviceId)\n    return buildProfileRtspEntries({\n      deviceId,\n      assignmentMap,\n      resolveEntry: (profileBrokerId) => this.rtspProvider.getEntry(profileBrokerId, hostname),\n      resolveMeta: (camStreamId) => {\n        const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n        const cam = streamMap?.get(camStreamId)\n        const codec = broker?.getStats()?.codec ?? cam?.codec\n        return {\n          ...(codec !== undefined ? { codec } : {}),\n          ...(cam?.resolution !== undefined ? { resolution: cam.resolution } : {}),\n        }\n      },\n    })\n  }\n\n  /**\n   * Keep one RTSP restreamer per assigned profile slot, aliasing the\n   * assigned source broker's `RtspRestreamer` + a warm-up that drives\n   * the source broker's single on-demand pull. So a profile consumer\n   * (HAP/Alexa/recording) and a WebRTC consumer of the same profile\n   * converge on ONE physical camera pull. Diff-based + idempotent: only\n   * (re)registers when a profile's backing source changes, so a stable\n   * token survives across reads (no mid-session token rotation).\n   */\n  private syncProfileRestreamers(deviceId: number): void {\n    const assignmentMap = this.assignments.get(deviceId)?.map ?? {}\n    const actions = computeProfileRestreamerActions({\n      deviceId,\n      assignmentMap,\n      sourceBrokerExists: (camStreamId) => this.brokers.has(brokerIdFor(deviceId, camStreamId)),\n      currentBindings: this.profileRestreamerBindings,\n    })\n    if (actions.register.length === 0 && actions.unregister.length === 0) return\n    for (const profileBrokerId of actions.unregister) {\n      this.rtspServer.unregisterRestreamer(profileBrokerId)\n      this.profileRestreamerBindings.delete(profileBrokerId)\n    }\n    for (const { profileBrokerId, camStreamId } of actions.register) {\n      const sourceBrokerId = brokerIdFor(deviceId, camStreamId)\n      const sourceBroker = this.brokers.get(sourceBrokerId)\n      if (!sourceBroker) continue\n      const existingToken = this.persistedTokens.get(profileBrokerId)\n      // Warm-up: subscribing a no-op encoded consumer to the SOURCE\n      // broker keeps it dialed for the lifetime of an RTSP session, so\n      // the profile DESCRIBE doesn't race a cold broker.\n      const warmUp = (): (() => void) => sourceBroker.onEncodedData(() => { /* warm-up no-op */ })\n      this.rtspServer.registerRestreamer(\n        profileBrokerId,\n        sourceBroker.getRtspRestreamer(),\n        warmUp,\n        existingToken,\n      )\n      this.profileRestreamerBindings.set(profileBrokerId, sourceBrokerId)\n      if (!existingToken) this.rtspProvider.persistTokens()\n    }\n  }\n\n  /**\n   * Enrich a bare RTSP entry with the source camStream's `codec` /\n   * `resolution`. The provider doesn't know — it only owns tokens + URL\n   * minting — but the consumer-side pickers\n   * (`pickPreferredRtspEntry`, the exporter's `resolveSourceParams`)\n   * need that shape to resolve `streamPreference: 'auto'` to the slot\n   * whose source is closest to the consumer's target. Falls back to\n   * the bare entry when the brokerId points at a stream the manager\n   * never registered (legacy + race scenarios) — the picker then\n   * silently degrades to first-enabled.\n   */\n  private enrichRtspEntry(entry: RtspRestreamEntry): RtspRestreamEntry {\n    const parsed = parseBrokerId(entry.brokerId)\n    if (!parsed) return entry\n    const stream = this.cameraStreams.get(parsed.deviceId)?.get(parsed.camStreamId)\n    if (!stream) return entry\n    return {\n      ...entry,\n      ...(stream.codec !== undefined ? { codec: stream.codec } : {}),\n      ...(stream.resolution !== undefined ? { resolution: stream.resolution } : {}),\n    }\n  }\n\n  async getRtspEntry(input: { brokerId: string; hostname?: string }): Promise<RtspRestreamEntry | null> {\n    return this.rtspProvider.getEntry(input.brokerId, input.hostname)\n  }\n\n  async regenerateRtspToken(input: { brokerId: string }): Promise<string | null> {\n    this.logger.info('regenerateRtspToken called', { meta: { brokerId: input.brokerId } })\n    const result = this.rtspProvider.regenerateToken(input.brokerId)\n    if (result) {\n      this.logger.info('regenerateRtspToken succeeded', {\n        meta: { brokerId: input.brokerId, newTokenPrefix: result.slice(0, 8) },\n      })\n    } else {\n      this.logger.warn('regenerateRtspToken failed — broker not found', {\n        meta: { brokerId: input.brokerId },\n      })\n    }\n    return result\n  }\n\n  async setRtspEnabled(input: { brokerId: string; enabled: boolean }): Promise<void> {\n    this.rtspProvider.setEnabled(input.brokerId, input.enabled)\n  }\n\n  async isRtspEnabled(input: { brokerId: string }): Promise<boolean> {\n    return this.rtspProvider.isEnabled(input.brokerId)\n  }\n\n  /**\n   * Fan-out a placeholder reason to every broker registered for a\n   * given device. Internal — drives the runtime-state-driven\n   * placeholder behaviour: the broker manager subscribes to events\n   * that reflect device state (battery sleep, online/offline,\n   * disabled), translates them to a `PlaceholderReason`, and pushes\n   * to the per-device broker set. Providers stay oblivious — they\n   * only mutate their own runtime state.\n   */\n  private applyPlaceholderReasonToDevice(deviceId: number, reason: PlaceholderReason): void {\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) return\n    for (const camStreamId of streamMap.keys()) {\n      const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n      broker?.setPlaceholderReason(reason)\n    }\n  }\n\n  // ── Device overrides (kept for per-profile pre-buffer tuning) ──────\n\n  setDeviceOverridePersister(cb: (overrides: ReadonlyMap<number, DeviceOverride>) => void): void {\n    this.deviceOverridePersister = cb\n  }\n\n  loadPersistedDeviceOverrides(overrides: ReadonlyMap<number, DeviceOverride>): void {\n    this.deviceOverrides.clear()\n    for (const [k, v] of overrides) this.deviceOverrides.set(k, v)\n\n    // Queue any persisted derived streams for boot-time hydration.\n    // publishCameraStream({kind:'derived'}) requires the source broker to\n    // already be registered, which is not guaranteed at boot. For each\n    // derived definition whose source is already up, we attempt the publish\n    // immediately (inside the IIFE). If SOURCE_NOT_FOUND is thrown, the\n    // definition is recorded in pendingDerivedStreams, keyed by the source\n    // brokerId. publishCameraStream drains the queue when the source lands.\n    void (async () => {\n      for (const [deviceId, override] of overrides) {\n        if (!override.derivedStreams) continue\n        for (const [name, def] of Object.entries(override.derivedStreams)) {\n          // Unconfigured placeholder (no source yet) — keep it persisted but\n          // don't try to materialize a broker for it.\n          if (!def.sourceCamStreamId) continue\n          try {\n            await this.publishCameraStream({\n              deviceId,\n              camStreamId: `derived:${name}`,\n              kind: 'derived',\n              sourceCamStreamId: def.sourceCamStreamId,\n              encodeProfile: def.encodeProfile,\n            })\n          } catch (err) {\n            if (err instanceof Error && err.message.includes('SOURCE_NOT_FOUND')) {\n              const sourceBrokerId = brokerIdFor(deviceId, def.sourceCamStreamId)\n              const list = this.pendingDerivedStreams.get(sourceBrokerId) ?? []\n              list.push({ deviceId, name, def })\n              this.pendingDerivedStreams.set(sourceBrokerId, list)\n            }\n          }\n        }\n      }\n    })()\n  }\n\n  /**\n   * Effective pre-buffer duration for a brokerId's profile.\n   *\n   * Priority: per-profile override → legacy single-value override →\n   * battery-aware default (0 when the device has BatteryOperated and\n   * no override) → addon default.\n   */\n  getEffectivePreBufferSec(\n    deviceId: number,\n    camStreamId: string,\n    addonDefault: number,\n    forceForDetection = false,\n  ): number {\n    const override = this.deviceOverrides.get(deviceId)\n    const perStream = override?.preBuffer?.[camStreamId]\n    if (perStream) {\n      const enabled = perStream.enabled || forceForDetection\n      return enabled ? perStream.seconds : 0\n    }\n    if (override?.preBufferSecOverride !== undefined) return override.preBufferSecOverride\n    if (this.deviceHasFeature(deviceId, DeviceFeature.BatteryOperated)) return 0\n    return addonDefault\n  }\n\n  /**\n   * Per-device streaming debug flag — controls verbose info-level\n   * logging across the broker, WebRTC server, and session for one\n   * device. Off by default.\n   */\n  isStreamingDebug(deviceId: number): boolean {\n    return this.deviceOverrides.get(deviceId)?.streamingDebug === true\n  }\n\n  /**\n   * Per-device \"Advanced WebRTC debug logging\" flag — when true, every\n   * WebRTC session created for this device runs with `debug = true` so\n   * the full negotiation/media lifecycle is logged. Off by default.\n   * Independent of `isStreamingDebug`.\n   */\n  isWebrtcDebug(deviceId: number): boolean {\n    return this.deviceOverrides.get(deviceId)?.webrtcDebug === true\n  }\n\n  /**\n   * Returns an EncodeProfile-shaped snapshot of the camStream's *probed*\n   * source parameters — the codec / resolution / fps the publisher\n   * advertised when it called `publishCameraStream`, optionally enriched\n   * with live broker stats (bitrateKbps) when available.\n   *\n   * Used as `widgetConfig.sourceParams` so the FFmpeg parameters widget\n   * can render a read-only \"Source (probed)\" chips card next to the\n   * override editor. Returns `undefined` when the camStream isn't\n   * registered or has no advertised media shape (the widget then\n   * omits the card rather than showing fictitious zeros).\n   *\n   * Note: audio is reported as `'passthrough'` because the broker's\n   * source side doesn't probe audio shape today — the widget's audio\n   * chip will read \"passthrough\" rather than guess a sample rate.\n   */\n  getCameraStreamProbedParams(deviceId: number, camStreamId: string): EncodeProfile | undefined {\n    const stream = this.cameraStreams.get(deviceId)?.get(camStreamId)\n    if (stream === undefined) return undefined\n    const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n    const stats = broker?.getStats()\n    const rawCodec = (stream.codec ?? stats?.codec ?? '').toLowerCase()\n    const codec: VideoEncode['codec'] = rawCodec === 'h265' ? 'h265' : 'h264'\n    const video: VideoEncode = {\n      codec,\n      ...(stream.resolution?.width !== undefined ? { width: stream.resolution.width } : {}),\n      ...(stream.resolution?.height !== undefined ? { height: stream.resolution.height } : {}),\n      ...(stream.fps !== undefined ? { fps: stream.fps } : {}),\n      ...(stats?.bitrateKbps !== undefined && stats.bitrateKbps > 0 ? { bitrateKbps: Math.round(stats.bitrateKbps) } : {}),\n    }\n    // Nothing meaningful was advertised — omit so the widget hides the\n    // probed card instead of showing only a codec chip.\n    if (video.width === undefined && video.height === undefined && video.fps === undefined) {\n      return undefined\n    }\n    return { video, audio: 'passthrough' as const }\n  }\n\n  // ── Device-details aggregator contribution ─────────────────────────\n\n  async getDeviceSettingsContribution(input: {\n    deviceId: number\n  }): Promise<ConfigUISchemaWithValues | null> {\n    const deviceId = input.deviceId\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap || streamMap.size === 0) return null\n    const assignment = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    const streams = [...streamMap.values()].sort((a, b) => a.order - b.order)\n    const override = this.deviceOverrides.get(deviceId)\n\n    const streamOptions = [\n      { value: '', label: 'Not assigned' },\n      ...streams.map((s) => ({\n        value: s.camStreamId,\n        label: this.labelForDropdown(s),\n      })),\n    ]\n\n    const PROFILES: ReadonlyArray<{ key: CamProfile; label: string }> = [\n      { key: 'high', label: 'High' },\n      { key: 'mid', label: 'Mid' },\n      { key: 'low', label: 'Low' },\n    ]\n\n    // ── Sub-tab \"General\" — three profile→stream selectors + derived stream names ──────\n    const generalTab: ConfigSubTabDefinitionWithValue = {\n      id: 'general',\n      label: 'General',\n      fields: [\n        ...PROFILES.map(({ key: profile, label }) => ({\n          type: 'select' as const,\n          key: `streamProfile:${profile}`,\n          label: `${label} Quality`,\n          description: `Which camera stream serves the \"${label}\" profile slot`,\n          options: streamOptions,\n          span: 2 as const,\n          value: assignment.map[profile] ?? '',\n        })),\n        {\n          type: 'text' as const,\n          key: 'derivedStreamNames',\n          label: 'Derived stream names',\n          description: 'Add a name to create a new derived stream; configure its source and encode params in its tab below.',\n          placeholder: 'e.g. web-720p',\n          span: 2 as const,\n          value: Object.keys(override?.derivedStreams ?? {}),\n          multiple: {\n            min: 0,\n            max: 10,\n            addLabel: 'Add derived stream',\n            itemLabel: 'Stream ${n}',\n          },\n        },\n      ],\n    }\n\n    // ── One sub-tab per stream source with its broker settings ─────\n    // Every published cam stream owns a broker for as long as the\n    // source is published — there is no \"Active\" toggle. Pre-buffer\n    // is OFF by default and opt-in per stream.\n    const streamTabs: ConfigSubTabDefinitionWithValue[] = streams.map((stream) => {\n      const camStreamId = stream.camStreamId\n      const brokerId = brokerIdFor(deviceId, camStreamId)\n      const profile = findProfileForStream(assignment.map, camStreamId)\n      const rtspEnabled = this.rtspProvider.isEnabled(brokerId)\n      const rtspEntry = this.rtspProvider.getEntry(brokerId)\n      const perStream = override?.preBuffer?.[camStreamId]\n      const preBufferEnabled = perStream?.enabled ?? false\n      const preBufferSec = perStream?.seconds ?? this.defaultPreBufferSec\n\n      const fields: ConfigFieldWithValue[] = [\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: `rtspEnabled:${camStreamId}`,\n          label: 'RTSP Restream',\n          description: 'Expose this stream via the RTSP restream server',\n          default: true,\n          value: rtspEnabled,\n        },\n        {\n          type: 'text' as const,\n          key: `rtspUrl:${camStreamId}`,\n          label: 'RTSP URL',\n          readonlyField: true,\n          span: 2 as const,\n          value: rtspEntry?.url ?? 'inactive',\n          actions: [\n            { action: 'copy-value', icon: 'copy', tooltip: 'Copy URL to clipboard' },\n            {\n              action: 'regenerate-rtsp-token',\n              icon: 'refresh-cw',\n              tooltip: 'Regenerate RTSP token',\n              variant: 'danger' as const,\n              confirmMessage: `Rotate the RTSP token for ${stream.label ?? camStreamId}? Current URL will stop working immediately.`,\n            },\n          ],\n        },\n        {\n          type: 'text' as const,\n          key: `rtspMutedUrl:${camStreamId}`,\n          label: 'RTSP URL (muted)',\n          description: 'Video-only stream — no audio track',\n          readonlyField: true,\n          span: 2 as const,\n          value: rtspEntry?.url ? `${rtspEntry.url}/muted` : 'inactive',\n          actions: [\n            { action: 'copy-value', icon: 'copy', tooltip: 'Copy muted URL to clipboard' },\n          ],\n        },\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: `preBufferEnabled:${camStreamId}`,\n          label: 'Pre-buffer',\n          description:\n            'Keep a rolling buffer for instant playback on events. Forced on when detection uses this stream.',\n          default: false,\n          value: preBufferEnabled,\n        },\n        {\n          type: 'slider' as const,\n          key: `preBufferSec:${camStreamId}`,\n          label: 'Pre-buffer Duration',\n          min: 0,\n          max: 30,\n          step: 1,\n          unit: 's',\n          default: this.defaultPreBufferSec,\n          showValue: true,\n          value: preBufferSec,\n        },\n      ]\n\n      const tabBadge = profile\n        ? PROFILES.find((p) => p.key === profile)?.label\n        : undefined\n      const tabLabel = stream.label ?? camStreamId\n      const def: ConfigSubTabDefinitionWithValue = {\n        id: `stream:${camStreamId}`,\n        label: tabLabel,\n        fields,\n        ...(tabBadge ? { badge: tabBadge } : {}),\n      }\n      return def\n    })\n\n    const sourceStreamOptions = streams.map((s) => ({ value: s.camStreamId, label: s.label ?? s.camStreamId }))\n\n    const derivedTabs: ConfigSubTabDefinitionWithValue[] = Object.entries(override?.derivedStreams ?? {}).map(([name, def]) => ({\n      id: `derived:${name}`,\n      label: name,\n      fields: [\n        {\n          type: 'select' as const,\n          key: `derivedSource:${name}`,\n          label: 'Source stream',\n          description: 'Which source camStream this derived stream reads from.',\n          options: sourceStreamOptions,\n          span: 2 as const,\n          value: def.sourceCamStreamId,\n        },\n        {\n          type: 'widget' as const,\n          key: `derivedEncodeWidget:${name}`,\n          label: 'Encode parameters',\n          span: 2 as const,\n          widgetId: 'stream-broker/ffmpeg-params',\n          widgetConfig: {\n            fieldKey: `derivedProfile:${name}`,\n            label: 'Encode parameters',\n            // The derived stream's \"source\" is its upstream camStream —\n            // surface that camStream's probed shape so the operator sees\n            // what the transcode pipeline is consuming.\n            sourceParams: this.getCameraStreamProbedParams(deviceId, def.sourceCamStreamId),\n            initialValue: def.encodeProfile,\n            writerCapName: 'stream-broker',\n            writerAddonId: 'stream-broker',\n          },\n        },\n        // Toggle pair: Pre-buffer + RTSP restream render side by side\n        // (span 1 each). Order matters — adjacent span-1 fields share a\n        // row in the two-column form grid.\n        {\n          type: 'boolean' as const,\n          key: `derivedPrebufferEnabled:${name}`,\n          label: 'Pre-buffer',\n          style: 'switch' as const,\n          span: 1 as const,\n          default: false,\n          value: def.preBuffer?.enabled ?? false,\n        },\n        {\n          type: 'boolean' as const,\n          key: `derivedRtspEnabled:${name}`,\n          label: 'RTSP restream',\n          style: 'switch' as const,\n          span: 1 as const,\n          default: false,\n          value: def.rtspEnabled ?? false,\n        },\n        // Full-width duration slider, shown only when Pre-buffer is on —\n        // no orphan slider when the feature is disabled.\n        {\n          type: 'slider' as const,\n          key: `derivedPrebufferSec:${name}`,\n          label: 'Pre-buffer Duration',\n          min: 0,\n          max: 30,\n          step: 1,\n          unit: 's',\n          showValue: true,\n          span: 2 as const,\n          showWhen: { field: `derivedPrebufferEnabled:${name}`, equals: true },\n          default: this.defaultPreBufferSec,\n          value: def.preBuffer?.seconds ?? this.defaultPreBufferSec,\n        },\n      ],\n    }))\n\n    const streamBrokerSection: ConfigSectionWithValues = {\n      id: 'stream-broker-streams',\n      title: '',\n      tab: 'stream-broker',\n      // Hoist out of the Config tab into a dedicated device-detail\n      // top-level \"Stream Broker\" tab — DeviceDetail discovers the tab\n      // from `location: 'top-tab'` sections, the way recording/zones/ptz\n      // do. Without this the section nests as a Config inner-tab.\n      location: 'top-tab',\n      order: 10,\n      fields: [\n        {\n          type: 'sub-tabs' as const,\n          key: 'streamBroker:streams',\n          label: '',\n          span: 2 as const,\n          tabs: [generalTab, ...streamTabs, ...derivedTabs],\n        },\n      ],\n    }\n\n    const diagnosticsSection: ConfigSectionWithValues = {\n      id: 'stream-broker-diagnostics',\n      title: 'Diagnostics',\n      tab: 'stream-broker',\n      // Same top-tab as the streams section (sections sharing a tab id\n      // stack in `order` sequence inside that one tab body).\n      location: 'top-tab',\n      order: 100,\n      fields: [\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: 'streamingDebug',\n          label: 'Streaming debug logs',\n          description:\n            'Enable verbose info-level logs for the broker, WebRTC server, and H.265 repacketizer for this device. Off by default; turn on while diagnosing playback issues.',\n          default: false,\n          value: override?.streamingDebug === true,\n        },\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: 'webrtcDebug',\n          label: 'Advanced WebRTC debug logging',\n          description:\n            'Log EVERY step of the WebRTC pipeline for this device: the local ICE candidates we put in the answer (with type host/srflx/relay + address), the remote offer candidates, DTLS state transitions, ICE nomination, codec negotiation (offered vs negotiated), and first video/audio RTP. Use to diagnose hard cases (Alexa streaming, A/V sync). Off by default — turn on only while debugging.',\n          default: false,\n          value: override?.webrtcDebug === true,\n        },\n      ],\n    }\n\n    return { sections: [streamBrokerSection, diagnosticsSection] }\n  }\n\n  async getDeviceLiveContribution(_input: {\n    deviceId: number\n  }): Promise<ConfigUISchemaWithValues | null> {\n    return null\n  }\n\n  async applyDeviceSettingsPatch(input: {\n    deviceId: number\n    patch: Record<string, unknown>\n  }): Promise<{ success: true }> {\n    const deviceId = input.deviceId\n    if (!this.cameraStreams.has(deviceId)) {\n      this.logger.debug('applyDeviceSettingsPatch: unknown deviceId — no-op', {\n        tags: { deviceId },\n      })\n      return { success: true as const }\n    }\n\n    let overrideDirty = false\n    const nextOverride: DeviceOverride = { ...(this.deviceOverrides.get(deviceId) ?? {}) }\n\n    for (const [fieldKey, value] of Object.entries(input.patch)) {\n      // Profile → camStreamId selectors live in the Assignment sub-tab\n      // (`streamProfile:<profile>`).\n      if (fieldKey.startsWith('streamProfile:')) {\n        const profile = fieldKey.slice('streamProfile:'.length) as CamProfile\n        if (!CAM_PROFILE_ORDER.includes(profile)) continue\n        const newCamId = typeof value === 'string' && value !== '' ? value : null\n        if (newCamId === null) {\n          await this.unassignProfile({ deviceId, profile })\n        } else {\n          await this.assignProfile({ deviceId, profile, camStreamId: newCamId })\n        }\n        continue\n      }\n      if (fieldKey.startsWith('rtspEnabled:')) {\n        const camStreamId = fieldKey.slice('rtspEnabled:'.length)\n        const brokerId = brokerIdFor(deviceId, camStreamId)\n        this.rtspProvider.setEnabled(brokerId, Boolean(value))\n        continue\n      }\n      if (fieldKey === 'preBufferSecOverride') {\n        overrideDirty = true\n        if (value === null || value === undefined) {\n          delete nextOverride.preBufferSecOverride\n        } else if (typeof value === 'number' && Number.isFinite(value)) {\n          nextOverride.preBufferSecOverride = value\n        }\n        continue\n      }\n      if (fieldKey.startsWith('preBufferEnabled:')) {\n        const camStreamId = fieldKey.slice('preBufferEnabled:'.length)\n        overrideDirty = true\n        const pb = { ...(nextOverride.preBuffer ?? {}) }\n        const current = pb[camStreamId] ?? this.defaultStreamPreBuffer(deviceId, camStreamId)\n        const enabled = Boolean(value)\n        pb[camStreamId] = { ...current, enabled }\n        nextOverride.preBuffer = pb\n        const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n        if (broker) {\n          broker.setPreBufferDuration(enabled ? pb[camStreamId]?.seconds ?? this.defaultPreBufferSec : 0)\n          broker.setPreBufferEnabled(enabled)\n        }\n        continue\n      }\n      if (fieldKey.startsWith('preBufferSec:')) {\n        const camStreamId = fieldKey.slice('preBufferSec:'.length)\n        const seconds =\n          typeof value === 'number' && Number.isFinite(value) ? value : this.defaultPreBufferSec\n        overrideDirty = true\n        const pb = { ...(nextOverride.preBuffer ?? {}) }\n        const current = pb[camStreamId] ?? this.defaultStreamPreBuffer(deviceId, camStreamId)\n        pb[camStreamId] = { ...current, seconds }\n        nextOverride.preBuffer = pb\n        const broker = this.brokers.get(brokerIdFor(deviceId, camStreamId))\n        if (broker && pb[camStreamId]?.enabled !== false) {\n          broker.setPreBufferDuration(seconds)\n        }\n        continue\n      }\n      if (fieldKey === 'streamingDebug') {\n        overrideDirty = true\n        const next = value === null || value === undefined ? undefined : Boolean(value)\n        if (next === undefined) {\n          delete nextOverride.streamingDebug\n        } else {\n          nextOverride.streamingDebug = next\n        }\n        // Push to every active broker for this device immediately so\n        // the toggle takes effect without restarting the stream.\n        for (const [bid, broker] of this.brokers) {\n          if (bid.startsWith(`${deviceId}/`)) {\n            broker.setStreamingDebug(next === true)\n          }\n        }\n        continue\n      }\n      if (fieldKey === 'webrtcDebug') {\n        overrideDirty = true\n        const next = value === null || value === undefined ? undefined : Boolean(value)\n        if (next === undefined) {\n          delete nextOverride.webrtcDebug\n        } else {\n          nextOverride.webrtcDebug = next\n        }\n        // No live broker push needed: each new WebRTC session reads\n        // `isWebrtcDebug(deviceId)` at creation (see\n        // `device-scoped-providers.ts`), so the next session for this\n        // device picks the flag up. Existing sessions keep their flag.\n        continue\n      }\n\n      if (fieldKey === 'derivedStreamNames' && Array.isArray(value)) {\n        overrideDirty = true\n        const names = value.filter((v): v is string => typeof v === 'string' && v.length > 0)\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        // Add missing entries with empty profile placeholders\n        for (const n of names) {\n          if (!next[n]) {\n            next[n] = {\n              sourceCamStreamId: '',\n              encodeProfile: { video: { codec: 'h264' as const }, audio: 'passthrough' as const },\n            }\n          }\n        }\n        // Remove names no longer in the list\n        for (const existing of Object.keys(next)) {\n          if (!names.includes(existing)) delete next[existing]\n        }\n        nextOverride.derivedStreams = next\n        // Tear down derived brokers that were removed from the list\n        for (const existing of Object.keys(nextOverride.derivedStreams === next ? {} : (nextOverride.derivedStreams ?? {}))) {\n          if (!names.includes(existing)) {\n            void this.retractCameraStream({ deviceId, camStreamId: `derived:${existing}` })\n          }\n        }\n        // Materialise any derived streams whose source is already available\n        for (const n of names) {\n          const def = next[n]\n          if (!def || !def.sourceCamStreamId) continue\n          const sourceBrokerId = brokerIdFor(deviceId, def.sourceCamStreamId)\n          if (!this.brokers.has(sourceBrokerId)) continue\n          void this.publishCameraStream({\n            deviceId,\n            camStreamId: `derived:${n}`,\n            kind: 'derived',\n            sourceCamStreamId: def.sourceCamStreamId,\n            encodeProfile: def.encodeProfile,\n          }).catch((err: unknown) => {\n            if (err instanceof Error && err.message.includes('SOURCE_NOT_FOUND')) return\n            this.logger.warn('applyDeviceSettingsPatch: derivedStreamNames materialise failed', {\n              tags: { deviceId }, meta: { name: n, error: err instanceof Error ? err.message : String(err) },\n            })\n          })\n        }\n        continue\n      }\n\n      if (fieldKey.startsWith('derivedSource:')) {\n        const name = fieldKey.slice('derivedSource:'.length)\n        overrideDirty = true\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        if (!next[name]) {\n          next[name] = { sourceCamStreamId: '', encodeProfile: { video: { codec: 'h264' as const }, audio: 'passthrough' as const } }\n        }\n        next[name] = { ...next[name], sourceCamStreamId: typeof value === 'string' ? value : '' }\n        nextOverride.derivedStreams = next\n        continue\n      }\n\n      if (fieldKey.startsWith('derivedProfile:')) {\n        const name = fieldKey.slice('derivedProfile:'.length)\n        overrideDirty = true\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        if (!next[name]) {\n          next[name] = { sourceCamStreamId: '', encodeProfile: { video: { codec: 'h264' as const }, audio: 'passthrough' as const } }\n        }\n        next[name] = { ...next[name], encodeProfile: EncodeProfileSchema.parse(value) }\n        nextOverride.derivedStreams = next\n        continue\n      }\n\n      if (fieldKey.startsWith('derivedPrebufferEnabled:')) {\n        const name = fieldKey.slice('derivedPrebufferEnabled:'.length)\n        overrideDirty = true\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        if (next[name]) {\n          next[name] = {\n            ...next[name],\n            preBuffer: { ...(next[name].preBuffer ?? { enabled: false, seconds: this.defaultPreBufferSec }), enabled: Boolean(value) },\n          }\n          nextOverride.derivedStreams = next\n        }\n        continue\n      }\n\n      if (fieldKey.startsWith('derivedPrebufferSec:')) {\n        const name = fieldKey.slice('derivedPrebufferSec:'.length)\n        overrideDirty = true\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        if (next[name] && typeof value === 'number') {\n          next[name] = {\n            ...next[name],\n            preBuffer: { ...(next[name].preBuffer ?? { enabled: false, seconds: 0 }), seconds: value },\n          }\n          nextOverride.derivedStreams = next\n        }\n        continue\n      }\n\n      if (fieldKey.startsWith('derivedRtspEnabled:')) {\n        const name = fieldKey.slice('derivedRtspEnabled:'.length)\n        overrideDirty = true\n        const next = { ...(nextOverride.derivedStreams ?? {}) }\n        if (next[name]) {\n          next[name] = { ...next[name], rtspEnabled: Boolean(value) }\n          nextOverride.derivedStreams = next\n        }\n        continue\n      }\n\n    }\n\n    if (overrideDirty) {\n      const hasContent =\n        nextOverride.preBufferSecOverride !== undefined ||\n        (nextOverride.preBuffer !== undefined && Object.keys(nextOverride.preBuffer).length > 0) ||\n        nextOverride.streamingDebug !== undefined ||\n        nextOverride.webrtcDebug !== undefined ||\n        (nextOverride.derivedStreams !== undefined && Object.keys(nextOverride.derivedStreams).length > 0)\n      if (!hasContent) {\n        this.deviceOverrides.delete(deviceId)\n      } else {\n        this.deviceOverrides.set(deviceId, nextOverride)\n      }\n      this.deviceOverridePersister?.(this.deviceOverrides)\n    }\n    return { success: true as const }\n  }\n\n  // ── Private helpers ─────────────────────────────────────────────────\n\n  private nextOrder(deviceId: number): number {\n    const next = (this.streamOrderSeq.get(deviceId) ?? 0) + 1\n    this.streamOrderSeq.set(deviceId, next)\n    return next\n  }\n\n  private toCameraStream(s: CameraStreamInternal): CameraStream {\n    const { order: _order, deviceFeatures: _df, ...rest } = s\n    return rest\n  }\n\n  private deviceHasFeature(deviceId: number, feature: string): boolean {\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) return false\n    for (const s of streamMap.values()) {\n      if (s.deviceFeatures.includes(feature)) return true\n    }\n    return false\n  }\n\n  private labelForDropdown(s: CameraStreamInternal): string {\n    const base = s.label ?? s.camStreamId\n    if (s.resolution) return `${base} (${s.resolution.width}×${s.resolution.height})`\n    return base\n  }\n\n  /**\n   * Compute the canonical profile map for a given ordered stream list.\n   *\n   *   1 stream  → { mid }\n   *   2 streams → { high, low }  (higher resolution → high)\n   *   3+ streams → { high, mid, low }\n   *\n   * Ties on resolution fall back to publish order (stream.order asc).\n   * Streams with no resolution sort below those with one — publish\n   * order breaks ties within each rank.\n   *\n   * Streams marked `autoEligible: false` are skipped — they remain\n   * published (visible + manually assignable via `assignProfile`) but\n   * never become the default. If every stream is non-eligible the\n   * algorithm degrades gracefully and considers them all (so a device\n   * with only ineligible streams still gets some default assignment).\n   */\n  private computeInitialAssignment(\n    streams: readonly CameraStreamInternal[],\n  ): Partial<Record<CamProfile, string>> {\n    if (streams.length === 0) return {}\n    const eligible = streams.filter((s) => s.autoEligible)\n    const pool = eligible.length > 0 ? eligible : streams\n    const ranked = [...pool].sort((a, b) => {\n      const aPx = a.resolution ? a.resolution.width * a.resolution.height : -1\n      const bPx = b.resolution ? b.resolution.width * b.resolution.height : -1\n      if (aPx !== bPx) return bPx - aPx\n      return a.order - b.order\n    })\n    if (ranked.length === 1) return { mid: ranked[0]!.camStreamId }\n    if (ranked.length === 2) {\n      return { high: ranked[0]!.camStreamId, low: ranked[1]!.camStreamId }\n    }\n    return {\n      high: ranked[0]!.camStreamId,\n      mid: ranked[1]!.camStreamId,\n      low: ranked[2]!.camStreamId,\n    }\n  }\n\n  /**\n   * Recompute the auto profile assignment for a device over its CURRENT\n   * complete cam-stream set and apply it once. Extracted from\n   * `publishCameraStream` so a multi-stream catalog publish can upsert\n   * every stream first and rank ONCE (no transient intermediate\n   * assignment). When the device's assignment is operator-locked\n   * (`auto === false`) the existing map is re-applied unchanged.\n   */\n  private async reconcileAutoAssignment(\n    deviceId: number,\n    reason: 'publish' | 'retract',\n  ): Promise<void> {\n    const current = this.assignments.get(deviceId)\n    const streams = [...(this.cameraStreams.get(deviceId)?.values() ?? [])]\n      .sort((a, b) => a.order - b.order)\n    const isAuto = current?.auto ?? true\n    const newMap: Partial<Record<CamProfile, string>> =\n      !current || (isAuto && this.wouldChangeAutoMap(current.map, streams))\n        ? this.computeInitialAssignment(streams)\n        : current.map\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: isAuto, reason })\n  }\n\n  private wouldChangeAutoMap(\n    current: Partial<Record<CamProfile, string>>,\n    streams: readonly CameraStreamInternal[],\n  ): boolean {\n    const computed = this.computeInitialAssignment(streams)\n    for (const p of CAM_PROFILE_ORDER) {\n      if (current[p] !== computed[p]) return true\n    }\n    return false\n  }\n\n  /**\n   * Core transition: diff `current.map` vs `newMap`, then translate the\n   * profile-level diff to demand/idle events for push-kind streams.\n   * Brokers are NOT touched here — every published cam stream owns\n   * its broker for as long as the source exists, regardless of which\n   * profile slot points at it. Profile reassignments are pure\n   * metadata changes from the broker's point of view.\n   */\n  private async applyAssignmentUpdate(\n    deviceId: number,\n    newMap: Partial<Record<CamProfile, string>>,\n    opts: { auto: boolean; reason: 'publish' | 'retract' | 'assign' | 'unassign' },\n  ): Promise<void> {\n    const current = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    const nextAssignment: DeviceAssignment = { map: newMap, auto: opts.auto }\n\n    // Pre-change demand view: which push cam streams have active consumers?\n    const oldPushConsumers = this.countPushConsumers(current.map, deviceId)\n    const newPushConsumers = this.countPushConsumers(newMap, deviceId)\n\n    let anyChange = false\n    for (const profile of CAM_PROFILE_ORDER) {\n      if (current.map[profile] !== newMap[profile]) {\n        anyChange = true\n        break\n      }\n    }\n\n    this.assignments.set(deviceId, nextAssignment)\n\n    // Demand transitions for push-kind streams.\n    for (const [camId, prevCount] of oldPushConsumers) {\n      if ((newPushConsumers.get(camId) ?? 0) === 0 && prevCount > 0) {\n        this.emitCamStreamIdle(deviceId, camId)\n      }\n    }\n    for (const [camId, nextCount] of newPushConsumers) {\n      if ((oldPushConsumers.get(camId) ?? 0) === 0 && nextCount > 0) {\n        // Figure out which profile triggered the demand (first one in canonical order).\n        let triggerProfile: CamProfile = 'mid'\n        for (const p of CAM_PROFILE_ORDER) {\n          if (newMap[p] === camId) { triggerProfile = p; break }\n        }\n        this.emitCamStreamDemand(deviceId, camId, triggerProfile)\n      }\n    }\n\n    if (anyChange || current.auto !== nextAssignment.auto) {\n      this.persistProfileMap()\n      this.emitProfileSlotsChanged(deviceId)\n    } else if (opts.reason === 'publish') {\n      // Re-publish with unchanged assignment (e.g. provider restart): still\n      // emit so the orchestrator can re-dispatch detection on provider reconnect.\n      this.emitProfileSlotsChanged(deviceId)\n    }\n    // Re-point profile-keyed restreamers at their (possibly new) sources.\n    // On a FIRST publish this runs before ensureBroker creates the source\n    // broker, so profiles whose source isn't live yet are skipped here and\n    // wired by the syncProfileRestreamers call inside the publish path once\n    // the broker exists — don't remove that second call thinking it redundant.\n    this.syncProfileRestreamers(deviceId)\n  }\n\n  private countPushConsumers(\n    map: Partial<Record<CamProfile, string>>,\n    deviceId: number,\n  ): Map<string, number> {\n    const out = new Map<string, number>()\n    const streamMap = this.cameraStreams.get(deviceId)\n    for (const profile of CAM_PROFILE_ORDER) {\n      const camId = map[profile]\n      if (!camId) continue\n      const cam = streamMap?.get(camId)\n      if (!cam || !PUSH_KINDS.has(cam.kind)) continue\n      out.set(camId, (out.get(camId) ?? 0) + 1)\n    }\n    return out\n  }\n\n  private persistProfileMap(): void {\n    this.profileMapPersister?.(this.assignments)\n  }\n\n  private buildSourceFromCamStream(cam: CameraStreamInternal): StreamSource {\n    const label = cam.label ?? cam.camStreamId\n    const codec = cam.codec !== undefined ? { videoCodec: cam.codec } : {}\n    // Battery-flagged devices want the stall watchdog relaxed (Reolink\n    // Argus and friends intentionally go silent between motion events).\n    // Derive the per-source flag from the device-level feature list so\n    // publishers don't have to carry their own per-stream flag.\n    const stall = cam.deviceFeatures.includes(DeviceFeature.BatteryOperated)\n      ? { allowStall: true as const }\n      : {}\n    // Pre-compose pass-through metadata so the publisher's opaque payload\n    // (e.g. `pull-rfc4571` SDP) reaches the broker reader unchanged.\n    const passthroughMeta = cam.metadata ?? {}\n    switch (cam.kind) {\n      case 'push-annexb':\n        return { type: 'push', url: '', ...codec, ...stall, metadata: { kind: 'push-annexb', label, ...passthroughMeta } }\n      case 'pull-rtsp':\n        return { type: 'rtsp', url: cam.url ?? '', ...codec, metadata: { kind: 'pull-rtsp', label, ...passthroughMeta } }\n      case 'pull-rtmp':\n        return { type: 'rtmp', url: cam.url ?? '', ...codec, metadata: { kind: 'pull-rtmp', label, ...passthroughMeta } }\n      case 'pull-http':\n        return { type: 'http-mjpeg', url: cam.url ?? '', ...codec, metadata: { kind: 'pull-http', label, ...passthroughMeta } }\n      case 'pull-rfc4571':\n        return { type: 'rfc4571', url: cam.url ?? '', ...codec, ...stall, metadata: { kind: 'pull-rfc4571', label, ...passthroughMeta } }\n      case 'derived':\n        // Derived streams bypass ensureBroker — their source plane is the\n        // transcode pipeline, not a pull/push transport. This branch is\n        // unreachable in practice (publishCameraStream short-circuits before\n        // ensureBroker is called for derived entries) but satisfies the\n        // exhaustive switch so the type-checker accepts the full CamStreamKind\n        // union.\n        throw new Error(`buildSourceFromCamStream called for derived stream '${cam.camStreamId}' — derived streams must not reach ensureBroker`)\n      default: {\n        const _exhaustive: never = cam.kind\n        throw new Error(`Unknown cam stream kind: ${String(_exhaustive)}`)\n      }\n    }\n  }\n\n  /**\n   * Acquire (or create) the broker for a (deviceId, camStreamId) pair.\n   * Idempotent — returns the existing broker when one is already\n   * registered. Tied to publish lifecycle: `publishCameraStream` calls\n   * this once per stream; `retractCameraStream` releases it.\n   */\n  private async ensureBroker(\n    deviceId: number,\n    camStreamId: string,\n  ): Promise<void> {\n    const cam = this.cameraStreams.get(deviceId)?.get(camStreamId)\n    if (!cam) return\n    const brokerId = brokerIdFor(deviceId, camStreamId)\n    if (this.brokers.has(brokerId)) return\n\n    const brokerLogger = this.logger\n      .child(camStreamId)\n      .withTags({ deviceId, camStreamId })\n    const source = this.buildSourceFromCamStream(cam)\n    const broker = new StreamBroker(\n      brokerId,\n      this.buildDecoderCapApi(deviceId),\n      brokerLogger,\n      buildAudioCodecCapApi(this.api),\n    )\n    broker.setRestreamers(this.restreamers)\n    this.applyPreBufferConfig(broker, deviceId, camStreamId, cam)\n    const override = this.deviceOverrides.get(deviceId)\n    if (override?.streamingDebug) broker.setStreamingDebug(true)\n\n    // Hook the broker into the cam-stream registry: every dial\n    // attempt re-derives the source so a re-publish (e.g. Reolink\n    // refreshing its rfc4571 URL) is observed transparently — the\n    // source provider closure re-resolves a fresh URL on every dial.\n    broker.setSourceProvider(() => {\n      const latest = this.cameraStreams.get(deviceId)?.get(camStreamId)\n      return latest ? this.buildSourceFromCamStream(latest) : null\n    })\n    broker.setRequestSourceRefresh(() => {\n      this.emitRequestStreamSourceRefresh(deviceId, camStreamId, brokerId)\n    })\n\n    await broker.start(source)\n    this.brokers.set(brokerId, broker)\n\n    const existingToken = this.persistedTokens.get(brokerId)\n    // Pass a warm-up closure that subscribes a no-op encoded-data\n    // consumer for the duration of an RTSP restream session. Triggers\n    // `checkDemand` → upstream dial when an external client (ffprobe,\n    // recording sidecar, NVR) connects, so the DESCRIBE doesn't race a\n    // cold broker and return an empty SDP. The handle is released when\n    // the RTSP socket closes, letting the broker idle-suspend normally.\n    const warmUp = (): (() => void) => broker.onEncodedData(() => { /* warm-up no-op */ })\n    this.rtspServer.registerRestreamer(brokerId, broker.getRtspRestreamer(), warmUp, existingToken)\n    if (!existingToken) this.rtspProvider.persistTokens()\n\n    await this.fanOutRegistration(deviceId, brokerId, cam, broker)\n    // A new source broker may back one or more assigned profiles — wire\n    // their profile-keyed restreamers now that the source exists.\n    this.syncProfileRestreamers(deviceId)\n  }\n\n  /**\n   * Tear down the broker for a `(deviceId, camStreamId)` pair. No-op\n   * when no broker exists. Called by `retractCameraStream` and by the\n   * teardown half of `restartBroker`.\n   */\n  private async releaseBroker(\n    deviceId: number,\n    camStreamId: string,\n  ): Promise<void> {\n    const brokerId = brokerIdFor(deviceId, camStreamId)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) return\n\n    await this.fanOutUnregistration(deviceId, brokerId)\n    this.rtspServer.unregisterRestreamer(brokerId)\n    // Emit terminal stream.offline so subscribers can clear their\n    // aggregate before the broker disappears. Skip if the broker never\n    // became healthy (no transition to report).\n    const wasHealthy = this.streamHealthByBroker.get(brokerId)\n    if (wasHealthy && this.eventBus) {\n      this.emitStreamHealth(false, {\n        deviceId,\n        camStreamId,\n        profile: findProfileForStream(this.assignments.get(deviceId)?.map ?? {}, camStreamId),\n        brokerId,\n        sourceType: broker.getActiveSourceType(),\n        lastPacketAt: broker.getLastPacketAt(),\n        reason: 'broker-destroyed',\n      })\n    }\n    this.streamHealthByBroker.delete(brokerId)\n    await broker.stop()\n    this.brokers.delete(brokerId)\n    this.sweepFrameSubscriptions(brokerId)\n    this.rtspProvider.persistTokens()\n    // The source for one or more profiles just disappeared — drop the\n    // now-dangling profile-keyed restreamers.\n    this.syncProfileRestreamers(deviceId)\n  }\n\n  /**\n   * Default `StreamPreBuffer` for a brand-new override entry — mirrors\n   * the UI-side default in `getDeviceSettingsContribution`. Pre-buffer\n   * is OFF by default for every stream. Operators opt in per-stream\n   * via the device-settings sub-tabs; detection / recording paths\n   * pass `forceForDetection` to `getEffectivePreBufferSec` when they\n   * need a buffered window irrespective of the operator toggle.\n   */\n  private defaultStreamPreBuffer(_deviceId: number, _camStreamId: string): StreamPreBuffer {\n    return { enabled: false, seconds: this.defaultPreBufferSec }\n  }\n\n  /**\n   * Apply the effective pre-buffer config to a fresh broker instance.\n   * Default behaviour is OFF for every cam stream — keeps RAM and the\n   * upstream reader idle until the operator opts in or a detection /\n   * recording subscriber attaches.\n   */\n  private applyPreBufferConfig(\n    broker: StreamBroker,\n    deviceId: number,\n    camStreamId: string,\n    _cam: CameraStreamInternal,\n  ): void {\n    const override = this.deviceOverrides.get(deviceId)\n    const perStream = override?.preBuffer?.[camStreamId]\n    if (perStream) {\n      broker.setPreBufferEnabled(perStream.enabled)\n      broker.setPreBufferDuration(perStream.enabled ? perStream.seconds : 0)\n      return\n    }\n    // Default OFF for every camera (battery or AC) when the operator\n    // hasn't toggled the per-stream pre-buffer setting. The broker\n    // stays strictly on-demand: it dials upstream only when a consumer\n    // (WebRTC session, RTSP restream client, recording subscriber,\n    // detection plane) attaches, and idle-suspends after the consumer\n    // count drops back to zero. Operators opt in to \"always warm\"\n    // explicitly per stream via the Streams settings panel. Reverts an\n    // earlier always-on default that was surprising — every camera\n    // started dialing upstream at boot even when nothing was watching.\n    broker.setPreBufferEnabled(false)\n    broker.setPreBufferDuration(0)\n  }\n\n  private async fanOutRegistration(\n    deviceId: number,\n    brokerId: string,\n    cam: CameraStreamInternal,\n    broker: StreamBroker,\n  ): Promise<void> {\n    const log = this.logger.withTags({ deviceId, brokerId, camStreamId: cam.camStreamId })\n    const localStreamUrl = broker.getLocalStreamUrl()\n    const codec = 'h264'\n    const label = cam.label ?? brokerId\n\n    for (const restreamer of this.getLiveRestreamers()) {\n      try {\n        await restreamer.registerDevice(deviceId, [\n          { streamId: brokerId, label, codec, type: 'video' as const, sourceUrl: localStreamUrl },\n        ])\n      } catch (err) {\n        log.error('Failed to register with restreamer', {\n          meta: { restreamerId: restreamer.id, error: errMsg(err) },\n        })\n      }\n    }\n\n    // Register broker with the built-in WebRTC server (direct encoded\n    // packet subscription — no ffmpeg, no RTSP double-hop).\n    if (this.webrtcServer) {\n      this.webrtcServer.registerBroker(brokerId, broker)\n    }\n  }\n\n  private async fanOutUnregistration(deviceId: number, brokerId: string): Promise<void> {\n    const log = this.logger.withTags({ deviceId, brokerId })\n    for (const restreamer of this.getLiveRestreamers()) {\n      try {\n        await restreamer.unregisterDevice(deviceId)\n      } catch (err) {\n        log.error('Failed to unregister from restreamer', {\n          meta: { restreamerId: restreamer.id, error: errMsg(err) },\n        })\n      }\n    }\n    // Unregister from built-in WebRTC server.\n    if (this.webrtcServer) {\n      this.webrtcServer.unregisterBroker(brokerId)\n    }\n  }\n\n  private brokerStatusToSlotStatus(status: BrokerStats['status']): ProfileSlotStatus {\n    // BrokerStatus has 'stopped' (broker tore down cleanly) which the\n    // ProfileSlotStatus schema does not model — collapse to 'idle' since\n    // a stopped-but-assigned slot from the consumer's POV is equivalent\n    // to idle awaiting restart.\n    if (status === 'stopped') return 'idle'\n    return status\n  }\n\n  private snapshotProfileSlots(deviceId: number): ProfileSlot[] {\n    const assignment = this.assignments.get(deviceId) ?? { map: {}, auto: true }\n    const streamMap = this.cameraStreams.get(deviceId)\n    const slots: ProfileSlot[] = []\n    for (const profile of CAM_PROFILE_ORDER) {\n      const camId = assignment.map[profile] ?? null\n      // brokerId is keyed by camStreamId (the canonical broker identity);\n      // for unassigned slots there's no live broker, so we surface a\n      // synthetic profile-shaped key purely so consumers that ignore\n      // the field for unassigned slots keep working unchanged.\n      const brokerId = camId ? brokerIdFor(deviceId, camId) : makeProfileBrokerId(deviceId, profile)\n      const broker = camId ? this.brokers.get(brokerId) : undefined\n      const stats = broker?.getStats()\n      const cam = camId ? streamMap?.get(camId) : undefined\n      const status: ProfileSlotStatus = camId\n        ? stats ? this.brokerStatusToSlotStatus(stats.status) : 'idle'\n        : 'unassigned'\n      // Codec resolution: prefer the live broker's detected codec\n      // (`stats.codec` — populated after first frame), fall back to the\n      // provider-declared `cam.codec`. Reolink cameras don't always\n      // populate cam.codec at registration, so without this fallback\n      // every slot reported codec=undefined and the UI rendered \"—\".\n      const resolvedCodec = stats?.codec ?? cam?.codec\n      const slot: ProfileSlot = {\n        deviceId,\n        profile,\n        brokerId,\n        sourceCamStreamId: camId,\n        status,\n        ...(cam?.resolution !== undefined ? { resolution: cam.resolution } : {}),\n        ...(resolvedCodec !== undefined ? { codec: resolvedCodec } : {}),\n        ...(stats !== undefined ? { preBufferSec: stats.preBufferSec } : {}),\n      }\n      slots.push(slot)\n    }\n    return slots\n  }\n\n  // ── Event emission (via IEventBus) ──────────────────────────────────\n\n  private emitCapEvent(category: string, deviceId: number, data: Record<string, unknown>): void {\n    const bus = this.eventBus\n    if (!bus) return\n    bus.emit({\n      id: `cap-${Date.now()}-${Math.random().toString(36).slice(2, 8)}`,\n      timestamp: new Date(),\n      source: { type: 'device', id: deviceId, addonId: 'stream-broker', deviceId },\n      category,\n      data,\n    })\n  }\n\n  private emitCamStreamDemand(deviceId: number, camStreamId: string, profile: CamProfile): void {\n    this.emitCapEvent(EventCategory.StreamBrokerOnCamStreamDemand, deviceId, { deviceId, camStreamId, profile })\n  }\n\n  private emitCamStreamIdle(deviceId: number, camStreamId: string): void {\n    this.emitCapEvent(EventCategory.StreamBrokerOnCamStreamIdle, deviceId, { deviceId, camStreamId })\n  }\n\n  private emitRequestStreamSourceRefresh(deviceId: number, camStreamId: string, brokerId: string): void {\n    this.emitCapEvent(EventCategory.StreamBrokerOnRequestStreamSourceRefresh, deviceId, {\n      deviceId,\n      camStreamId,\n      brokerId,\n    })\n  }\n\n  private emitCamStreamsChanged(deviceId: number): void {\n    const camStreams = this.getCameraStreamsForDevice(deviceId)\n    this.emitCapEvent('camera-streams.onCamStreamsChanged', deviceId, { deviceId, camStreams })\n  }\n\n  private emitProfileSlotsChanged(deviceId: number): void {\n    const profileSlots = this.snapshotProfileSlots(deviceId)\n    this.emitCapEvent('camera-streams.onProfileSlotsChanged', deviceId, { deviceId, profileSlots })\n    void this.writeCameraStreamsSlice(deviceId, profileSlots)\n  }\n\n  /**\n   * Mirror per-device runtime state into the `camera-streams` slice.\n   * Fire-and-forget — failures land on the debug log and don't impact\n   * broker behavior (the slice is observability + RTSP autonomous-online\n   * source, both fault-tolerant). Mirrors the same `setCapSlice` pattern\n   * pipeline-runner uses for the `motion` slice.\n   */\n  private async writeCameraStreamsSlice(deviceId: number, slots: readonly ProfileSlot[]): Promise<void> {\n    const api = this.api\n    if (!api) return\n    const slotStatuses: { high?: ProfileSlotStatus; mid?: ProfileSlotStatus; low?: ProfileSlotStatus } = {}\n    const slotErrors: { high?: string; mid?: string; low?: string } = {}\n    let online = false\n    for (const slot of slots) {\n      if (slot.status !== 'unassigned') {\n        slotStatuses[slot.profile] = slot.status\n        if (slot.errorMessage) slotErrors[slot.profile] = slot.errorMessage\n        if (slot.status === 'streaming') online = true\n      }\n    }\n    try {\n      await api.deviceState.setCapSlice.mutate({\n        deviceId,\n        capName: 'camera-streams',\n        slice: { online, slotStatuses, slotErrors, lastChangedAt: Date.now() },\n      })\n    } catch (err) {\n      this.logger.debug('writeCameraStreamsSlice failed', {\n        tags: { deviceId },\n        meta: { error: err instanceof Error ? err.message : String(err) },\n      })\n    }\n  }\n}\n","import type { EncodeProfile } from '@camstack/types'\n\n/**\n * Resolve an EncodeProfile from an optional persisted override and a\n * consumer-baked default.\n *\n *   override → consumerDefault (by reference)\n *\n * Pure function. The override-or-not decision is local to the caller.\n * In practice `withSourceResolution` adapts a base `EncodeProfile` to a\n * probed source resolution; the WebRTC path passes\n * `WEBRTC_FALLBACK_PROFILE` as that base. The preset library is\n * intentionally NOT a resolver parameter — the snapshot-semantics rule\n * means a supplied override carries its own materialised profile.\n */\nexport function resolveEncodeProfile(\n  override: EncodeProfile | undefined,\n  consumerDefault: EncodeProfile,\n): EncodeProfile {\n  return override ?? consumerDefault\n}\n\n/**\n * Adapt a consumer's baked default to a publisher-probed source shape.\n *\n * When the probed source advertises both width and height, the returned\n * profile overrides `consumerDefault.video.{width,height}` so the\n * forced-transcode path scales to the source's real resolution instead\n * of the baked 720p — a 4K H.265 camera served to a no-HW-HEVC browser\n * stays 4K H.264 on the wire instead of dropping to 720p.\n *\n * Returns the consumer default by reference when the probed source is\n * absent or carries no dimensions, so callers can rely on identity for\n * \"no smart fallback applied\" checks.\n *\n * Other video fields (codec, profile, fps, bitrate, GOP, preset, tune)\n * are NOT touched — those encode the consumer's protocol contract, not\n * the source. Audio is left untouched for the same reason.\n *\n * Pure function. The decision to apply this fallback (only when the\n * operator hasn't set an explicit override) lives at the call site —\n * see `WebrtcSessionProvider.resolveTranscodeProfile`.\n */\nexport function withSourceResolution(\n  consumerDefault: EncodeProfile,\n  probedSource: EncodeProfile | undefined,\n): EncodeProfile {\n  const probedWidth = probedSource?.video.width\n  const probedHeight = probedSource?.video.height\n  if (probedWidth === undefined || probedHeight === undefined) return consumerDefault\n  return {\n    ...consumerDefault,\n    video: { ...consumerDefault.video, width: probedWidth, height: probedHeight },\n  }\n}\n","import type { EncodeProfile } from '@camstack/types'\n\n/**\n * Hardcoded fallback encode profile for the WebRTC viewer path. Used only\n * when the broker must spawn ffmpeg (source codec ∉ the browser-accepted\n * set, e.g. H.265 source on an H.264-only browser). Replaces the former\n * operator-configurable `streamWebrtcParams` override + `WEBRTC_BAKED_DEFAULT`\n * preset. Bit-identical to the old `web-default` profile.\n */\nexport const WEBRTC_FALLBACK_PROFILE: EncodeProfile = {\n  video: { codec: 'h264', profile: 'baseline', width: 1280, height: 720, fps: 25, bitrateKbps: 3000, gopFrames: 30, bf: 0, preset: 'ultrafast', tune: 'zerolatency' },\n  audio: 'passthrough',\n}\n","/**\n * NAL unit parsing utilities.\n * Protocol-agnostic functions for working with Annex-B NAL units.\n */\n\nimport type { VideoCodec } from \"./types.js\";\n\nconst NAL_START_CODE_4B = Buffer.from([0x00, 0x00, 0x00, 0x01]);\nconst NAL_START_CODE_3B = Buffer.from([0x00, 0x00, 0x01]);\n\n/**\n * Returns true if the buffer starts with an Annex-B start code (0x00000001 or 0x000001).\n */\nexport function hasStartCodes(data: Buffer): boolean {\n  if (data.length < 4) return false;\n  if (data.subarray(0, 4).equals(NAL_START_CODE_4B)) return true;\n  if (data.subarray(0, 3).equals(NAL_START_CODE_3B)) return true;\n  return false;\n}\n\n/**\n * Split Annex-B data into individual NAL unit payloads (without start codes).\n * Works for both H.264 and H.265.\n */\nexport function splitAnnexBToNals(annexB: Buffer): Buffer[] {\n  const nals: Buffer[] = [];\n  const len = annexB.length;\n\n  const isStartCodeAt = (i: number): number => {\n    if (i + 3 <= len && annexB[i] === 0x00 && annexB[i + 1] === 0x00) {\n      if (annexB[i + 2] === 0x01) return 3;\n      if (i + 4 <= len && annexB[i + 2] === 0x00 && annexB[i + 3] === 0x01)\n        return 4;\n    }\n    return 0;\n  };\n\n  let i = 0;\n  // Find first start code\n  while (i < len) {\n    const sc = isStartCodeAt(i);\n    if (sc) break;\n    i++;\n  }\n\n  while (i < len) {\n    const sc = isStartCodeAt(i);\n    if (!sc) {\n      i++;\n      continue;\n    }\n    const nalStart = i + sc;\n    let j = nalStart;\n    while (j < len) {\n      const sc2 = isStartCodeAt(j);\n      if (sc2) break;\n      j++;\n    }\n    if (nalStart < j) {\n      const nal = annexB.subarray(nalStart, j);\n      if (nal.length > 0) nals.push(nal);\n    }\n    i = j;\n  }\n\n  return nals;\n}\n\n/**\n * Prepend a 4-byte Annex-B start code to a NAL payload.\n */\nexport function prependStartCode(nal: Buffer): Buffer {\n  return Buffer.concat([NAL_START_CODE_4B, nal]);\n}\n\n/**\n * Join multiple NAL payloads (without start codes) into an Annex-B access unit.\n */\nexport function joinNalsToAnnexB(\n  ...nals: Array<Buffer | null | undefined>\n): Buffer | undefined {\n  const present = nals.filter((n): n is Buffer => !!n && n.length > 0);\n  if (!present.length) return;\n  const parts: Buffer[] = [];\n  for (const nal of present) {\n    parts.push(NAL_START_CODE_4B, nal);\n  }\n  return Buffer.concat(parts);\n}\n\n/**\n * Detect the actual video codec from raw NAL data.\n * Some cameras report wrong codec (e.g. \"H264\" but send H.265 data).\n * Analyzes the NAL header to determine the real codec.\n *\n * @param data - Raw video data (either Annex-B or length-prefixed)\n * @returns Detected codec type or null if detection fails\n */\nexport function detectVideoCodecFromNal(data: Buffer): VideoCodec | null {\n  if (!data || data.length < 5) return null;\n\n  // Find start code (00 00 00 01 or 00 00 01)\n  let nalStart = -1;\n  for (let i = 0; i < Math.min(data.length - 4, 100); i++) {\n    if (data[i] === 0 && data[i + 1] === 0) {\n      if (data[i + 2] === 0 && data[i + 3] === 1) {\n        nalStart = i + 4;\n        break;\n      }\n      if (data[i + 2] === 1) {\n        nalStart = i + 3;\n        break;\n      }\n    }\n  }\n\n  // If no start code, try length-prefixed (AVCC/HVCC)\n  if (nalStart < 0 && data.length >= 5) {\n    const len = data.readUInt32BE(0);\n    if (len > 0 && len <= data.length - 4) {\n      nalStart = 4;\n    }\n  }\n\n  if (nalStart < 0 || nalStart >= data.length) return null;\n\n  const nalByte = data[nalStart];\n  if (nalByte === undefined) return null;\n\n  // Check H.264 FIRST because H.264 NAL bytes can be misinterpreted as H.265.\n  const forbiddenBit264 = (nalByte >> 7) & 1;\n  const h264Type = nalByte & 0x1f;\n\n  if (forbiddenBit264 === 0 && h264Type > 0 && h264Type <= 12) {\n    if (h264Type === 7 || h264Type === 8) return \"H264\"; // SPS, PPS\n    if (h264Type === 5) return \"H264\"; // IDR\n    if (h264Type === 1) {\n      const nalRefIdc = (nalByte >> 5) & 0x03;\n      if (nalRefIdc >= 1) return \"H264\";\n    }\n  }\n\n  // H.265/HEVC detection\n  if (nalStart + 1 < data.length) {\n    const nalByte2 = data[nalStart + 1];\n    if (nalByte2 !== undefined) {\n      const forbiddenBit = (nalByte >> 7) & 1;\n      const hevcType = (nalByte >> 1) & 0x3f;\n      const temporalId = nalByte2 & 0x07;\n\n      if (forbiddenBit === 0 && temporalId > 0 && hevcType <= 40) {\n        if (hevcType === 32 || hevcType === 33 || hevcType === 34)\n          return \"H265\"; // VPS, SPS, PPS\n        if (hevcType === 19 || hevcType === 20 || hevcType === 21)\n          return \"H265\"; // IDR, CRA\n        if (hevcType <= 1 && nalByte <= 0x03) return \"H265\"; // TRAIL\n      }\n    }\n  }\n\n  return null;\n}\n\nexport { NAL_START_CODE_4B, NAL_START_CODE_3B };\n","/**\n * H.264/AVC utilities.\n * AVCC to Annex-B conversion, SPS/PPS extraction, keyframe detection, RTP depacketization.\n */\n\nimport { NAL_START_CODE_4B, hasStartCodes, splitAnnexBToNals } from \"./nal-utils.js\";\n\n// --- AVCC to Annex-B conversion helpers ---\n\nfunction tryConvertWithLengthReader(\n  data: Buffer,\n  readLen: (buf: Buffer, offset: number) => number,\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  while (offset < data.length) {\n    if (offset + 4 > data.length) return null;\n    const nalLength = readLen(data, offset);\n    offset += 4;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction tryConvertWithLengthReader16(\n  data: Buffer,\n  readLen: (buf: Buffer, offset: number) => number,\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  while (offset < data.length) {\n    if (offset + 2 > data.length) return null;\n    const nalLength = readLen(data, offset);\n    offset += 2;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction tryConvertWithLengthReader24(\n  data: Buffer,\n  endian: \"be\" | \"le\",\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  const readLen24 = (buf: Buffer, at: number): number => {\n    if (at + 3 > buf.length) return 0;\n    const b0 = buf[at]!;\n    const b1 = buf[at + 1]!;\n    const b2 = buf[at + 2]!;\n    return endian === \"be\"\n      ? ((b0 << 16) | (b1 << 8) | b2) >>> 0\n      : ((b2 << 16) | (b1 << 8) | b0) >>> 0;\n  };\n  while (offset < data.length) {\n    if (offset + 3 > data.length) return null;\n    const nalLength = readLen24(data, offset);\n    offset += 3;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction looksLikeSingleH264Nal(nalPayload: Buffer): boolean {\n  if (nalPayload.length < 1) return false;\n  const b0 = nalPayload[0];\n  if (b0 === undefined) return false;\n  if ((b0 & 0x80) !== 0) return false;\n  const nalType = b0 & 0x1f;\n  return nalType >= 1 && nalType <= 23;\n}\n\nfunction depacketizeRtpAggregationToAnnexB(payload: Buffer): Buffer | null {\n  if (payload.length < 1) return null;\n  const nalHeader = payload[0]!;\n  const nalType = nalHeader & 0x1f;\n  const out: Buffer[] = [];\n  const pushNal = (nal: Buffer) => {\n    if (nal.length === 0) return;\n    out.push(NAL_START_CODE_4B, nal);\n  };\n\n  // STAP-A (24)\n  if (nalType === 24) {\n    let off = 1;\n    while (off + 2 <= payload.length) {\n      const size = payload.readUInt16BE(off);\n      off += 2;\n      if (size <= 0 || off + size > payload.length) return null;\n      pushNal(payload.subarray(off, off + size));\n      off += size;\n    }\n    return out.length ? Buffer.concat(out) : null;\n  }\n\n  // STAP-B (25)\n  if (nalType === 25) {\n    let off = 1 + 2;\n    if (off > payload.length) return null;\n    while (off + 2 <= payload.length) {\n      const size = payload.readUInt16BE(off);\n      off += 2;\n      if (size <= 0 || off + size > payload.length) return null;\n      pushNal(payload.subarray(off, off + size));\n      off += size;\n    }\n    return out.length ? Buffer.concat(out) : null;\n  }\n\n  // MTAP16 (26)\n  if (nalType === 26) {\n    let off = 1 + 2;\n    if (off > payload.length) return null;\n    while (off + 2 <= payload.length) {\n      const size = payload.readUInt16BE(off);\n      off += 2;\n      if (off + 1 + 2 > payload.length) return null;\n      off += 1 + 2;\n      if (size <= 0 || off + size > payload.length) return null;\n      pushNal(payload.subarray(off, off + size));\n      off += size;\n    }\n    return out.length ? Buffer.concat(out) : null;\n  }\n\n  // MTAP24 (27)\n  if (nalType === 27) {\n    let off = 1 + 2;\n    if (off > payload.length) return null;\n    while (off + 2 <= payload.length) {\n      const size = payload.readUInt16BE(off);\n      off += 2;\n      if (off + 1 + 3 > payload.length) return null;\n      off += 1 + 3;\n      if (size <= 0 || off + size > payload.length) return null;\n      pushNal(payload.subarray(off, off + size));\n      off += size;\n    }\n    return out.length ? Buffer.concat(out) : null;\n  }\n\n  return null;\n}\n\n// --- Public API ---\n\n/**\n * Convert H.264 data from length-prefixed (AVCC) to Annex-B (start codes).\n * If already Annex-B, returns as-is.\n */\nexport function convertH264ToAnnexB(data: Buffer): Buffer {\n  if (hasStartCodes(data)) return data;\n\n  // Some models prepend a small header before an Annex-B access unit.\n  const sc4 = Buffer.from([0x00, 0x00, 0x00, 0x01]);\n  const sc3 = Buffer.from([0x00, 0x00, 0x01]);\n  const maxScan = Math.min(64, data.length);\n  const idx4 = data.subarray(0, maxScan).indexOf(sc4);\n  if (idx4 > 0) return data.subarray(idx4);\n  const idx3 = data.subarray(0, maxScan).indexOf(sc3);\n  if (idx3 > 0) return data.subarray(idx3);\n\n  // Try AVCC → AnnexB conversion (BE then LE)\n  const be = tryConvertWithLengthReader(data, (b, o) => b.readUInt32BE(o));\n  if (be) return be;\n  const le = tryConvertWithLengthReader(data, (b, o) => b.readUInt32LE(o));\n  if (le) return le;\n\n  // 3-byte lengths\n  const be24 = tryConvertWithLengthReader24(data, \"be\");\n  if (be24) return be24;\n  const le24 = tryConvertWithLengthReader24(data, \"le\");\n  if (le24) return le24;\n\n  // 2-byte lengths\n  const be16 = tryConvertWithLengthReader16(data, (b, o) => b.readUInt16BE(o));\n  if (be16) return be16;\n  const le16 = tryConvertWithLengthReader16(data, (b, o) => b.readUInt16LE(o));\n  if (le16) return le16;\n\n  // RTP aggregation payload\n  const agg = depacketizeRtpAggregationToAnnexB(data);\n  if (agg) return agg;\n\n  // Single NAL without start code\n  if (looksLikeSingleH264Nal(data)) {\n    return Buffer.concat([NAL_START_CODE_4B, data]);\n  }\n\n  return data;\n}\n\n/**\n * Check if an H.264 Annex-B access unit is a keyframe (IDR + SPS + PPS).\n */\nexport function isH264KeyframeAnnexB(annexB: Buffer): boolean {\n  const nals = splitAnnexBToNals(annexB);\n  let hasSps = false;\n  let hasPps = false;\n  let hasIdr = false;\n  for (const nal of nals) {\n    const t = (nal[0] ?? 0) & 0x1f;\n    if (t === 7) hasSps = true;\n    if (t === 8) hasPps = true;\n    if (t === 5) hasIdr = true;\n  }\n  return hasIdr && hasSps && hasPps;\n}\n\n/**\n * Check if an H.264 Annex-B access unit contains an IDR slice.\n */\n/**\n * Detect H.264 keyframe (IDR or I-frame with SPS).\n *\n * Some cameras emit non-IDR I-frames (NAL type 1 with slice_type=I)\n * instead of IDR frames (NAL type 5). Both are intra-coded and valid\n * as decoder entry points, but only IDR carries NAL type 5.\n *\n * We detect keyframes by:\n *   1. NAL type 5 (IDR) — canonical\n *   2. SPS (type 7) present — parameter sets are always sent with keyframes\n */\nexport function isH264IdrAccessUnit(annexB: Buffer): boolean {\n  const nals = splitAnnexBToNals(annexB);\n  let hasSps = false;\n  for (const nal of nals) {\n    if (nal.length < 1) continue;\n    const t = (nal[0] ?? 0) & 0x1f;\n    if (t === 5) return true;  // IDR slice\n    if (t === 7) hasSps = true; // SPS — indicates keyframe boundary\n  }\n  return hasSps;\n}\n\n/**\n * Extract SPS and PPS from an H.264 Annex-B access unit.\n */\nexport function extractH264ParamSets(annexB: Buffer): {\n  sps?: Buffer;\n  pps?: Buffer;\n  profileLevelId?: string;\n} {\n  const nals = splitAnnexBToNals(annexB);\n  let sps: Buffer | undefined;\n  let pps: Buffer | undefined;\n  let profileLevelId: string | undefined;\n\n  for (const nal of nals) {\n    if (nal.length < 1) continue;\n    const nalType = nal[0]! & 0x1f;\n    if (nalType === 7) {\n      sps = nal;\n      if (nal.length >= 4) {\n        profileLevelId = Buffer.from([nal[1]!, nal[2]!, nal[3]!]).toString(\n          \"hex\",\n        );\n      }\n    } else if (nalType === 8) {\n      pps = nal;\n    }\n  }\n\n  const out: { sps?: Buffer; pps?: Buffer; profileLevelId?: string } = {};\n  if (sps) out.sps = sps;\n  if (pps) out.pps = pps;\n  if (profileLevelId) out.profileLevelId = profileLevelId;\n  return out;\n}\n\n/**\n * True when an H.264 `profile-level-id` (6 hex chars: profile_idc +\n * constraint-set flags + level_idc) is Baseline / Constrained Baseline,\n * i.e. profile_idc 0x42 (66). iOS Safari/WebKit WebRTC decodes ONLY this\n * profile for H.264 RECEIVE — Main (0x4d) and High (0x64) negotiate but\n * never decode (endless PLI storm, black view). The test is on\n * profile_idc (the first byte) alone: the constraint flags and level\n * don't change Baseline decodability on iOS.\n */\nexport function isBaselineProfileLevelId(profileLevelId: string): boolean {\n  return profileLevelId.length >= 2 && profileLevelId.slice(0, 2).toLowerCase() === \"42\";\n}\n\n/**\n * Group a flat list of H.264 NAL units into access units (one decoded\n * picture each), per the boundary rules of H.264 §7.4.1.2.4. A new access\n * unit starts at:\n *   - an Access Unit Delimiter (type 9), or\n *   - a VCL slice (types 1–5) whose `first_mb_in_slice == 0` (a new\n *     primary coded picture — detected by the high bit of the byte after\n *     the NAL header), or\n *   - an SPS (7) / PPS (8) / SEI (6) that follows a completed picture.\n * AUD NALs are dropped from the output (they're delimiters only).\n *\n * This restores correct frame delimiting for the transcode feed: each\n * returned access unit is sent with exactly ONE RTP timestamp and a\n * single trailing marker bit, so the receiver can assemble and decode\n * each picture. (ffmpeg's raw-H.264 stdout is a bytestream with no frame\n * boundaries, so we recover them here.)\n */\nexport function groupNalsIntoAccessUnits(nals: readonly Buffer[]): Buffer[][] {\n  const accessUnits: Buffer[][] = [];\n  let current: Buffer[] = [];\n  let currentHasVcl = false;\n\n  const isVcl = (type: number): boolean => type >= 1 && type <= 5;\n  const startsNewAccessUnit = (type: number, nal: Buffer): boolean => {\n    // A VCL NAL begins a new picture when first_mb_in_slice == 0, which\n    // encodes as a leading '1' bit → high bit of the first slice-header byte.\n    if (isVcl(type)) return nal.length >= 2 && (nal[1]! & 0x80) === 0x80;\n    // AUD / SPS / PPS / SEI before the next picture belong to that picture.\n    return type === 9 || type === 7 || type === 8 || type === 6;\n  };\n\n  for (const nal of nals) {\n    if (nal.length < 1) continue;\n    const type = nal[0]! & 0x1f;\n    if (currentHasVcl && startsNewAccessUnit(type, nal)) {\n      accessUnits.push(current);\n      current = [];\n      currentHasVcl = false;\n    }\n    if (type === 9) continue; // drop the Access Unit Delimiter\n    current.push(nal);\n    if (isVcl(type)) currentHasVcl = true;\n  }\n  if (current.length > 0) accessUnits.push(current);\n  return accessUnits;\n}\n\n/**\n * H.264 RTP depacketizer (RFC 6184).\n * Handles single NAL units, STAP-A aggregation, and FU-A/FU-B fragmentation.\n * Returns complete NAL units in Annex-B format (with start codes).\n */\nexport class H264RtpDepacketizer {\n  private fuNalHeader: number | null = null;\n  private fuParts: Buffer[] = [];\n\n  private static parseRtpPayload(packet: Buffer): Buffer | null {\n    if (!packet || packet.length < 12) return null;\n    const version = (packet[0]! >> 6) & 0x03;\n    if (version !== 2) return null;\n    const padding = (packet[0]! & 0x20) !== 0;\n    const extension = (packet[0]! & 0x10) !== 0;\n    const csrcCount = packet[0]! & 0x0f;\n    let offset = 12 + csrcCount * 4;\n    if (offset > packet.length) return null;\n    if (extension) {\n      if (offset + 4 > packet.length) return null;\n      const extLenWords = packet.readUInt16BE(offset + 2);\n      offset += 4 + extLenWords * 4;\n      if (offset > packet.length) return null;\n    }\n    let end = packet.length;\n    if (padding) {\n      const padLen = packet[packet.length - 1]!;\n      if (padLen <= 0 || padLen > packet.length) return null;\n      end = packet.length - padLen;\n      if (end < offset) return null;\n    }\n    if (end <= offset) return null;\n    return packet.subarray(offset, end);\n  }\n\n  reset(): void {\n    this.fuNalHeader = null;\n    this.fuParts = [];\n  }\n\n  push(payload: Buffer): Buffer[] {\n    if (payload.length === 0) return [];\n\n    const rtpPayload = H264RtpDepacketizer.parseRtpPayload(payload);\n    if (rtpPayload) payload = rtpPayload;\n\n    if (hasStartCodes(payload)) return [payload];\n\n    const b0 = payload[0]!;\n    if ((b0 & 0x80) !== 0) return [];\n    const nalType = b0 & 0x1f;\n\n    // Single NAL unit\n    if (nalType >= 1 && nalType <= 23) {\n      return [Buffer.concat([NAL_START_CODE_4B, payload])];\n    }\n\n    // STAP-A (24)\n    if (nalType === 24) {\n      if (payload.length < 1 + 2) return [];\n      let off = 1;\n      const out: Buffer[] = [];\n      while (off + 2 <= payload.length) {\n        const size = payload.readUInt16BE(off);\n        off += 2;\n        if (size <= 0 || off + size > payload.length) return [];\n        const nal = payload.subarray(off, off + size);\n        off += size;\n        if (nal.length < 1) return [];\n        if ((nal[0]! & 0x80) !== 0) return [];\n        const t = nal[0]! & 0x1f;\n        if (t === 0 || t >= 24) return [];\n        out.push(Buffer.concat([NAL_START_CODE_4B, nal]));\n      }\n      return out;\n    }\n\n    // FU-A (28) / FU-B (29)\n    if (nalType === 28 || nalType === 29) {\n      if (payload.length < 2) return [];\n      const fuIndicator = payload[0]!;\n      const fuHeader = payload[1]!;\n      const start = (fuHeader & 0x80) !== 0;\n      const end = (fuHeader & 0x40) !== 0;\n      const origType = fuHeader & 0x1f;\n      const reconstructedHeader = (fuIndicator & 0xe0) | origType;\n\n      let off = 2;\n      if (nalType === 29) {\n        if (payload.length < off + 2) return [];\n        off += 2;\n      }\n      const frag = payload.subarray(off);\n\n      if (start) {\n        this.fuNalHeader = reconstructedHeader;\n        this.fuParts = [frag];\n      } else if (this.fuNalHeader != null) {\n        this.fuParts.push(frag);\n      } else {\n        return [];\n      }\n\n      if (end && this.fuNalHeader != null) {\n        const nal = Buffer.concat([\n          Buffer.from([this.fuNalHeader]),\n          ...this.fuParts,\n        ]);\n        this.reset();\n        return [Buffer.concat([NAL_START_CODE_4B, nal])];\n      }\n      return [];\n    }\n\n    return [];\n  }\n}\n","/**\n * H.265/HEVC utilities.\n * HVCC to Annex-B conversion, VPS/SPS/PPS extraction, keyframe detection, RTP depacketization.\n */\n\nimport { NAL_START_CODE_4B, hasStartCodes, splitAnnexBToNals } from \"./nal-utils.js\";\n\n// --- HVCC to Annex-B conversion helpers ---\n\nfunction tryConvertWithLengthReader(\n  data: Buffer,\n  readLen: (buf: Buffer, offset: number) => number,\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  while (offset < data.length) {\n    if (offset + 4 > data.length) return null;\n    const nalLength = readLen(data, offset);\n    offset += 4;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction tryConvertWithLengthReader16(\n  data: Buffer,\n  readLen: (buf: Buffer, offset: number) => number,\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  while (offset < data.length) {\n    if (offset + 2 > data.length) return null;\n    const nalLength = readLen(data, offset);\n    offset += 2;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction tryConvertWithLengthReader24(\n  data: Buffer,\n  endian: \"be\" | \"le\",\n): Buffer | null {\n  const result: Buffer[] = [];\n  let offset = 0;\n  let nalCount = 0;\n  const readLen24 = (buf: Buffer, at: number): number => {\n    if (at + 3 > buf.length) return 0;\n    const b0 = buf[at]!;\n    const b1 = buf[at + 1]!;\n    const b2 = buf[at + 2]!;\n    return endian === \"be\"\n      ? ((b0 << 16) | (b1 << 8) | b2) >>> 0\n      : ((b2 << 16) | (b1 << 8) | b0) >>> 0;\n  };\n  while (offset < data.length) {\n    if (offset + 3 > data.length) return null;\n    const nalLength = readLen24(data, offset);\n    offset += 3;\n    if (nalLength <= 0) return null;\n    if (nalLength > data.length - offset) return null;\n    result.push(NAL_START_CODE_4B);\n    result.push(data.subarray(offset, offset + nalLength));\n    offset += nalLength;\n    nalCount++;\n  }\n  if (nalCount === 0) return null;\n  return Buffer.concat(result);\n}\n\nfunction looksLikeSingleH265Nal(nalPayload: Buffer): boolean {\n  if (nalPayload.length < 2) return false;\n  const b0 = nalPayload[0];\n  if (b0 === undefined) return false;\n  if ((b0 & 0x80) !== 0) return false;\n  const nalType = (b0 >> 1) & 0x3f;\n  return nalType <= 40;\n}\n\n// --- Public API ---\n\n/**\n * Convert H.265 data from length-prefixed (HVCC) to Annex-B (start codes).\n * If already Annex-B, returns as-is.\n */\nexport function convertH265ToAnnexB(data: Buffer): Buffer {\n  if (hasStartCodes(data)) return data;\n\n  const sc4 = Buffer.from([0x00, 0x00, 0x00, 0x01]);\n  const sc3 = Buffer.from([0x00, 0x00, 0x01]);\n  const maxScan = Math.min(64, data.length);\n  const idx4 = data.subarray(0, maxScan).indexOf(sc4);\n  if (idx4 > 0) return data.subarray(idx4);\n  const idx3 = data.subarray(0, maxScan).indexOf(sc3);\n  if (idx3 > 0) return data.subarray(idx3);\n\n  // Try HVCC → AnnexB conversion\n  const be = tryConvertWithLengthReader(data, (b, o) => b.readUInt32BE(o));\n  if (be) return be;\n  const le = tryConvertWithLengthReader(data, (b, o) => b.readUInt32LE(o));\n  if (le) return le;\n\n  const be24 = tryConvertWithLengthReader24(data, \"be\");\n  if (be24) return be24;\n  const le24 = tryConvertWithLengthReader24(data, \"le\");\n  if (le24) return le24;\n\n  const be16 = tryConvertWithLengthReader16(data, (b, o) => b.readUInt16BE(o));\n  if (be16) return be16;\n  const le16 = tryConvertWithLengthReader16(data, (b, o) => b.readUInt16LE(o));\n  if (le16) return le16;\n\n  if (looksLikeSingleH265Nal(data)) {\n    return Buffer.concat([NAL_START_CODE_4B, data]);\n  }\n  return data;\n}\n\n/**\n * Get H.265 NAL unit type from a NAL payload (without start code).\n */\nexport function getH265NalType(nalPayload: Buffer): number | null {\n  if (nalPayload.length < 1) return null;\n  const b0 = nalPayload[0];\n  if (b0 === undefined) return null;\n  if ((b0 & 0x80) !== 0) return null;\n  return (b0 >> 1) & 0x3f;\n}\n\n/**\n * Check if an H.265 NAL unit type is an IRAP (Intra Random Access Point) picture.\n * IRAP types: BLA (16-18), IDR (19-20), CRA (21).\n */\nexport function isH265Irap(nalType: number): boolean {\n  return nalType >= 16 && nalType <= 23;\n}\n\n/**\n * Check if an H.265 Annex-B access unit is a keyframe (IRAP + VPS + SPS + PPS).\n */\nexport function isH265KeyframeAnnexB(annexB: Buffer): boolean {\n  const nals = splitAnnexBToNals(annexB);\n  let hasVps = false;\n  let hasSps = false;\n  let hasPps = false;\n  let hasIrap = false;\n\n  for (const nal of nals) {\n    const nalType = getH265NalType(nal);\n    if (nalType === null) continue;\n    if (nalType === 32) hasVps = true;\n    if (nalType === 33) hasSps = true;\n    if (nalType === 34) hasPps = true;\n    if (isH265Irap(nalType)) hasIrap = true;\n  }\n\n  return hasIrap && hasVps && hasSps && hasPps;\n}\n\n/**\n * Check if an H.265 Annex-B access unit contains an IRAP picture.\n */\nexport function isH265IrapAccessUnit(annexB: Buffer): boolean {\n  const nals = splitAnnexBToNals(annexB);\n  for (const nal of nals) {\n    if (nal.length < 2) continue;\n    const b0 = nal[0];\n    if (b0 === undefined) continue;\n    if ((b0 & 0x80) !== 0) continue;\n    const nalType = (b0 >> 1) & 0x3f;\n    if (isH265Irap(nalType)) return true;\n  }\n  return false;\n}\n\n/**\n * Extract VPS, SPS, and PPS from an H.265 Annex-B access unit.\n */\nexport function extractH265ParamSets(annexB: Buffer): {\n  vps?: Buffer;\n  sps?: Buffer;\n  pps?: Buffer;\n} {\n  const nals = splitAnnexBToNals(annexB);\n  let vps: Buffer | undefined;\n  let sps: Buffer | undefined;\n  let pps: Buffer | undefined;\n\n  for (const nal of nals) {\n    if (nal.length < 2) continue;\n    const nalType = (nal[0]! >> 1) & 0x3f;\n    if (nalType === 32) vps = nal;\n    else if (nalType === 33) sps = nal;\n    else if (nalType === 34) pps = nal;\n  }\n\n  const out: { vps?: Buffer; sps?: Buffer; pps?: Buffer } = {};\n  if (vps) out.vps = vps;\n  if (sps) out.sps = sps;\n  if (pps) out.pps = pps;\n  return out;\n}\n\n/**\n * H.265 RTP depacketizer (RFC 7798).\n * Handles single NAL units, AP aggregation, and FU fragmentation.\n * Returns complete NAL units in Annex-B format (with start codes).\n */\nexport class H265RtpDepacketizer {\n  private fuParts: Buffer[] | null = null;\n\n  private static parseRtpPayload(packet: Buffer): Buffer | null {\n    if (!packet || packet.length < 12) return null;\n    const version = (packet[0]! >> 6) & 0x03;\n    if (version !== 2) return null;\n    const padding = (packet[0]! & 0x20) !== 0;\n    const extension = (packet[0]! & 0x10) !== 0;\n    const csrcCount = packet[0]! & 0x0f;\n    let offset = 12 + csrcCount * 4;\n    if (offset > packet.length) return null;\n    if (extension) {\n      if (offset + 4 > packet.length) return null;\n      const extLenWords = packet.readUInt16BE(offset + 2);\n      offset += 4 + extLenWords * 4;\n      if (offset > packet.length) return null;\n    }\n    let end = packet.length;\n    if (padding) {\n      const padLen = packet[packet.length - 1]!;\n      if (padLen <= 0 || padLen > packet.length) return null;\n      end = packet.length - padLen;\n      if (end < offset) return null;\n    }\n    if (end <= offset) return null;\n    return packet.subarray(offset, end);\n  }\n\n  reset(): void {\n    this.fuParts = null;\n  }\n\n  push(payload: Buffer): Buffer[] {\n    if (!payload || payload.length < 2) return [];\n\n    const rtpPayload = H265RtpDepacketizer.parseRtpPayload(payload);\n    if (rtpPayload) payload = rtpPayload;\n\n    const h0 = payload[0]!;\n    const h1 = payload[1]!;\n    if ((h0 & 0x80) !== 0) return [];\n    const nalType = (h0 >> 1) & 0x3f;\n\n    // AP (48): aggregation packet\n    if (nalType === 48) {\n      let off = 2;\n      const out: Buffer[] = [];\n      while (off + 2 <= payload.length) {\n        const size = payload.readUInt16BE(off);\n        off += 2;\n        if (size <= 0 || off + size > payload.length) return [];\n        const nal = payload.subarray(off, off + size);\n        off += size;\n        if (nal.length) out.push(NAL_START_CODE_4B, nal);\n      }\n      return out.length ? [Buffer.concat(out)] : [];\n    }\n\n    // FU (49): fragmentation unit\n    if (nalType === 49) {\n      if (payload.length < 3) return [];\n      const fuHeader = payload[2]!;\n      const start = (fuHeader & 0x80) !== 0;\n      const end = (fuHeader & 0x40) !== 0;\n      const origType = fuHeader & 0x3f;\n      const orig0 = (h0 & 0x81) | ((origType & 0x3f) << 1);\n      const orig1 = h1;\n      const frag = payload.subarray(3);\n\n      if (start) {\n        this.fuParts = [NAL_START_CODE_4B, Buffer.from([orig0, orig1]), frag];\n      } else {\n        if (!this.fuParts) return [];\n        this.fuParts.push(frag);\n      }\n\n      if (end) {\n        if (!this.fuParts) return [];\n        const out = Buffer.concat(this.fuParts);\n        this.fuParts = null;\n        return [out];\n      }\n\n      return [];\n    }\n\n    // Single NAL unit\n    return [Buffer.concat([NAL_START_CODE_4B, payload])];\n  }\n}\n","import { lookup as dnsLookup } from 'node:dns/promises'\nimport { errMsg } from '@camstack/types'\nimport type { Logger } from './types.js'\n\n/**\n * Rewrite mDNS (`*.local`) candidates in an SDP by resolving each\n * hostname to an IP via the OS resolver.\n *\n * Chrome emits anonymised `<uuid>.local` host candidates by default\n * for privacy; werift's ICE agent has no mDNS resolver, so those\n * candidates never pair and the session sits at \"checking\" forever.\n *\n * On macOS the OS resolver delegates `.local` to Bonjour (built in); on\n * Linux it usually requires Avahi's NSS module. We parallelise the\n * lookups and silently drop candidates that don't resolve — a failed\n * lookup is the same as \"werift didn't use it\", so we're not making\n * the worst case worse. The original line is replaced in place, keeping\n * the rest of the candidate fields (priority, ufrag, type, …) intact.\n *\n * Used by both `AdaptiveSession` and `SharedSession` — pulled out into\n * its own module so the two paths stay identical.\n */\nexport async function resolveMdnsCandidatesInSdp(\n  sdp: string,\n  logger: Logger,\n  sessionTag: string,\n): Promise<string> {\n  const mdnsHostRe = /([0-9a-fA-F-]+\\.local)/g\n  const hosts = new Set<string>()\n  for (const match of sdp.matchAll(mdnsHostRe)) hosts.add(match[1]!)\n  if (hosts.size === 0) return sdp\n\n  // Cap each lookup hard. A `.local` from a REMOTE peer (cellular/Tailscale)\n  // is unresolvable here, and the OS resolver (getaddrinfo) blocks ~5s before\n  // giving up — that delay stalls `setRemoteDescription` and breaks the whole\n  // ICE flow even when the answer ALSO carries usable srflx/relay candidates\n  // (the remote case). A same-LAN peer's `.local` resolves via Bonjour/Avahi\n  // in a few ms, so a short cap keeps the LAN benefit while never blocking ICE\n  // on the (pointless-for-remote) host candidates.\n  const MDNS_LOOKUP_TIMEOUT_MS = 400\n  const replacements = await Promise.all(\n    [...hosts].map(async (host) => {\n      try {\n        const address = await Promise.race([\n          dnsLookup(host, { family: 4 }).then((r) => r.address),\n          new Promise<never>((_, reject) =>\n            setTimeout(() => reject(new Error(`mDNS lookup timed out after ${MDNS_LOOKUP_TIMEOUT_MS}ms`)), MDNS_LOOKUP_TIMEOUT_MS).unref?.(),\n          ),\n        ])\n        logger.info('mDNS resolve succeeded', { meta: { sessionTag, host, address } })\n        return [host, address] as const\n      } catch (err) {\n        logger.warn('mDNS resolve failed (dropping host candidate, srflx/relay still used)', { meta: { sessionTag, host, error: errMsg(err) } })\n        return [host, null] as const\n      }\n    }),\n  )\n\n  let rewritten = sdp\n  for (const [host, address] of replacements) {\n    if (!address) continue\n    const escaped = host.replace(/\\./g, '\\\\.')\n    rewritten = rewritten.replace(new RegExp(escaped, 'g'), address)\n  }\n  return rewritten\n}\n","/**\n * Used by H265Repacketizer to reorder out-of-order RTP packets within a\n * small jitter window before re-fragmenting. Reordering matters for FU\n * fragments arriving across an unreliable transport — over UDP, a late\n * FU_middle could otherwise break the running re-fragmentation state.\n *\n * No semantic changes — only types narrowed for our codebase.\n */\n\n/**\n * This is a subset of Werift's RtpPacket.\n */\nexport interface RtpPacket {\n    payload: Buffer;\n    header: {\n        padding: boolean;\n        marker: boolean;\n        timestamp: number;\n        sequenceNumber: number;\n    };\n    clone(): RtpPacket;\n    serialize(): Buffer;\n}\n\nexport function sequenceNumberDistance(s1: number, s2: number): number {\n    if (s2 === s1)\n        return 0;\n    const distance = s2 - s1;\n    let rolloverDistance: number;\n    if (s2 > s1)\n        rolloverDistance = s1 + 0x10000 - s2;\n    else\n        rolloverDistance = s2 + 0x10000 - s1;\n\n    if (Math.abs(distance) < Math.abs(rolloverDistance))\n        return distance;\n    return rolloverDistance;\n}\n\nexport function nextSequenceNumber(current: number, increment = 1) {\n    return (current + increment + 0x10000) % 0x10000;\n}\n\nconst maxRtpTimestamp = BigInt(0xFFFFFFFF);\nexport function addRtpTimestamp(current: number, adjust: number) {\n    return Number(maxRtpTimestamp & (BigInt(current) + BigInt(adjust)));\n}\n\nexport function isNextSequenceNumber(current: number, next: number) {\n    return nextSequenceNumber(current) === next;\n}\n\nexport class JitterBuffer {\n    lastSequenceNumber: number | undefined;\n    pending: (RtpPacket | undefined)[] = [];\n\n    constructor(public console: Console, public jitterSize: number,) {\n    }\n\n    flushPending(afterSequenceNumber: number, ret: RtpPacket[]): RtpPacket[] {\n        if (!this.pending)\n            return ret;\n\n        const start = nextSequenceNumber(afterSequenceNumber);\n\n        for (let i = 0; i < this.jitterSize; i++) {\n            const index = (start + i) % this.jitterSize;\n            const packet = this.pending[index];\n            if (!packet)\n                continue;\n            const { sequenceNumber } = packet.header;\n            const sd = sequenceNumberDistance(this.lastSequenceNumber ?? sequenceNumber, sequenceNumber);\n            // packet needs to be purged from the the buffer for being too old.\n            if (sd <= 0) {\n                this.console.log('jitter buffer purged packet:', sequenceNumber);\n                this.pending[index] = undefined;\n                ret.push(packet);\n            }\n            else if (sd === 1) {\n                this.pending[index] = undefined;\n                this.lastSequenceNumber = sequenceNumber;\n                ret.push(packet);\n            }\n            else {\n                // can't do anything with this packet yet.\n            }\n        }\n        return ret;\n    }\n\n    queue(packet: RtpPacket): RtpPacket[] {\n        if (this.lastSequenceNumber === undefined || isNextSequenceNumber(this.lastSequenceNumber, packet.header.sequenceNumber)) {\n            this.lastSequenceNumber = packet.header.sequenceNumber;\n            return this.flushPending(this.lastSequenceNumber, [packet]);\n        }\n\n        const { sequenceNumber } = packet.header;\n        const packetDistance = sequenceNumberDistance(this.lastSequenceNumber, sequenceNumber);\n        // late/duplicate packet\n        if (packetDistance <= 0)\n            return [];\n\n        const ret: RtpPacket[] = [];\n\n        // missed/late bunch of packets\n        if (packetDistance > this.jitterSize) {\n            // this.console.log('jitter buffer skipped packets:', packetDistance);\n            const { lastSequenceNumber } = this;\n            this.lastSequenceNumber = sequenceNumber - this.jitterSize;\n            // use the previous sequence number to flush any packets that are too old compared\n            // to the new sequence number.\n            this.flushPending(lastSequenceNumber, ret);\n        }\n\n        this.pending[packet.header.sequenceNumber % this.jitterSize] = packet;\n        return this.flushPending(this.lastSequenceNumber, ret);\n    }\n}\n","/**\n * Re-fragments RTP packets coming straight off the camera so they fit\n * the WebRTC peer's MTU (~1200 bytes) WITHOUT re-encoding or rebuilding\n * the RTP header from scratch. Source SSRC/seq/timestamp are preserved\n * (only adjusted by werift's `replaceRTP` offset machinery on send).\n *\n * Forwarding source RTP through this repacketizer fixes the\n * `framesAssembledFromMultiplePackets = 0` issue our prior\n * \"depacketize → AnnexB → re-packetize from scratch\" path produced\n * for H.265 over WebRTC. Original RTP header layout is what Chrome's\n * HEVC depacketizer expects.\n */\nimport { isNextSequenceNumber, JitterBuffer, RtpPacket } from \"./jitter-buffer.js\";\n\n// H.265 NAL unit types\n// https://github.com/FFmpeg/FFmpeg/blob/master/libavcodec/hevc/hevc.h\nconst NAL_TYPE_TRAIL_N = 0;\nconst _UNUSED_NAL_TYPE_TRAIL_R = 1;\nconst _UNUSED_NAL_TYPE_TSA_N = 2;\nconst _UNUSED_NAL_TYPE_TSA_R = 3;\nconst _UNUSED_NAL_TYPE_STSA_N = 4;\nconst _UNUSED_NAL_TYPE_STSA_R = 5;\nconst _UNUSED_NAL_TYPE_RADL_N = 6;\nconst _UNUSED_NAL_TYPE_RADL_R = 7;\nconst _UNUSED_NAL_TYPE_RASL_N = 8;\nconst _UNUSED_NAL_TYPE_RASL_R = 9;\nconst _UNUSED_NAL_TYPE_VCL_N10 = 10;\nconst _UNUSED_NAL_TYPE_VCL_R11 = 11;\nconst _UNUSED_NAL_TYPE_VCL_N12 = 12;\nconst _UNUSED_NAL_TYPE_VCL_R13 = 13;\nconst _UNUSED_NAL_TYPE_VCL_N14 = 14;\nconst _UNUSED_NAL_TYPE_VCL_R15 = 15;\nconst NAL_TYPE_BLA_W_LP = 16;\nconst NAL_TYPE_BLA_W_RADL = 17;\nconst NAL_TYPE_BLA_N_LP = 18;\nconst NAL_TYPE_IDR_W_RADL = 19;\nconst NAL_TYPE_IDR_N_LP = 20;\nconst NAL_TYPE_CRA_NUT = 21;\nconst _UNUSED_NAL_TYPE_RSV_IRAP_VCL22 = 22;\nconst NAL_TYPE_RSV_IRAP_VCL23 = 23;\nconst _UNUSED_NAL_TYPE_RSV_VCL24 = 24;\nconst _UNUSED_NAL_TYPE_RSV_VCL25 = 25;\nconst _UNUSED_NAL_TYPE_RSV_VCL26 = 26;\nconst _UNUSED_NAL_TYPE_RSV_VCL27 = 27;\nconst _UNUSED_NAL_TYPE_RSV_VCL28 = 28;\nconst _UNUSED_NAL_TYPE_RSV_VCL29 = 29;\nconst _UNUSED_NAL_TYPE_RSV_VCL30 = 30;\nconst _UNUSED_NAL_TYPE_RSV_VCL31 = 31;\nconst NAL_TYPE_VPS = 32;\nconst NAL_TYPE_SPS = 33;\nconst NAL_TYPE_PPS = 34;\nconst NAL_TYPE_AUD = 35;\nconst _UNUSED_NAL_TYPE_EOS_NUT = 36;\nconst _UNUSED_NAL_TYPE_EOB_NUT = 37;\nconst _UNUSED_NAL_TYPE_FD_NUT = 38;\nconst NAL_TYPE_SEI_PREFIX = 39;\nconst NAL_TYPE_SEI_SUFFIX = 40;\nconst _UNUSED_NAL_TYPE_RSV_NVCL41 = 41;\nconst _UNUSED_NAL_TYPE_RSV_NVCL42 = 42;\nconst _UNUSED_NAL_TYPE_RSV_NVCL43 = 43;\nconst _UNUSED_NAL_TYPE_RSV_NVCL44 = 44;\nconst _UNUSED_NAL_TYPE_RSV_NVCL45 = 45;\nconst _UNUSED_NAL_TYPE_RSV_NVCL46 = 46;\nconst _UNUSED_NAL_TYPE_RSV_NVCL47 = 47;\n// RTP payload format for H.265 defines these special types\nconst NAL_TYPE_AP = 48;  // Aggregation Packet\nconst NAL_TYPE_FU = 49;  // Fragmentation Unit\nconst _UNUSED_NAL_TYPE_UNSPEC50 = 50;\nconst _UNUSED_NAL_TYPE_UNSPEC51 = 51;\nconst _UNUSED_NAL_TYPE_UNSPEC52 = 52;\nconst _UNUSED_NAL_TYPE_UNSPEC53 = 53;\nconst _UNUSED_NAL_TYPE_UNSPEC54 = 54;\nconst _UNUSED_NAL_TYPE_UNSPEC55 = 55;\nconst _UNUSED_NAL_TYPE_UNSPEC56 = 56;\nconst _UNUSED_NAL_TYPE_UNSPEC57 = 57;\nconst _UNUSED_NAL_TYPE_UNSPEC58 = 58;\nconst _UNUSED_NAL_TYPE_UNSPEC59 = 59;\nconst _UNUSED_NAL_TYPE_UNSPEC60 = 60;\nconst _UNUSED_NAL_TYPE_UNSPEC61 = 61;\nconst _UNUSED_NAL_TYPE_UNSPEC62 = 62;\nconst _UNUSED_NAL_TYPE_UNSPEC63 = 63;\n\n\nconst NAL_HEADER_SIZE = 2;  // H265 has 2-byte NAL header\nconst FU_HEADER_SIZE = 3;   // 2-byte NAL header + 1-byte FU header\nconst LENGTH_FIELD_SIZE = 2;\nconst AP_HEADER_SIZE = NAL_HEADER_SIZE + LENGTH_FIELD_SIZE;\n\n// Function to extract NAL unit type from H.265 NAL header\nfunction getNalType(data: Buffer): number {\n    return (data[0]! & 0x7E) >> 1;  // 6 bits starting from bit 1\n}\n\nfunction isKeyFrame(nalType: number): boolean {\n    // For IDR frames, send codec info first\n    if (nalType === NAL_TYPE_IDR_W_RADL || nalType === NAL_TYPE_IDR_N_LP ||\n        nalType === NAL_TYPE_BLA_W_LP || nalType === NAL_TYPE_BLA_W_RADL ||\n        nalType === NAL_TYPE_BLA_N_LP || nalType === NAL_TYPE_CRA_NUT) {\n     return true;\n    }\n    return false;\n}\n\n// Function to depacketize Aggregation Packets (similar to STAP-A in H.264)\nexport function depacketizeAP(data: Buffer): Buffer[] {\n    const ret: Buffer[] = [];\n    let lastPos: number | undefined;\n    let pos = NAL_HEADER_SIZE;\n    while (pos < data.length) {\n        if (lastPos !== undefined)\n            ret.push(data.subarray(lastPos, pos));\n        const naluSize = data.readUInt16BE(pos);\n        pos += LENGTH_FIELD_SIZE;\n        lastPos = pos;\n        pos += naluSize;\n    }\n    if (lastPos !== undefined) ret.push(data.subarray(lastPos));\n    return ret;\n}\n\nexport function splitH265NaluStartCode(data: Buffer) {\n    const ret: Buffer[] = [];\n    let previous = 0;\n    let offset = 0;\n    const maybeAddSlice = () => {\n        const slice = data.subarray(previous, offset);\n        if (slice.length)\n            ret.push(slice);\n        offset += 4;\n        previous = offset;\n    }\n\n    while (offset < data.length - 4) {\n        const startCode = data.readUInt32BE(offset);\n        if (startCode === 1) {\n            maybeAddSlice();\n        }\n        else {\n            offset++;\n        }\n    }\n    offset = data.length;\n    maybeAddSlice();\n\n    return ret;\n}\n\nexport interface H265CodecInfo {\n    vps?: Buffer;\n    sps?: Buffer;\n    pps?: Buffer;\n    seiPrefix?: Buffer;\n    seiSuffix?: Buffer;\n}\n\nexport class H265Repacketizer {\n    extraPackets = 0;\n    fuMax!: number;\n    pendingFU: RtpPacket[] | undefined;\n    // the AP packet that will be sent before an IDR frame.\n    ap: RtpPacket | undefined;\n    fuMin!: number;\n    // h265 can send multiple IDR and TRAIL_R frames per RTP timestamp.\n    // the repurposed h264-packetizer code sends codec information before every\n    // IDR frame.\n    // the IDR burst frames are interleaved with codec information which causes decode failure.\n    // The interleaved codec issues does not cause issues with Safari, but does cause\n    // Chrome to send repeated Picture Loss Indication (PLI) requests.\n    // the fix is to send the codec information before the first IDR frame,\n    // then wait for the marker bit, which denotes the last packet for a given rtp\n    // timestamp, to reset the flag.\n    // expected result:\n    // AP codec info -> IDR frame 1 -> IDR frame 2 -> ... -> IDR frame N with marker bit set\n    sentMarker = true;\n\n    constructor(public console: Console, private maxPacketSize: number, public codecInfo?: H265CodecInfo, public jitterBuffer = new JitterBuffer(console, 4)) {\n        this.setMaxPacketSize(maxPacketSize);\n    }\n\n    setMaxPacketSize(maxPacketSize: number) {\n        this.maxPacketSize = maxPacketSize;\n        // 12 is the rtp/srtp header size.\n        this.fuMax = maxPacketSize - FU_HEADER_SIZE;\n        this.fuMin = Math.round(maxPacketSize * .8);\n    }\n\n    ensureCodecInfo(): H265CodecInfo {\n        if (!this.codecInfo) {\n            this.codecInfo = {};\n        }\n        return this.codecInfo;\n    }\n\n    updateVps(vps: Buffer): void {\n        this.ensureCodecInfo().vps = vps;\n    }\n\n    updateSps(sps: Buffer): void {\n        this.ensureCodecInfo().sps = sps;\n    }\n\n    updatePps(pps: Buffer): void {\n        this.ensureCodecInfo().pps = pps;\n    }\n\n    updateSeiPrefix(sei: Buffer): void {\n        this.ensureCodecInfo().seiPrefix = sei;\n    }\n\n    updateSeiSuffix(sei: Buffer): void {\n        this.ensureCodecInfo().seiSuffix = sei;\n    }\n\n    shouldFilter(_nalType: number): boolean {\n        // Currently nothing is filtered, but this could be customized\n        return false;\n    }\n\n    // Fragmentation Unit (FU) for H.265\n    // https://datatracker.ietf.org/doc/html/rfc7798#section-4.4.3\n    packetizeFU(data: Buffer, noStart?: boolean, noEnd?: boolean): Buffer[] {\n        // Handle both normal packets and FU packets.\n        const initialNalType = getNalType(data);\n\n        // Check if the data is already a fragmentation unit\n        if (initialNalType === NAL_TYPE_FU) {\n            // Extract original NAL header information\n            const originalNalType = data[2]! & 0x3F;  // 6 bits\n            const isFuStart = !!(data[2]! & 0x80);\n            const isFuEnd = !!(data[2]! & 0x40);\n            const isFuMiddle = !isFuStart && !isFuEnd;\n\n            // Reconstruct the original NAL header\n            const layerId = ((data[0]! & 0x01) << 5) | ((data[1]! & 0xF8) >> 3);\n            const tid = data[1]! & 0x07;\n\n            const originalNalHeader = Buffer.alloc(2);\n            originalNalHeader[0] = (originalNalType << 1) | (layerId >> 5);\n            originalNalHeader[1] = ((layerId & 0x1F) << 3) | tid;\n\n            data = Buffer.concat([originalNalHeader, data.subarray(FU_HEADER_SIZE)]);\n\n            if (isFuStart) {\n                noEnd = true;\n            }\n            else if (isFuEnd) {\n                noStart = true;\n            }\n            else if (isFuMiddle) {\n                noStart = true;\n                noEnd = true;\n            }\n        }\n\n        // Extract information from the NAL header\n        const nalType = getNalType(data);\n        const layerId = ((data[0]! & 0x01) << 5) | ((data[1]! & 0xF8) >> 3);\n        const tid = data[1]! & 0x07;\n\n        // Construct the FU NAL header\n        const fuNalHeader = Buffer.alloc(2);\n        fuNalHeader[0] = (NAL_TYPE_FU << 1) | (layerId >> 5);\n        fuNalHeader[1] = ((layerId & 0x1F) << 3) | tid;\n\n        // Construct FU headers for different positions\n        const fuHeaderMiddle = Buffer.from([...fuNalHeader, nalType]);\n        const fuHeaderStart = noStart ? fuHeaderMiddle : Buffer.from([...fuNalHeader, nalType | 0x80]);\n        const fuHeaderEnd = noEnd ? fuHeaderMiddle : Buffer.from([...fuNalHeader, nalType | 0x40]);\n        let fuHeader = fuHeaderStart;\n\n        const packages: Buffer[] = [];\n        let offset = NAL_HEADER_SIZE;\n\n        while (offset < data.length) {\n            let payload: Buffer;\n            const packageSize = Math.min(this.fuMax, data.length - offset);\n            payload = data.subarray(offset, offset + packageSize);\n            offset += packageSize;\n\n            if (offset === data.length) {\n                fuHeader = fuHeaderEnd;\n            }\n\n            packages.push(Buffer.concat([fuHeader, payload]));\n\n            fuHeader = fuHeaderMiddle;\n        }\n\n        return packages;\n    }\n\n    // Aggregation Packet (AP) for H.265 (similar to STAP-A in H.264)\n    // https://datatracker.ietf.org/doc/html/rfc7798#section-4.4.2\n    packetizeOneAP(datas: Buffer[]): Buffer {\n        if (!datas.length)\n            throw new Error('packetizeOneAP requires at least one NAL');\n\n        let counter = 0;\n        let availableSize = this.maxPacketSize - AP_HEADER_SIZE;\n\n        // In H.265, AP uses a fixed header with NAL type 48\n        const apHeader = Buffer.alloc(2);\n        apHeader[0] = NAL_TYPE_AP << 1;  // Type 48, no layer ID in first byte\n        apHeader[1] = 0x01;  // Default temporal ID = 1, no layer ID\n\n        const payload: Buffer[] = [apHeader];\n\n        while (datas.length && datas[0]!.length + LENGTH_FIELD_SIZE <= availableSize && counter < 9) {\n            const nalu = datas.shift()!;\n            availableSize -= LENGTH_FIELD_SIZE + nalu.length;\n            counter += 1;\n            const lengthField = Buffer.alloc(2);\n            lengthField.writeUInt16BE(nalu.length, 0);\n            payload.push(lengthField, nalu);\n        }\n\n        // If no NALUs fit, return the first one for FU packetization\n        if (counter === 0)\n            return datas.shift()!;\n\n        // A single NALU AP is unnecessary, return the NALU itself\n        if (counter === 1) {\n            return payload[2]!; // Skip header and length field\n        }\n\n        return Buffer.concat(payload);\n    }\n\n    packetizeAP(datas: Buffer[]) {\n        const ret: Buffer[] = [];\n        while (datas.length) {\n            const nalu = this.packetizeOneAP(datas);\n            if (nalu.length < this.maxPacketSize) {\n                ret.push(nalu);\n                continue;\n            }\n            const fus = this.packetizeFU(nalu);\n            ret.push(...fus);\n        }\n        return ret;\n    }\n\n    createPacket(rtp: RtpPacket, data: Buffer, marker: boolean) {\n        const ret = rtp.clone();\n        ret.header.sequenceNumber = (rtp.header.sequenceNumber + this.extraPackets + 0x10000) % 0x10000;\n        ret.header.marker = marker;\n        ret.header.padding = false;\n        ret.payload = data;\n        if (data.length > this.maxPacketSize)\n            this.console.warn('packet exceeded max packet size. this may be a bug.');\n        this.sentMarker = ret.header.marker;\n        return ret;\n    }\n\n    flushPendingFU(ret: RtpPacket[]) {\n        const pending = this.pendingFU;\n        if (!pending || pending.length === 0)\n            return;\n\n        // Defragmenting assumes packets are sorted by sequence number,\n        // and are all available, which is guaranteed over rtsp/tcp, but not over rtp/udp.\n        const first = pending[0]!;\n        const last = pending[pending.length - 1]!;\n        const originalNalType = first.payload[2]! & 0x3F;\n        const hasFuStart = !!(first.payload[2]! & 0x80);\n        const hasFuEnd = !!(last.payload[2]! & 0x40);\n\n        // Extract layerId and tid from FU header\n        const layerId = ((first.payload[0]! & 0x01) << 5) | ((first.payload[1]! & 0xF8) >> 3);\n        const tid = first.payload[1]! & 0x07;\n\n        // Reconstruct original NAL header\n        const originalNalHeader = Buffer.alloc(2);\n        originalNalHeader[0] = (originalNalType << 1) | (layerId >> 5);\n        originalNalHeader[1] = ((layerId & 0x1F) << 3) | tid;\n\n        const getDefragmentedPendingFu = (): Buffer => {\n            const originalFragments = pending.map(packet => packet.payload.subarray(FU_HEADER_SIZE));\n            originalFragments.unshift(originalNalHeader);\n            return Buffer.concat(originalFragments);\n        }\n\n        // Handle special case for VPS/SPS/PPS in FU (not standard but seen in some implementations)\n        if (originalNalType === NAL_TYPE_VPS || originalNalType === NAL_TYPE_SPS) {\n            const defragmented = getDefragmentedPendingFu();\n            const splits = splitH265NaluStartCode(defragmented);\n\n            while (splits.length) {\n                const split = splits.shift()!;\n                const splitNaluType = getNalType(split);\n\n                if (splitNaluType === NAL_TYPE_VPS) {\n                    this.updateVps(split);\n                }\n                else if (splitNaluType === NAL_TYPE_SPS) {\n                    this.updateSps(split);\n                }\n                else if (splitNaluType === NAL_TYPE_PPS) {\n                    this.updatePps(split);\n                }\n                else {\n                    // For IDR frames, send codec info first\n                    if (isKeyFrame(splitNaluType)) {\n                        this.maybeSendAPCodecInfo(first, ret);\n                    }\n\n                    this.fragment(first, ret, {\n                        payload: split,\n                        noStart: !hasFuStart,\n                        noEnd: !hasFuEnd,\n                        marker: last.header.marker,\n                    });\n                }\n            }\n        }\n        else {\n            // Process regular fragmentation units\n            while (pending.length) {\n                const fu = pending[0]!;\n                if (fu.payload.length > this.maxPacketSize || fu.payload.length < this.fuMin)\n                    break;\n                pending.shift();\n                ret.push(this.createPacket(fu, fu.payload, fu.header.marker));\n            }\n\n            if (!pending.length) {\n                this.pendingFU = undefined;\n                return;\n            }\n\n            // Re-fragment remaining FU packets\n            const refragFirst = pending[0]!;\n            const refragLast = pending[pending.length - 1]!;\n            const refragHasFuStart = !!(refragFirst.payload[2]! & 0x80);\n            const refragHasFuEnd = !!(refragLast.payload[2]! & 0x40);\n\n            const defragmented = getDefragmentedPendingFu();\n\n            this.fragment(refragFirst, ret, {\n                payload: defragmented,\n                noStart: !refragHasFuStart,\n                noEnd: !refragHasFuEnd,\n                marker: refragLast.header.marker\n            });\n        }\n\n        this.extraPackets -= pending.length - 1;\n        this.pendingFU = undefined;\n    }\n\n    createRtpPackets(packet: RtpPacket, nalus: Buffer[], ret: RtpPacket[], hadMarker = packet.header.marker) {\n        nalus.forEach((packetized, index) => {\n            if (index !== 0)\n                this.extraPackets++;\n            const marker = hadMarker && index === nalus.length - 1;\n            ret.push(this.createPacket(packet, packetized, marker));\n        });\n    }\n\n    maybeSendAPCodecInfo(packet: RtpPacket, ret: RtpPacket[]) {\n        if (this.ap) {\n            // AP with codec information was sent recently, no need to send codec info.\n            this.ap = undefined;\n            return;\n        }\n\n        // can not send codec info if in the middle of sending packets for a specific rtp timestamp.\n        if (!this.sentMarker)\n            return;\n\n        // vps is not required.\n        if (!this.codecInfo?.sps || !this.codecInfo?.pps)\n            return;\n\n        const agg = [this.codecInfo.sps, this.codecInfo.pps];\n        if (this.codecInfo.vps)\n            agg.unshift(this.codecInfo.vps);\n        if (this.codecInfo?.seiPrefix)\n            agg.push(this.codecInfo.seiPrefix);\n        if (this.codecInfo?.seiSuffix)\n            agg.push(this.codecInfo.seiSuffix);\n\n        const aggregates = this.packetizeAP(agg);\n        if (aggregates.length !== 1) {\n            this.console.error('expected only 1 packet for vps/sps/pps AP');\n            return;\n        }\n        // This AP only contains codec info (and no frame data), thus the marker bit should not be set.\n        this.createRtpPackets(packet, aggregates, ret, false);\n        this.extraPackets++;\n    }\n\n    // Fragment payload into multiple packets as needed\n    fragment(packet: RtpPacket, ret: RtpPacket[], fuOptions: {\n        payload: Buffer;\n        noStart: boolean;\n        noEnd: boolean;\n        marker: boolean;\n    } = {\n            payload: packet.payload,\n            noStart: false,\n            noEnd: false,\n            marker: packet.header.marker\n        }) {\n        const { payload, noStart, noEnd, marker } = fuOptions;\n        if (payload.length > this.maxPacketSize || noStart || noEnd) {\n            const fragments = this.packetizeFU(payload, noStart, noEnd);\n            this.createRtpPackets(packet, fragments, ret, marker);\n        }\n        else {\n            // Can send this packet as is\n            ret.push(this.createPacket(packet, payload, marker));\n        }\n    }\n\n    repacketize<T extends RtpPacket>(packet: T): T[] {\n        const ret: T[] = [];\n        for (const dejittered of this.jitterBuffer.queue(packet)) {\n            this.repacketizeOne(dejittered, ret);\n        }\n        return ret;\n    }\n\n    repacketizeOne(packet: RtpPacket, ret: RtpPacket[]) {\n        // Filter empty packets\n        if (!packet.payload.length) {\n            this.flushPendingFU(ret);\n            this.extraPackets--;\n            return;\n        }\n\n        const nalType = getNalType(packet.payload);\n\n        // Fragmented packets must share a timestamp\n        if (this.pendingFU && this.pendingFU[0]!.header.timestamp !== packet.header.timestamp) {\n            this.flushPendingFU(ret);\n        }\n\n        if (nalType === NAL_TYPE_FU) {\n            // Handle Fragmentation Units\n            const data = packet.payload;\n            const originalNalType = data[2]! & 0x3F;\n\n            if (this.shouldFilter(originalNalType)) {\n                this.extraPackets--;\n                return;\n            }\n\n            const isFuStart = !!(data[2]! & 0x80);\n            const isFuEnd = !!(data[2]! & 0x40);\n\n            if (isFuStart) {\n                if (this.pendingFU)\n                    this.console.error('FU restarted. skipping refragmentation of previous FU.', originalNalType);\n\n                this.pendingFU = undefined;\n\n                // If this is an IDR frame, but no codec info has been sent via an AP, send it\n                if (originalNalType === NAL_TYPE_IDR_W_RADL || originalNalType === NAL_TYPE_IDR_N_LP) {\n                    this.maybeSendAPCodecInfo(packet, ret);\n                }\n            }\n            else {\n                if (this.pendingFU) {\n                    // Check if packets were missing from the previously queued FU packets\n                    const last = this.pendingFU[this.pendingFU.length - 1]!;\n                    if (!isNextSequenceNumber(last.header.sequenceNumber, packet.header.sequenceNumber)) {\n                        this.console.error('FU packet missing. skipping refragmentation.', originalNalType);\n                        return;\n                    }\n                }\n            }\n\n            if (!this.pendingFU)\n                this.pendingFU = [];\n\n            this.pendingFU.push(packet);\n\n            if (isFuEnd) {\n                this.flushPendingFU(ret);\n            }\n            else if (this.pendingFU.reduce((p, c) => p + c.payload.length - FU_HEADER_SIZE, NAL_HEADER_SIZE) > this.maxPacketSize) {\n                // Refragment FU packets as they are received\n                const last = this.pendingFU[this.pendingFU.length - 1]!.clone();\n                const partial: RtpPacket[] = [];\n                this.flushPendingFU(partial);\n                // Retain a FU packet to validate subsequent FU packets\n                const retain = partial.pop()!;\n                last.payload = retain.payload;\n                this.pendingFU = [last];\n                ret.push(...partial);\n                this.sentMarker = false;\n            }\n        }\n        else if (nalType === NAL_TYPE_AP) {\n            this.flushPendingFU(ret);\n\n            let hasVps = false;\n            let hasSps = false;\n            let hasPps = false;\n\n            // Process Aggregation Packets\n            const depacketized = depacketizeAP(packet.payload);\n            depacketized.forEach(payload => {\n                const nalType = getNalType(payload);\n                if (nalType === NAL_TYPE_VPS) {\n                    hasVps = true;\n                    this.updateVps(payload);\n                }\n                else if (nalType === NAL_TYPE_SPS) {\n                    hasSps = true;\n                    this.updateSps(payload);\n                }\n                else if (nalType === NAL_TYPE_PPS) {\n                    hasPps = true;\n                    this.updatePps(payload);\n                }\n                else if (nalType === NAL_TYPE_SEI_PREFIX) {\n                    this.updateSeiPrefix(payload);\n                }\n                else if (nalType === NAL_TYPE_SEI_SUFFIX) {\n                    this.updateSeiSuffix(payload);\n                }\n                else if (nalType === NAL_TYPE_AUD) {\n                    // Access Unit Delimiter - typically a no-op\n                }\n                else if (nalType >= NAL_TYPE_TRAIL_N && nalType <= 9 /* NAL_TYPE_RASL_R */) {\n                    // Various slice types - typically VCL NAL units\n                }\n                else if (nalType === NAL_TYPE_IDR_W_RADL) {\n                    // IDR\n                }\n                else if (nalType === 0) {\n                    // NAL delimiter or something. usually empty.\n                }\n                else {\n                    this.console.warn('Skipped an AP type.', nalType);\n                }\n            });\n\n            // Log that an AP with codec info was sent\n            if (hasVps && hasSps && hasPps && packet.payload.length <= this.maxPacketSize) {\n                this.ap = packet;\n\n                const ap = this.packetizeAP(depacketized);\n                this.createRtpPackets(packet, ap, ret);\n            }\n            else {\n                // h265 can send multiple IDR or TRAIL_R per rtp timestamp\n                // so this can result in a large aggregation packet\n                // if the MTU is large (like 64k on localhost udp).\n                // the aggregation packet must be depacketized and potentially\n                // fragmented.\n                const fus: Buffer[] = [];\n                while (depacketized.length) {\n                    const next = depacketized.shift()!;\n                    if (next.length <= this.maxPacketSize) {\n                        fus.push(next);\n                        continue;\n                    }\n                    fus.push(...this.packetizeFU(next));\n                }\n                this.createRtpPackets(packet, fus, ret);\n            }\n        }\n        else if (nalType <= NAL_TYPE_RSV_IRAP_VCL23 || (nalType >= NAL_TYPE_VPS && nalType <= NAL_TYPE_SEI_SUFFIX)) {\n            this.flushPendingFU(ret);\n\n            if (this.shouldFilter(nalType)) {\n                this.extraPackets--;\n                return;\n            }\n\n            // Handle codec information\n            if (nalType === NAL_TYPE_VPS) {\n                this.extraPackets--;\n                this.updateVps(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_SPS) {\n                this.extraPackets--;\n                this.updateSps(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_PPS) {\n                this.extraPackets--;\n                this.updatePps(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_SEI_PREFIX) {\n                this.extraPackets--;\n                this.updateSeiPrefix(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_SEI_SUFFIX) {\n                this.extraPackets--;\n                this.updateSeiSuffix(packet.payload);\n                return;\n            }\n\n            if (this.shouldFilter(nalType)) {\n                this.extraPackets--;\n                return;\n            }\n\n            // For IDR frames, send codec info first\n            if (isKeyFrame(nalType)) {\n                this.maybeSendAPCodecInfo(packet, ret);\n            }\n\n            this.fragment(packet, ret);\n        }\n        else {\n            this.console.error('unknown NAL unit type ' + nalType);\n            this.extraPackets--;\n        }\n\n        return;\n    }\n}\n// Keep unused NAL_TYPE_ constants alive — they document the full RFC\n// 7798 NAL-type table even when only a subset is referenced by the\n// packetizer logic.\nvoid [_UNUSED_NAL_TYPE_TRAIL_R,_UNUSED_NAL_TYPE_TSA_N,_UNUSED_NAL_TYPE_TSA_R,_UNUSED_NAL_TYPE_STSA_N,_UNUSED_NAL_TYPE_STSA_R,_UNUSED_NAL_TYPE_RADL_N,_UNUSED_NAL_TYPE_RADL_R,_UNUSED_NAL_TYPE_RASL_N,_UNUSED_NAL_TYPE_RASL_R,_UNUSED_NAL_TYPE_VCL_N10,_UNUSED_NAL_TYPE_VCL_R11,_UNUSED_NAL_TYPE_VCL_N12,_UNUSED_NAL_TYPE_VCL_R13,_UNUSED_NAL_TYPE_VCL_N14,_UNUSED_NAL_TYPE_VCL_R15,_UNUSED_NAL_TYPE_RSV_IRAP_VCL22,_UNUSED_NAL_TYPE_RSV_VCL24,_UNUSED_NAL_TYPE_RSV_VCL25,_UNUSED_NAL_TYPE_RSV_VCL26,_UNUSED_NAL_TYPE_RSV_VCL27,_UNUSED_NAL_TYPE_RSV_VCL28,_UNUSED_NAL_TYPE_RSV_VCL29,_UNUSED_NAL_TYPE_RSV_VCL30,_UNUSED_NAL_TYPE_RSV_VCL31,_UNUSED_NAL_TYPE_EOS_NUT,_UNUSED_NAL_TYPE_EOB_NUT,_UNUSED_NAL_TYPE_FD_NUT,_UNUSED_NAL_TYPE_RSV_NVCL41,_UNUSED_NAL_TYPE_RSV_NVCL42,_UNUSED_NAL_TYPE_RSV_NVCL43,_UNUSED_NAL_TYPE_RSV_NVCL44,_UNUSED_NAL_TYPE_RSV_NVCL45,_UNUSED_NAL_TYPE_RSV_NVCL46,_UNUSED_NAL_TYPE_RSV_NVCL47,_UNUSED_NAL_TYPE_UNSPEC50,_UNUSED_NAL_TYPE_UNSPEC51,_UNUSED_NAL_TYPE_UNSPEC52,_UNUSED_NAL_TYPE_UNSPEC53,_UNUSED_NAL_TYPE_UNSPEC54,_UNUSED_NAL_TYPE_UNSPEC55,_UNUSED_NAL_TYPE_UNSPEC56,_UNUSED_NAL_TYPE_UNSPEC57,_UNUSED_NAL_TYPE_UNSPEC58,_UNUSED_NAL_TYPE_UNSPEC59,_UNUSED_NAL_TYPE_UNSPEC60,_UNUSED_NAL_TYPE_UNSPEC61,_UNUSED_NAL_TYPE_UNSPEC62,_UNUSED_NAL_TYPE_UNSPEC63]\n","/**\n * Re-fragments H.264 RTP packets coming straight off the camera so they\n * fit the WebRTC peer's MTU (~1200 bytes) WITHOUT re-encoding or rebuilding\n * the RTP header from scratch. Source SSRC/seq/timestamp are preserved\n * (only adjusted by werift's `replaceRTP` offset machinery on send), and\n * SPS/PPS are aggregated into a STAP-A sent before every IDR.\n *\n * This is the H.264 sibling of `H265Repacketizer`. Forwarding source RTP\n * through it — instead of the \"depacketize → Annex-B → re-packetize from\n * scratch\" `writeVideoNals` path — is what makes Main/High passthrough\n * decodable on iOS (WebKit), whose H.264 depacketizer is as strict as its\n * HEVC one. Desktop Chrome tolerates the synthesized shape; iOS does not.\n */\nimport { isNextSequenceNumber, JitterBuffer, RtpPacket } from \"./jitter-buffer.js\";\n\n// https://yumichan.net/video-processing/video-compression/introduction-to-h264-nal-unit/\nexport const NAL_TYPE_STAP_A = 24;\nexport const NAL_TYPE_FU_A = 28;\nexport const NAL_TYPE_NON_IDR = 1;\nexport const NAL_TYPE_IDR = 5;\nexport const NAL_TYPE_SEI = 6;\nexport const NAL_TYPE_SPS = 7;\nexport const NAL_TYPE_PPS = 8;\nexport const NAL_TYPE_DELIMITER = 9;\n\nconst NAL_HEADER_SIZE = 1;\nconst FU_A_HEADER_SIZE = 2;\nconst LENGTH_FIELD_SIZE = 2;\nconst STAP_A_HEADER_SIZE = NAL_HEADER_SIZE + LENGTH_FIELD_SIZE;\n\n// A STAP-A packet aggregates multiple NALs. Split it back into them.\nexport function depacketizeStapA(data: Buffer): Buffer[] {\n    const ret: Buffer[] = [];\n    let lastPos: number | undefined;\n    let pos = NAL_HEADER_SIZE;\n    while (pos < data.length) {\n        if (lastPos !== undefined)\n            ret.push(data.subarray(lastPos, pos));\n        const naluSize = data.readUInt16BE(pos);\n        pos += LENGTH_FIELD_SIZE;\n        lastPos = pos;\n        pos += naluSize;\n    }\n    ret.push(data.subarray(lastPos));\n    return ret;\n}\n\nexport function splitH264NaluStartCode(data: Buffer): Buffer[] {\n    const ret: Buffer[] = [];\n    let previous = 0;\n    let offset = 0;\n    const maybeAddSlice = (): void => {\n        const slice = data.subarray(previous, offset);\n        if (slice.length)\n            ret.push(slice);\n        offset += 4;\n        previous = offset;\n    };\n\n    while (offset < data.length - 4) {\n        const startCode = data.readUInt32BE(offset);\n        if (startCode === 1) {\n            maybeAddSlice();\n        }\n        else {\n            offset++;\n        }\n    }\n    offset = data.length;\n    maybeAddSlice();\n\n    return ret;\n}\n\nexport interface H264CodecInfo {\n    sps?: Buffer;\n    pps?: Buffer;\n    // SEI may only apply to a number/time range of frames; resending it with\n    // every keyframe is a best-effort to keep decoders that need the SEI in sync.\n    sei?: Buffer;\n}\n\ninterface FuaFragmentOptions {\n    payload: Buffer;\n    noStart: boolean;\n    noEnd: boolean;\n    marker: boolean;\n}\n\nexport class H264Repacketizer {\n    extraPackets = 0;\n    fuaMax!: number;\n    pendingFuA: RtpPacket[] | undefined;\n    // the stapa packet that will be sent before an idr frame.\n    stapa: RtpPacket | undefined;\n    fuaMin!: number;\n\n    constructor(public console: Console, private maxPacketSize: number, public codecInfo?: H264CodecInfo, public jitterBuffer = new JitterBuffer(console, 4)) {\n        this.setMaxPacketSize(maxPacketSize);\n    }\n\n    setMaxPacketSize(maxPacketSize: number): void {\n        this.maxPacketSize = maxPacketSize;\n        // 12 is the rtp/srtp header size.\n        this.fuaMax = maxPacketSize - FU_A_HEADER_SIZE;\n        this.fuaMin = Math.round(maxPacketSize * .8);\n    }\n\n    ensureCodecInfo(): H264CodecInfo {\n        if (!this.codecInfo) {\n            this.codecInfo = {};\n        }\n        return this.codecInfo;\n    }\n\n    updateSps(sps: Buffer): void {\n        this.ensureCodecInfo().sps = sps;\n    }\n\n    updatePps(pps: Buffer): void {\n        this.ensureCodecInfo().pps = pps;\n    }\n\n    updateSei(sei: Buffer): void {\n        this.ensureCodecInfo().sei = sei;\n    }\n\n    shouldFilter(_nalType: number): boolean {\n        // Currently nothing is filtered, but this could be customized.\n        return false;\n    }\n\n    // a fragmentation unit (fua) is a NAL unit broken into multiple fragments.\n    // https://datatracker.ietf.org/doc/html/rfc6184#section-5.8\n    packetizeFuA(data: Buffer, noStart?: boolean, noEnd?: boolean): Buffer[] {\n        // handle both normal packets and fua packets.\n        // a fua packet can be fragmented easily into smaller packets, as\n        // it is already a fragment, and splitting segments is trivial.\n\n        const initialNalType = data[0]! & 0x1f;\n\n        if (initialNalType === NAL_TYPE_FU_A) {\n            const fnri = data[0]! & (0x80 | 0x60);\n            const originalNalType = data[1]! & 0x1f;\n            const isFuStart = !!(data[1]! & 0x80);\n            const isFuEnd = !!(data[1]! & 0x40);\n            const isFuMiddle = !isFuStart && !isFuEnd;\n\n            const originalNalHeader = Buffer.from([fnri | originalNalType]);\n            data = Buffer.concat([originalNalHeader, data.subarray(FU_A_HEADER_SIZE)]);\n\n            if (isFuStart) {\n                noEnd = true;\n            }\n            else if (isFuEnd) {\n                noStart = true;\n            }\n            else if (isFuMiddle) {\n                noStart = true;\n                noEnd = true;\n            }\n        }\n\n        const fnri = data[0]! & (0x80 | 0x60);\n        const nalType = data[0]! & 0x1F;\n\n        const fuIndicator = fnri | NAL_TYPE_FU_A;\n\n        const fuHeaderMiddle = Buffer.from([fuIndicator, nalType]);\n        const fuHeaderStart = noStart ? fuHeaderMiddle : Buffer.from([fuIndicator, nalType | 0x80]);\n        const fuHeaderEnd = noEnd ? fuHeaderMiddle : Buffer.from([fuIndicator, nalType | 0x40]);\n        let fuHeader = fuHeaderStart;\n\n        const packages: Buffer[] = [];\n        let offset = NAL_HEADER_SIZE;\n\n        while (offset < data.length) {\n            const packageSize = Math.min(this.fuaMax, data.length - offset);\n            const payload = data.subarray(offset, offset + packageSize);\n            offset += packageSize;\n\n            if (offset === data.length) {\n                fuHeader = fuHeaderEnd;\n            }\n\n            packages.push(Buffer.concat([fuHeader, payload]));\n\n            fuHeader = fuHeaderMiddle;\n        }\n\n        return packages;\n    }\n\n    // https://datatracker.ietf.org/doc/html/rfc6184#section-5.7.1\n    packetizeOneStapA(datas: Buffer[]): Buffer {\n        const payload: Buffer[] = [];\n\n        if (!datas.length)\n            throw new Error('packetizeOneStapA requires at least one NAL');\n\n        let counter = 0;\n        let availableSize = this.maxPacketSize - STAP_A_HEADER_SIZE;\n\n        // h264/rtp spec: https://datatracker.ietf.org/doc/html/rfc6184#section-5.6\n        // homekit does not want NRI aggregation in the sps/pps stap-a.\n        const stapHeader = NAL_TYPE_STAP_A;\n\n        while (datas.length && datas[0]!.length + LENGTH_FIELD_SIZE <= availableSize && counter < 9) {\n            const nalu = datas.shift()!;\n            availableSize -= LENGTH_FIELD_SIZE + nalu.length;\n            counter += 1;\n            const packed = Buffer.alloc(2);\n            packed.writeUInt16BE(nalu.length, 0);\n            payload.push(packed, nalu);\n        }\n\n        // when a stapa packet has a p frame inside it, it may exceed the max packet size.\n        // it needs to be returned as is to be turned into a fua packet.\n        if (counter === 0)\n            return datas.shift()!;\n\n        // a single nalu stapa is unnecessary, return the nalu itself.\n        if (counter === 1) {\n            return payload[1]!;\n        }\n\n        payload.unshift(Buffer.from([stapHeader]));\n        return Buffer.concat(payload);\n    }\n\n    packetizeStapA(datas: Buffer[]): Buffer[] {\n        const ret: Buffer[] = [];\n        while (datas.length) {\n            const nalu = this.packetizeOneStapA(datas);\n            if (nalu.length < this.maxPacketSize) {\n                ret.push(nalu);\n                continue;\n            }\n            const fuas = this.packetizeFuA(nalu);\n            ret.push(...fuas);\n        }\n        return ret;\n    }\n\n    createPacket(rtp: RtpPacket, data: Buffer, marker: boolean): RtpPacket {\n        const ret = rtp.clone();\n        ret.header.sequenceNumber = (rtp.header.sequenceNumber + this.extraPackets + 0x10000) % 0x10000;\n        ret.header.marker = marker;\n        ret.header.padding = false;\n        ret.payload = data;\n        if (data.length > this.maxPacketSize)\n            this.console.warn('packet exceeded max packet size. this may be a bug.');\n        return ret;\n    }\n\n    flushPendingFuA(ret: RtpPacket[]): void {\n        const pending = this.pendingFuA;\n        if (!pending || pending.length === 0)\n            return;\n\n        // defragmenting assumes packets are sorted by sequence number,\n        // and are all available, which is guaranteed over rtsp/tcp, but not over rtp/udp.\n        const first = pending[0]!;\n        const last = pending[pending.length - 1]!;\n        const originalNalType = first.payload[1]! & 0x1f;\n        const hasFuStart = !!(first.payload[1]! & 0x80);\n        const hasFuEnd = !!(last.payload[1]! & 0x40);\n\n        const fnri = first.payload[0]! & (0x80 | 0x60);\n        const originalNalHeader = Buffer.from([fnri | originalNalType]);\n\n        const getDefragmentedPendingFua = (): Buffer => {\n            const originalFragments = pending.map(packet => packet.payload.subarray(FU_A_HEADER_SIZE));\n            originalFragments.unshift(originalNalHeader);\n            return Buffer.concat(originalFragments);\n        };\n\n        // have seen cameras that toss sps/pps/idr into a fua, delimited by start codes.\n        // so the fua packet looks like: sps | start code | pps | start code | idr\n        if (originalNalType === NAL_TYPE_SPS) {\n            const defragmented = getDefragmentedPendingFua();\n\n            const splits = splitH264NaluStartCode(defragmented);\n            while (splits.length) {\n                const split = splits.shift()!;\n                const splitNaluType = split[0]! & 0x1f;\n                if (splitNaluType === NAL_TYPE_SPS) {\n                    this.updateSps(split);\n                }\n                else if (splitNaluType === NAL_TYPE_PPS) {\n                    this.updatePps(split);\n                }\n                else {\n                    if (splitNaluType === NAL_TYPE_IDR)\n                        this.maybeSendStapACodecInfo(first, ret);\n\n                    this.fragment(first, ret, {\n                        payload: split,\n                        noStart: !hasFuStart,\n                        noEnd: !hasFuEnd,\n                        marker: last.header.marker,\n                    });\n                }\n            }\n        }\n        else {\n            while (pending.length) {\n                const fua = pending[0]!;\n                if (fua.payload.length > this.maxPacketSize || fua.payload.length < this.fuaMin)\n                    break;\n                pending.shift();\n                ret.push(this.createPacket(fua, fua.payload, fua.header.marker));\n            }\n\n            if (!pending.length) {\n                this.pendingFuA = undefined;\n                return;\n            }\n\n            const refragFirst = pending[0]!;\n            const refragLast = pending[pending.length - 1]!;\n            const refragHasFuStart = !!(refragFirst.payload[1]! & 0x80);\n            const refragHasFuEnd = !!(refragLast.payload[1]! & 0x40);\n\n            const defragmented = getDefragmentedPendingFua();\n\n            this.fragment(refragFirst, ret, {\n                payload: defragmented,\n                noStart: !refragHasFuStart,\n                noEnd: !refragHasFuEnd,\n                marker: refragLast.header.marker,\n            });\n        }\n\n        this.extraPackets -= pending.length - 1;\n        this.pendingFuA = undefined;\n    }\n\n    createRtpPackets(packet: RtpPacket, nalus: Buffer[], ret: RtpPacket[], hadMarker = packet.header.marker): void {\n        nalus.forEach((packetized, index) => {\n            if (index !== 0)\n                this.extraPackets++;\n            const marker = hadMarker && index === nalus.length - 1;\n            ret.push(this.createPacket(packet, packetized, marker));\n        });\n    }\n\n    maybeSendStapACodecInfo(packet: RtpPacket, ret: RtpPacket[]): void {\n        if (this.stapa) {\n            // stapa with codec information was sent recently, no need to send codec info.\n            this.stapa = undefined;\n            return;\n        }\n\n        if (!this.codecInfo?.sps || !this.codecInfo?.pps)\n            return;\n\n        const agg = [this.codecInfo.sps, this.codecInfo.pps];\n        if (this.codecInfo?.sei)\n            agg.push(this.codecInfo.sei);\n        const aggregates = this.packetizeStapA(agg);\n        if (aggregates.length !== 1) {\n            this.console.error('expected only 1 packet for sps/pps stapa');\n            return;\n        }\n        // this stapa only contains sps and pps (and no frame data), thus the marker bit should not be set.\n        this.createRtpPackets(packet, aggregates, ret, false);\n        this.extraPackets++;\n    }\n\n    // given the packet, fragment it into multiple packets as needed.\n    // a fragment of a payload may be provided via fuaOptions.\n    fragment(packet: RtpPacket, ret: RtpPacket[], fuaOptions: FuaFragmentOptions = {\n        payload: packet.payload,\n        noStart: false,\n        noEnd: false,\n        marker: packet.header.marker,\n    }): void {\n        const { payload, noStart, noEnd, marker } = fuaOptions;\n        if (payload.length > this.maxPacketSize || noStart || noEnd) {\n            const fragments = this.packetizeFuA(payload, noStart, noEnd);\n            this.createRtpPackets(packet, fragments, ret, marker);\n        }\n        else {\n            // can send this packet as is!\n            ret.push(this.createPacket(packet, payload, marker));\n        }\n    }\n\n    repacketize<T extends RtpPacket>(packet: T): T[] {\n        const ret: T[] = [];\n        for (const dejittered of this.jitterBuffer.queue(packet)) {\n            this.repacketizeOne(dejittered, ret);\n        }\n        return ret;\n    }\n\n    repacketizeOne(packet: RtpPacket, ret: RtpPacket[]): void {\n        // empty packets are apparently valid from webrtc. filter those out.\n        if (!packet.payload.length) {\n            this.flushPendingFuA(ret);\n            this.extraPackets--;\n            return;\n        }\n\n        const nalType = packet.payload[0]! & 0x1F;\n\n        // fragmented packets must share a timestamp\n        if (this.pendingFuA && this.pendingFuA[0]!.header.timestamp !== packet.header.timestamp) {\n            this.flushPendingFuA(ret);\n        }\n\n        if (nalType === NAL_TYPE_FU_A) {\n            // ideally send the packets through as is from the upstream source.\n            // refragment only if the incoming fua packet is larger than the max packet size.\n            const data = packet.payload;\n            const originalNalType = data[1]! & 0x1f;\n\n            if (this.shouldFilter(originalNalType)) {\n                this.extraPackets--;\n                return;\n            }\n\n            const isFuStart = !!(data[1]! & 0x80);\n\n            if (isFuStart) {\n                if (this.pendingFuA)\n                    this.console.error('fua restarted. skipping refragmentation of previous fua.', originalNalType);\n\n                this.pendingFuA = undefined;\n\n                // if this is an idr frame, but no sps has been sent via a stapa, dummy one up.\n                // the stream may not contain codec information in stapa or may be sending it\n                // in separate sps/pps packets.\n                if (originalNalType === NAL_TYPE_IDR) {\n                    this.maybeSendStapACodecInfo(packet, ret);\n                }\n            }\n            else {\n                if (this.pendingFuA) {\n                    // check if packets were missing earlier from the previously queued fua packets.\n                    // if so, don't queue the current packet. all further fua packets are dropped\n                    // until a later fua start is received.\n                    const last = this.pendingFuA[this.pendingFuA.length - 1]!;\n                    if (!isNextSequenceNumber(last.header.sequenceNumber, packet.header.sequenceNumber)) {\n                        this.console.error('fua packet missing. skipping refragmentation.', originalNalType);\n                        return;\n                    }\n                }\n            }\n\n            if (!this.pendingFuA)\n                this.pendingFuA = [];\n\n            this.pendingFuA.push(packet);\n\n            const isFuEnd = !!(packet.payload[1]! & 0x40);\n            if (isFuEnd) {\n                this.flushPendingFuA(ret);\n            }\n            else if (this.pendingFuA.reduce((p, c) => p + c.payload.length - FU_A_HEADER_SIZE, NAL_HEADER_SIZE) > this.maxPacketSize) {\n                // refragment fua packets as they are received, saving the last undersized packet for\n                // the next fua packet.\n                const last = this.pendingFuA[this.pendingFuA.length - 1]!.clone();\n                const partial: RtpPacket[] = [];\n                this.flushPendingFuA(partial);\n                // retain a fua packet to validate subsequent fua packets.\n                const retain = partial.pop();\n                if (retain)\n                    last.payload = retain.payload;\n                this.pendingFuA = [last];\n                ret.push(...partial);\n            }\n        }\n        else if (nalType === NAL_TYPE_STAP_A) {\n            this.flushPendingFuA(ret);\n\n            let hasSps = false;\n            let hasPps = false;\n\n            // break the aggregated packet up to update codec information.\n            const depacketized = depacketizeStapA(packet.payload);\n            depacketized.forEach(payload => {\n                const stapNalType = payload[0]! & 0x1F;\n                if (stapNalType === NAL_TYPE_SPS) {\n                    hasSps = true;\n                    this.updateSps(payload);\n                }\n                else if (stapNalType === NAL_TYPE_PPS) {\n                    hasPps = true;\n                    this.updatePps(payload);\n                }\n                else if (stapNalType === NAL_TYPE_SEI) {\n                    this.updateSei(payload);\n                }\n                // NAL_TYPE_DELIMITER / NON_IDR / IDR / 0 inside a stapa are uncommon\n                // but harmless — re-aggregated below as-is.\n            });\n\n            // log that a stapa with codec info was sent\n            if (hasSps && hasPps)\n                this.stapa = packet;\n\n            const stapa = this.packetizeStapA(depacketized);\n            this.createRtpPackets(packet, stapa, ret);\n        }\n        else if (nalType >= 1 && nalType < 24) {\n            this.flushPendingFuA(ret);\n\n            if (this.shouldFilter(nalType)) {\n                this.extraPackets--;\n                return;\n            }\n\n            // codec information should be aggregated into a stapa. usually around 50 bytes total.\n            if (nalType === NAL_TYPE_SPS) {\n                this.extraPackets--;\n                this.updateSps(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_PPS) {\n                this.extraPackets--;\n                this.updatePps(packet.payload);\n                return;\n            }\n            else if (nalType === NAL_TYPE_SEI) {\n                this.extraPackets--;\n                this.updateSei(packet.payload);\n                return;\n            }\n\n            if (nalType === NAL_TYPE_IDR) {\n                // if this is an idr frame, but no sps has been sent, dummy one up.\n                this.maybeSendStapACodecInfo(packet, ret);\n            }\n\n            this.fragment(packet, ret);\n        }\n        else {\n            this.console.error('unknown nal unit type ' + nalType);\n            this.extraPackets--;\n        }\n    }\n}\n","/**\n * Adaptive WebRTC session — extends the base session pattern with:\n * - RTCP Receiver Report monitoring (packet loss, jitter, RTT)\n * - Source replacement (replaceTrack for seamless quality switching)\n * - Stats emission for the AdaptiveController\n *\n * Uses werift (optional peer dependency) for server-side WebRTC.\n */\n\nimport type {\n  FrameSource,\n  Logger,\n  VideoCodec,\n  AudioCodec,\n} from \"./types.js\";\nimport { splitAnnexBToNals } from \"./nal-utils.js\";\nimport { convertH264ToAnnexB, isH264IdrAccessUnit, groupNalsIntoAccessUnits } from \"./h264-utils.js\";\nimport { convertH265ToAnnexB, isH265IrapAccessUnit } from \"./h265-utils.js\";\nimport { resolveMdnsCandidatesInSdp } from \"./mdns-resolve.js\";\nimport { networkInterfaces } from \"node:os\";\nimport { H265Repacketizer, type H265CodecInfo } from \"./h265-repacketizer.js\";\nimport { H264Repacketizer } from \"./h264-repacketizer.js\";\nimport type {\n  WeriftModule,\n  WeriftPeerConnection,\n  WeriftMediaStreamTrack,\n  WeriftRtpSender,\n  WeriftTransceiver,\n  WeriftPcOptions,\n  WeriftIceServer,\n  WeriftReceiverReport,\n  WeriftIceCandidate,\n} from \"./werift-types.js\";\n\n/** A trickle ICE candidate exchanged over the signaling channel. */\nexport interface IceCandidatePayload {\n  candidate: string;\n  sdpMid: string | null;\n  sdpMLineIndex: number | null;\n}\nimport { errMsg, type EncodeProfile, type WebrtcStreamTarget } from '@camstack/types'\nimport { WEBRTC_FALLBACK_PROFILE } from './webrtc-fallback-profile.js'\nimport { audioEncoderArgs, logBannerArgs } from '../stream-broker/ffmpeg-args-common.js'\n\n// ---------------------------------------------------------------------------\n// Werift type aliases (lazy-loaded optional peer dep)\n// ---------------------------------------------------------------------------\n\nlet _werift: WeriftModule | undefined;\n\nasync function loadWerift(): Promise<WeriftModule> {\n  if (_werift) return _werift;\n  try {\n    // werift is patched in place via patches/werift+0.22.9.patch (repo-root patch-package):\n    // under iceTransportPolicy:'relay' it now emits a relay-only SDP (forceTurn candidate-leak fix)\n    // so remote/NAT viewers don't nominate a dead host candidate. Do not bump/replace werift\n    // without re-vendoring that patch — see docs/superpowers/specs/2026-05-25-werift-turn-nat-fix-design.md\n    const moduleName = \"werift\";\n    // Dynamic import via Function to avoid bundler static analysis\n    // eslint-disable-next-line @typescript-eslint/no-unsafe-call -- dynamic import via Function to avoid bundler static analysis\n    _werift = await (Function(\"m\", \"return import(m)\")(moduleName) as Promise<WeriftModule>);\n    return _werift!;\n  } catch {\n    throw new Error(\n      \"The 'werift' package is required for WebRTC support but is not installed. \" +\n      \"Install it with: npm install werift\",\n    );\n  }\n}\n\n// ---------------------------------------------------------------------------\n// Tailscale host-candidate detection\n// ---------------------------------------------------------------------------\n\n/**\n * Minimal shape of an `os.networkInterfaces()` entry we rely on. Kept\n * structural (not the full `os.NetworkInterfaceInfo`) so the detector\n * can be unit-tested with a hand-rolled map and no `os` mock.\n */\nexport interface NetworkInterfaceAddress {\n  readonly address: string;\n  readonly family: string | number;\n  readonly internal: boolean;\n}\n\n/** Map shape returned by `os.networkInterfaces()` (subset we consume). */\nexport type NetworkInterfaceMap = Readonly<\n  Record<string, readonly NetworkInterfaceAddress[] | undefined>\n>;\n\n/** True for an IPv4 address inside the Tailscale CGNAT range 100.64.0.0/10. */\nfunction isTailscaleIpv4(address: string): boolean {\n  const parts = address.split(\".\");\n  if (parts.length !== 4) return false;\n  const a = Number.parseInt(parts[0] ?? \"\", 10);\n  const b = Number.parseInt(parts[1] ?? \"\", 10);\n  if (Number.isNaN(a) || Number.isNaN(b)) return false;\n  // 100.64.0.0 – 100.127.255.255\n  return a === 100 && b >= 64 && b <= 127;\n}\n\n/** True for an IPv6 address inside the Tailscale ULA range fd7a::/16. */\nfunction isTailscaleIpv6(address: string): boolean {\n  // Strip a zone id (`fd7a::1%utun4`) before the prefix check.\n  const pct = address.indexOf(\"%\");\n  const clean = (pct === -1 ? address : address.slice(0, pct)).toLowerCase();\n  return clean.startsWith(\"fd7a:\");\n}\n\n/**\n * Enumerate the host's network interfaces and return any Tailscale\n * overlay addresses — IPv4 in 100.64.0.0/10 (CGNAT) and IPv6 in\n * fd7a::/16 (ULA). Loopback/internal interfaces are skipped.\n *\n * werift gathers ICE candidates from its own socket binds and (in\n * 0.22.9) skips point-to-point `utun` interfaces, so a Tailscale host\n * candidate is never offered by default. Feeding the detected\n * address(es) into `pcOptions.iceAdditionalHostAddresses` makes werift\n * advertise them explicitly, so a remote Tailscale viewer can form a\n * direct host↔host pair (native H.264, no relay, no re-encode).\n *\n * The address is detected dynamically — never hardcoded — so it follows\n * the host's actual Tailscale assignment.\n *\n * @param interfaces injected for testing; defaults to `os.networkInterfaces()`.\n */\nexport function getTailscaleHostAddresses(\n  interfaces: NetworkInterfaceMap = networkInterfaces(),\n): readonly string[] {\n  const found = new Set<string>();\n  for (const entries of Object.values(interfaces)) {\n    if (!entries) continue;\n    for (const entry of entries) {\n      if (entry.internal) continue;\n      const fam = entry.family;\n      const isV4 = fam === \"IPv4\" || fam === 4;\n      const isV6 = fam === \"IPv6\" || fam === 6;\n      if (isV4 && isTailscaleIpv4(entry.address)) found.add(entry.address);\n      else if (isV6 && isTailscaleIpv6(entry.address)) {\n        // Drop any zone id so werift sees a bare literal.\n        const pct = entry.address.indexOf(\"%\");\n        found.add(pct === -1 ? entry.address : entry.address.slice(0, pct));\n      }\n    }\n  }\n  return [...found];\n}\n\n// ---------------------------------------------------------------------------\n// Video transcode decision\n// ---------------------------------------------------------------------------\n\nexport interface VideoTranscodeInput {\n  /** Camera source video codec. */\n  readonly sourceCodec: 'H264' | 'H265';\n  /** Codec the browser selected in its SDP answer. */\n  readonly negotiatedCodec: 'H264' | 'H265';\n  /**\n   * True when the H.264 source profile is non-Baseline (Main/High) so the\n   * egress must be re-encoded to Constrained Baseline for iOS. Decided by\n   * the broker from the camera's SPS; irrelevant for H.265 sources.\n   */\n  readonly transcodeToBaseline: boolean;\n}\n\n/**\n * Whether the session must run ffmpeg to produce a browser-decodable\n * egress instead of forwarding the camera's native bytes:\n *\n *  - H.265 source + H.264-only browser       → transcode (pre-existing).\n *  - H.264 Main/High source + H.264 browser   → transcode to Baseline (iOS fix).\n *  - Baseline H.264, or H.265↔H.265           → passthrough (zero re-encode).\n */\nexport function resolveVideoTranscode(input: VideoTranscodeInput): boolean {\n  const { sourceCodec, negotiatedCodec, transcodeToBaseline } = input;\n  if (sourceCodec === 'H265' && negotiatedCodec === 'H264') return true;\n  if (sourceCodec === 'H264' && negotiatedCodec === 'H264' && transcodeToBaseline) return true;\n  return false;\n}\n\n/**\n * Read the video codec the peer settled on from a negotiated SDP — the\n * browser's answer (server-offer flow) or the server's answer to a client\n * offer (client-offer flow). Returns the first `H264`/`H265` rtpmap codec\n * (the primary payload type listed first on the `m=video` line), or\n * undefined when the SDP carries no video. Drives `negotiatedCodec`, which\n * `needsTranscode` reads to decide passthrough vs ffmpeg.\n */\nexport function detectNegotiatedVideoCodec(\n  sdp: string,\n  preferred?: 'H264' | 'H265',\n): 'H264' | 'H265' | undefined {\n  // The werift client-offer answer lists H.264 BEFORE H.265 (it preserves the\n  // browser's offer order, which puts the 10+ H.264 PTs ahead of the two\n  // H.265 PTs). The raw \"first match\" would then return H.264 even when both\n  // codecs were negotiated and we configured H.265 as preferred. With a\n  // `preferred` hint: if that codec appears anywhere in the SDP, return it;\n  // otherwise fall back to the first-match behaviour. Callers thread\n  // `this.sourceCodec` so an H.265 source ends up on the H.265 wire path.\n  if (preferred && new RegExp(`a=rtpmap:\\\\d+ ${preferred}/90000`, 'i').test(sdp)) {\n    return preferred;\n  }\n  const m = sdp.match(/a=rtpmap:\\d+ (H264|H265)\\/90000/i)?.[1]?.toUpperCase();\n  return m === 'H264' || m === 'H265' ? m : undefined;\n}\n\n/** Parsed view of one SDP `a=candidate:` line — the fields a debug log needs. */\nexport interface CandidateSummary {\n  /** ICE candidate type: `host` | `srflx` | `prflx` | `relay` | `unknown`. */\n  readonly type: string;\n  /** `address:port` (the connection address advertised in the candidate). */\n  readonly address: string;\n  /** `udp` | `tcp` | the raw transport token. */\n  readonly protocol: string;\n}\n\n/**\n * Parse a single SDP candidate line into `{ type, address, protocol }`.\n * The grammar (RFC 5245 §15.1) is:\n *   `candidate:<foundation> <component> <transport> <priority> <addr> <port> typ <type> ...`\n * possibly prefixed with `a=`. Returns an `unknown`-typed summary when the\n * line doesn't match — never throws, so it's safe in a debug log path.\n */\nexport function parseCandidateLine(line: string): CandidateSummary {\n  const body = line.startsWith('a=') ? line.slice(2) : line;\n  const parts = body.trim().split(/\\s+/);\n  // parts: ['candidate:F', comp, transport, prio, addr, port, 'typ', type, ...]\n  const protocol = (parts[2] ?? 'unknown').toLowerCase();\n  const addr = parts[4];\n  const port = parts[5];\n  const typIdx = parts.indexOf('typ');\n  const type = typIdx >= 0 ? parts[typIdx + 1] ?? 'unknown' : 'unknown';\n  const address = addr !== undefined && port !== undefined ? `${addr}:${port}` : 'unknown';\n  return { type, address, protocol };\n}\n\n/**\n * Extract every `a=candidate:` line from an SDP and summarise each as\n * `{ type, address, protocol }`. Used by the advanced WebRTC debug log to\n * SHOW exactly which local candidates we put in the answer (the suspected\n * unreachable Tailscale/private host candidates on the Alexa path) and which\n * remote candidates the peer offered.\n */\nexport function summarizeCandidates(sdp: string): readonly CandidateSummary[] {\n  return sdp\n    .split('\\n')\n    .map((l) => l.trim())\n    .filter((l) => l.startsWith('a=candidate:'))\n    .map(parseCandidateLine);\n}\n\n/**\n * Extract the `a=fmtp:` and `a=rtpmap:` lines from an SDP — the codec\n * parameter lines carrying `profile-level-id` and the codec/clock mapping.\n * Used by the advanced WebRTC debug log to compare offered vs negotiated\n * codec profiles (the iOS/Alexa H.264 passthrough discriminator).\n */\nexport function extractFmtpLines(sdp: string): readonly string[] {\n  return sdp\n    .split('\\n')\n    .map((l) => l.trim())\n    .filter((l) => l.startsWith('a=fmtp:') || l.startsWith('a=rtpmap:'));\n}\n\n/**\n * ffmpeg `-f` input demuxer for the transcode feed, keyed on the source\n * codec: H.265 Annex-B → `hevc`, H.264 Annex-B → `h264`. Output is always\n * Constrained Baseline H.264 (see `startTranscodeFeed`).\n */\nexport function transcodeInputFormat(sourceCodec: 'H264' | 'H265'): 'hevc' | 'h264' {\n  return sourceCodec === 'H265' ? 'hevc' : 'h264';\n}\n\n// ---------------------------------------------------------------------------\n// A/V sync diagnostic timeline math\n// ---------------------------------------------------------------------------\n\n/**\n * One track's sampled clock state at the instant the diagnostic runs: the\n * last RTP timestamp we sent on that track, that track's RTP clock rate\n * (90000 for video, 8000 for PCMU audio), and the wall-clock elapsed since\n * the session was created. Pure input — no werift, no timers.\n */\nexport interface AvSyncTrackSample {\n  /** Last RTP timestamp sent on this track (RTP ticks at `rate`). */\n  readonly rtpTs: number;\n  /** RTP clock rate in Hz (90000 video / 8000 PCMU audio). */\n  readonly rate: number;\n  /** Wall-clock ms elapsed since the session was created. */\n  readonly elapsedMs: number;\n}\n\n/** Per-track derived timeline figures for the av-sync diagnostic. */\nexport interface AvSyncTrackResult {\n  /** Media-time = rtpTs / rate, in seconds. */\n  readonly mediaSeconds: number;\n  /** Skew = media-time − wall-clock-elapsed (seconds). Negative ⇒ the RTP\n   *  clock has fallen behind real time. */\n  readonly skewSeconds: number;\n}\n\n/** Result of the av-sync timeline computation for one diagnostic tick. */\nexport interface AvSyncSkewResult {\n  /** Video track figures, or null when video is absent / rate is invalid. */\n  readonly video: AvSyncTrackResult | null;\n  /** Audio track figures, or null when audio is absent / rate is invalid. */\n  readonly audio: AvSyncTrackResult | null;\n  /**\n   * Inter-track skew = audio media-seconds − video media-seconds, in seconds.\n   * Negative ⇒ the audio media clock lags the video media clock (the suspected\n   * \"audio plays ~0.5s behind video\" signature). null when either track is\n   * absent / has an invalid clock rate, so no comparison is possible.\n   */\n  readonly interTrackSkewSeconds: number | null;\n}\n\n/** Convert one sampled track to media-seconds + per-track skew, or null when\n *  the clock rate is non-positive (would divide by zero). */\nfunction deriveAvSyncTrack(sample: AvSyncTrackSample | null): AvSyncTrackResult | null {\n  if (!sample || sample.rate <= 0) return null;\n  const mediaSeconds = sample.rtpTs / sample.rate;\n  const skewSeconds = mediaSeconds - sample.elapsedMs / 1000;\n  return { mediaSeconds, skewSeconds };\n}\n\n/**\n * Compute the per-track and inter-track timeline skew for the av-sync\n * diagnostic. Given the last sent RTP timestamp, clock rate, and wall-clock\n * elapsed for each track, derive media-time (rtpTs/rate) and how far each\n * track's media clock has drifted from real time, plus the inter-track skew\n * (audio media-seconds − video media-seconds) that quantifies the observed\n * \"audio behind video\" lag. A missing track or an invalid (≤0) clock rate\n * yields null for that track and null for the inter-track figure. Pure.\n */\nexport function computeAvSyncSkew(input: {\n  readonly video: AvSyncTrackSample | null;\n  readonly audio: AvSyncTrackSample | null;\n}): AvSyncSkewResult {\n  const video = deriveAvSyncTrack(input.video);\n  const audio = deriveAvSyncTrack(input.audio);\n  const interTrackSkewSeconds =\n    video && audio ? audio.mediaSeconds - video.mediaSeconds : null;\n  return { video, audio, interTrackSkewSeconds };\n}\n\n/**\n * Input to `computeAudioSilencePadPackets` — the pure decision of how many\n * PCMU silence packets to pre-stream BEFORE the first real audio packet so the\n * audio first-send wall-clock aligns to the video first-send wall-clock. Used\n * by `AdaptiveSession.writeAudio` once per session, on the very first audio\n * frame, to close the constant startup gap seen on transcoded sources (e.g.\n * Reolink AAC → PCMU runs ~400 ms behind video → browser inflates audio jitter\n * buffer → user perceives ~1 s of audio-late lag).\n */\nexport interface AudioSilencePadInput {\n  /** Wall-clock at the moment the first real audio frame is about to send (ms). */\n  readonly nowMs: number;\n  /** Wall-clock at which the FIRST video RTP packet was sent (ms), or null\n   *  when video hasn't sent yet — no anchor → emit nothing. */\n  readonly firstVideoSendWallMs: number | null;\n  /** PCMU frame size in samples per packet (160 = 20 ms at 8 kHz). */\n  readonly framesizeSamples: number;\n  /** Audio sample rate (Hz). PCMU is 8 kHz. */\n  readonly rateHz: number;\n  /** Upper bound on the silence pre-stream duration so a runaway gap can't\n   *  blast tens of seconds of silence onto the wire. */\n  readonly maxPadMs: number;\n}\n\n/**\n * Decide how many PCMU silence packets to pre-stream BEFORE the first real\n * audio packet so audio `firstSendWallMs` aligns to video `firstSendWallMs`.\n * Pure — no I/O, no clock reads, no side effects.\n *\n * Returns 0 when there's nothing to align to (no video anchor, gap ≤ 0,\n * sub-packet gap), and caps the count at `ceil(maxPadMs / frameMs)` to bound\n * the worst case. For positive gaps of one packet or more it returns\n * `ceil(gap / frameMs)` so the silence FULLY covers the measured gap.\n *\n * frameMs = `framesizeSamples / rateHz * 1000`. With PCMU 8 kHz / 160 samples\n * that's 20 ms — so a 401 ms gap returns 21 packets, and a 30 s gap saturates\n * at 100 packets (the 2000 ms cap / 20 ms-per-packet).\n */\nexport function computeAudioSilencePadPackets(input: AudioSilencePadInput): number {\n  if (input.firstVideoSendWallMs === null) return 0;\n  if (input.framesizeSamples <= 0 || input.rateHz <= 0) return 0;\n  if (input.maxPadMs <= 0) return 0;\n  const gapMs = input.nowMs - input.firstVideoSendWallMs;\n  if (gapMs <= 0) return 0;\n  const frameMs = (input.framesizeSamples / input.rateHz) * 1000;\n  if (gapMs < frameMs) return 0;\n  const desired = Math.ceil(gapMs / frameMs);\n  const cap = Math.ceil(input.maxPadMs / frameMs);\n  return Math.min(desired, cap);\n}\n\n/** One scheduled silence packet on the audio pre-stream plan — what\n *  `planAudioSilencePadPackets` emits and the writer turns into a werift\n *  `RtpPacket` (PCMU payload of `framesizeSamples` × `0xFF`). */\nexport interface AudioSilencePadPacketPlan {\n  /** RTP timestamp at the audio clock rate (already `>>> 0`-normalised). */\n  readonly rtpTs: number;\n  /** Sequence number (`& 0xffff`-normalised), contiguous from the prior\n   *  `audioSeqNum` and ending one BEFORE the real packet's seq. */\n  readonly seq: number;\n}\n\n/** Result of `planAudioSilencePadPackets` — the silence packets to emit + the\n *  updated `audioSeqNum` to carry forward into the real packet's seq. The plan\n *  is empty when `computeAudioSilencePadPackets` returns 0. */\nexport interface AudioSilencePadPlan {\n  readonly packets: readonly AudioSilencePadPacketPlan[];\n  /** The sequence number of the LAST silence packet emitted, OR the input\n   *  `audioSeqNum` when no packets were emitted. The next real packet's seq\n   *  is `(nextAudioSeqNum + 1) & 0xffff`. */\n  readonly nextAudioSeqNum: number;\n}\n\n/** Input to `planAudioSilencePadPackets` — same parameters as\n *  `computeAudioSilencePadPackets` plus the real first audio packet's `rtpTs`\n *  (so silence timestamps anchor contiguously to it) and the current\n *  `audioSeqNum` (so silence seqs continue from it). */\nexport interface AudioSilencePadPlanInput extends AudioSilencePadInput {\n  /** RTP timestamp the FIRST real audio packet would carry (already\n   *  `>>> 0`-normalised). Silence packets occupy `[realRtpTs − N·160, realRtpTs)`. */\n  readonly realRtpTs: number;\n  /** Current `audioSeqNum` BEFORE any silence packet. The first silence\n   *  packet's seq is `(audioSeqNum + 1) & 0xffff`. */\n  readonly audioSeqNum: number;\n}\n\n/**\n * Pure plan-builder for the PCMU silence pre-stream — turns\n * `computeAudioSilencePadPackets`'s count into the concrete (rtpTs, seq) list\n * the audio writer emits, plus the updated `audioSeqNum` to carry forward into\n * the real packet's seq.\n *\n * Silence timestamps run `realRtpTs − N·framesizeSamples … realRtpTs −\n * framesizeSamples`, then the real packet at `realRtpTs` — fully contiguous in\n * both ts and seq (32-bit/16-bit wrap-safe via `>>> 0` / `& 0xffff`). No I/O,\n * no clock reads, no side effects — safe to unit-test in isolation.\n */\nexport function planAudioSilencePadPackets(input: AudioSilencePadPlanInput): AudioSilencePadPlan {\n  const packetCount = computeAudioSilencePadPackets(input);\n  if (packetCount <= 0) {\n    return { packets: [], nextAudioSeqNum: input.audioSeqNum };\n  }\n  const frameSize = input.framesizeSamples;\n  const firstSilenceRtpTs = (input.realRtpTs - packetCount * frameSize) >>> 0;\n  const packets: AudioSilencePadPacketPlan[] = [];\n  let seq = input.audioSeqNum;\n  for (let i = 0; i < packetCount; i++) {\n    seq = (seq + 1) & 0xffff;\n    packets.push({\n      rtpTs: (firstSilenceRtpTs + i * frameSize) >>> 0,\n      seq,\n    });\n  }\n  return { packets, nextAudioSeqNum: seq };\n}\n\n/**\n * Build the ffmpeg argv for a transcode of `inputFormat` (h264 / hevc)\n * source bytes into the consumer profile's video/audio codecs.\n *\n * Layout:\n *   [global flags] [profile.inputArgs] -i pipe:0\n *   [video encode block] [audio encode block] [profile.outputArgs]\n *   -f h264 pipe:1\n *\n * Smart video copy:\n *   The operator's preset is the CONSUMER's requirement (\"I need h264\n *   output\"). When the source already speaks the requested codec, the\n *   broker elides the encode block entirely — no `-vf` / `-preset` /\n *   `-r` / `-maxrate` — and runs ffmpeg as a re-muxer on the video\n *   plane. Width / height / fps / bitrate constraints in the profile\n *   are treated as a downstream BUDGET, not a forced rescale: pick a\n *   source stream slot that already fits and let `-c:v copy` carry\n *   it. Operators who genuinely want a forced re-encode (rebrand,\n *   bitrate trim, GOP knobs, …) pick \"Custom\" in the widget — that\n *   path lands here with codec === source codec and STILL copies,\n *   because the broker cannot know which non-codec knob was the\n *   reason for opening the editor. If a forced re-encode at the\n *   source codec is ever required we'll add an explicit `force: true`\n *   flag to `VideoEncode`.\n *\n *   The explicit `codec: 'copy'` sentinel keeps working — it's the\n *   widget's \"Passthrough\" preset and the wire payload for legacy\n *   contributions. Both paths emit `-c:v copy` identically.\n *\n * - `audio === 'passthrough'` appends `-an`. Otherwise the audio\n *   encode block (`-c:a`, `-b:a`, `-ar`, `-ac`) lands before any\n *   outputArgs. Audio smart-copy needs the source audio codec\n *   plumbed alongside `inputFormat` and isn't here yet — for now the\n *   audio plane honours the profile literally.\n */\nexport function buildTranscodeFfmpegArgs(\n  inputFormat: 'hevc' | 'h264',\n  profile: EncodeProfile,\n  decodeHwAccel?: string | null,\n): string[] {\n  const args: string[] = [...logBannerArgs('error')]\n  if (profile.inputArgs?.length) args.push(...profile.inputArgs)\n  // Hardware decode (`-hwaccel`) is an input option — emit it right before the\n  // input. `null`/`'none'` ⇒ software decode (current default).\n  if (decodeHwAccel && decodeHwAccel !== 'none') args.push('-hwaccel', decodeHwAccel)\n  args.push('-f', inputFormat, '-i', 'pipe:0')\n\n  const v = profile.video\n  const sourceCodec: 'h264' | 'h265' = inputFormat === 'hevc' ? 'h265' : 'h264'\n  const shouldCopyVideo = v.codec === 'copy' || v.codec === sourceCodec\n  if (shouldCopyVideo) {\n    args.push('-c:v', 'copy')\n  } else {\n    if (v.width !== undefined && v.height !== undefined) {\n      args.push('-vf', `scale='min(${v.width},iw)':'min(${v.height},ih)':force_original_aspect_ratio=decrease:force_divisible_by=2`)\n    }\n    const encoder = v.codec === 'h265' ? 'libx265' : 'libx264'\n    args.push('-c:v', encoder)\n    if (v.preset)              args.push('-preset', v.preset)\n    if (v.tune)                args.push('-tune', v.tune)\n    if (v.profile)             args.push('-profile:v', v.profile)\n    if (v.fps !== undefined)        args.push('-r', String(v.fps))\n    if (v.gopFrames !== undefined)  args.push('-g', String(v.gopFrames))\n    if (v.bf !== undefined)         args.push('-bf', String(v.bf))\n    if (v.bitrateKbps !== undefined) {\n      args.push('-maxrate', `${v.bitrateKbps}k`, '-bufsize', `${v.bitrateKbps * 2}k`)\n    }\n  }\n\n  if (profile.audio === 'passthrough') {\n    args.push('-an')\n  } else {\n    const a = profile.audio\n    args.push(...audioEncoderArgs(a.codec, {\n      bitrateKbps: a.bitrateKbps,\n      sampleRateHz: a.sampleRateHz,\n      channels: a.channels,\n    }))\n  }\n\n  if (profile.outputArgs?.length) args.push(...profile.outputArgs)\n  args.push('-f', 'h264', 'pipe:1')\n  return args\n}\n\n// ---------------------------------------------------------------------------\n// Public types\n// ---------------------------------------------------------------------------\n\n/**\n * Unified ICE server entry — RTCIceServer-compatible. Covers both STUN\n * (no credentials) and TURN (username + credential). We keep a single\n * list instead of splitting stun/turn because the downstream werift API\n * takes a flat `iceServers` array and the split-only added ceremony.\n */\nexport interface IceServerEntry {\n  urls: string | string[];\n  username?: string;\n  credential?: string;\n}\n\nexport interface AdaptiveSessionOptions {\n  sessionId: string;\n  /**\n   * Owning device id (parsed from the brokerId `${deviceId}/${camStreamId}`).\n   * Threaded into every session log's `meta` so device-scoped operator\n   * log filters don't drop session lifecycle lines as `dev=?`. The\n   * `logger` passed in is ALSO expected to be device-tagged\n   * (`withTags({ deviceId })`) so the structured tag column is populated;\n   * this field mirrors it into `meta` for the diagnostic logs. `-1` for\n   * the bare-deviceKey shape with no parseable id.\n   */\n  deviceId: number;\n  source: FrameSource;\n  intercom?: {\n    onAudioReceived: (data: Buffer, format: AudioCodec) => void | Promise<void>;\n  };\n  iceConfig?: {\n    /** Flat RTCIceServer-compatible list (STUN + TURN merged). */\n    iceServers?: readonly IceServerEntry[];\n    portRange?: [number, number];\n    additionalHostAddresses?: readonly string[];\n    /**\n     * Suppress IPv6 host gathering and filter IPv6 entries out of\n     * `additionalHostAddresses`. Default false. Used for peers that\n     * document IPv4-only ICE (Alexa RTC). See `webrtc-session.cap.ts`.\n     */\n    disableIpv6?: boolean;\n  };\n  /** Callback for RTCP stats (called every ~3s). */\n  onStats?: (stats: SessionStats) => void;\n  /**\n   * Source-RTP pre-buffer accessor (broker's current GOP). On the\n   * repacketizer path, the session replays this on its first forwarded\n   * packet so the viewer's decoder starts immediately from a keyframe\n   * instead of waiting for the camera's next IDR. Returns [] when no\n   * bootstrap is available (non-RTP source, no keyframe yet, cap overflow).\n   */\n  rtpBootstrap?: () => readonly Buffer[];\n  /** Enable verbose frame/RTP logging. */\n  debug?: boolean;\n  /** Source video codec. Determines SDP negotiation + RTP packetization. */\n  sourceCodec?: 'H264' | 'H265';\n  /**\n   * H.264 source whose profile is non-Baseline (Main/High) — the egress\n   * must be re-encoded to Constrained Baseline so iOS (WebKit) can decode\n   * it. Decided by the broker from the camera's SPS. Ignored for H.265\n   * sources (their transcode is driven by codec negotiation). Default false.\n   */\n  transcodeToBaseline?: boolean;\n  /**\n   * EncodeProfile this session uses for its in-process transcode\n   * (when the source codec or client capabilities require it).\n   * Resolved from `WEBRTC_FALLBACK_PROFILE`, smart-adapted to the\n   * probed source resolution (raw fallback when no probed dimensions).\n   */\n  transcodeProfile?: EncodeProfile;\n  /**\n   * Decode `-hwaccel` backend for the in-process transcode feed (e.g.\n   * `'videotoolbox'`). `null`/absent ⇒ software decode. Probe-gated upstream;\n   * the feed falls back to software if hw decode produces no output.\n   */\n  transcodeDecodeHwAccel?: string | null;\n  logger: Logger;\n}\n\nexport interface SessionStats {\n  sessionId: string;\n  /** Fraction of packets lost (0.0–1.0). */\n  packetLoss: number;\n  /** Interarrival jitter in ms. */\n  jitterMs: number;\n  /** Round-trip time in ms (from RTCP SR/RR). */\n  rttMs: number;\n  /** Total packets received. */\n  packetsReceived: number;\n  /** Total packets lost. */\n  packetsLost: number;\n  /** Timestamp. */\n  timestamp: number;\n}\n\nexport interface SessionInfo {\n  sessionId: string;\n  state: \"new\" | \"connecting\" | \"connected\" | \"disconnected\" | \"closed\";\n  createdAt: number;\n}\n\n// ---------------------------------------------------------------------------\n// AdaptiveSession\n// ---------------------------------------------------------------------------\n\nexport class AdaptiveSession {\n  private readonly sessionId: string;\n  private readonly deviceId: number;\n  private source: FrameSource;\n  private readonly logger: Logger;\n  private readonly intercom: AdaptiveSessionOptions[\"intercom\"];\n  private readonly iceConfig: AdaptiveSessionOptions[\"iceConfig\"];\n  /** Force TURN-relay-only ICE — set for client-offer (Alexa) sessions whose\n   *  peer is a cloud media server unreachable via host/srflx behind NAT. */\n  private forceRelayOnly = false;\n  /** Latest RTCP Receiver Report from the remote viewer (jitter/loss),\n   *  captured via sender.onRtcp. null until the first RR arrives. */\n  private lastRr: WeriftReceiverReport | null = null;\n  /** Monotonic base (ms, performance.now) for send-time RTP timestamps. */\n  private videoRtpClockBaseMs: number | null = null;\n  /** Log the camera's actual H.264 profile-level-id once per session. */\n  private profileLogged = false;\n  private readonly onStats: AdaptiveSessionOptions[\"onStats\"];\n  debug: boolean;\n  private readonly sourceCodec: 'H264' | 'H265';\n  /**\n   * H.264 source profile is non-Baseline (Main/High) → egress must be\n   * re-encoded to Constrained Baseline for iOS. Set by the broker from the\n   * camera SPS; ignored for H.265 sources. Default false.\n   */\n  private readonly transcodeToBaseline: boolean;\n  /** EncodeProfile used when startTranscodeFeed spawns ffmpeg. */\n  private readonly transcodeProfile: EncodeProfile;\n  /** Decode hw-accel backend for the transcode feed; null = software decode. */\n  private readonly transcodeDecodeHwAccel: string | null;\n  /** Codec actually negotiated with the browser after SDP answer. */\n  private negotiatedCodec: 'H264' | 'H265' = 'H264';\n  /**\n   * True when the egress can't be the camera's native bytes: H.265 source\n   * with an H.264-only browser, or an H.264 Main/High source that must be\n   * forced to Constrained Baseline for iOS. See `resolveVideoTranscode`.\n   */\n  get needsTranscode(): boolean {\n    return resolveVideoTranscode({\n      sourceCodec: this.sourceCodec,\n      negotiatedCodec: this.negotiatedCodec,\n      transcodeToBaseline: this.transcodeToBaseline,\n    });\n  }\n  private _firstKeyFrame: number | undefined;\n  /**\n   * Last seen SPS and PPS NALs. Many cameras send SPS/PPS only once\n   * at stream start (not inline with every IDR). We cache them so\n   * PLI-triggered keyframe re-sends include the parameter sets the\n   * decoder needs to re-initialise.\n   */\n  private lastSps: Buffer | null = null;\n  private lastPps: Buffer | null = null;\n  /** H.265 VPS (Video Parameter Set) — required before every IRAP for decoder init. */\n  private lastVps: Buffer | null = null;\n  /** Whether replaceRTP has been called on the video sender to sync SSRC/seq. */\n  private videoRtpSynced = false;\n  private readonly createdAt: number;\n\n  private state: SessionInfo[\"state\"] = \"new\";\n  private pc: WeriftPeerConnection | null = null;\n  private videoTrack: WeriftMediaStreamTrack | null = null;\n  private audioTrack: WeriftMediaStreamTrack | null = null;\n  /** Transceiver senders for direct sendRtp (more reliable than track.writeRtp) */\n  private videoSender: WeriftRtpSender | null = null;\n  private audioSender: WeriftRtpSender | null = null;\n  private feedAbort: AbortController | null = null;\n  private closed = false;\n  private statsTimer: ReturnType<typeof setInterval> | null = null;\n\n  /**\n   * Notification hook invoked exactly once when `close()` finishes. The\n   * adaptive server wires this up after `createSession()` so the ICE\n   * disconnect/failed path (see the `iceConnectionStateChange` handler\n   * that calls `this.close()` on its own) reaches `cam.sessions.delete`\n   * + `scheduleCameraAutoStop`. Without this callback the server kept\n   * stale entries in `cam.sessions`, the auto-stop guard (`size > 0`)\n   * never fired, ffmpeg stayed up, and the RTSP client toward the\n   * broker leaked forever (`rtspClients: 2` on idle brokers).\n   */\n  onClosed: (() => void) | null = null;\n\n  /**\n   * Fires once ICE reaches connected; the server reads the nominated pair to\n   * tag the broker subscription with the remote address.\n   */\n  onConnected: (() => void) | null = null;\n\n  /** RTP sequence number counter (must increment per packet). */\n  private videoSeqNum = 0;\n  private audioSeqNum = 0;\n  /**\n   * Last RTP timestamp actually written on each track (RTP ticks at the\n   * track's clock rate — 90 kHz video, 8 kHz PCMU audio). Captured purely\n   * for the debug-only av-sync diagnostic, which converts them to media-time\n   * to expose whether the audio RTP clock runs behind video's and by how much.\n   * null until the first packet is sent on that track. Diagnostic-only — no\n   * send-path behaviour reads these.\n   */\n  private lastVideoRtpTs: number | null = null;\n  private lastAudioRtpTs: number | null = null;\n  /**\n   * First RTP timestamp seen on each track, captured to make the av-sync diag\n   * media-time SESSION-RELATIVE. Video is H.264 PASSTHROUGH, so its RTP ts is\n   * the camera's free-running source clock (a huge arbitrary value, hours of\n   * media); audio is a sample counter starting near 0. Subtracting the per-track\n   * first ts puts both on a common session-zero base so interTrackSkew is\n   * meaningful. null until the first packet on that track. Diagnostic-only.\n   */\n  private firstVideoRtpTs: number | null = null;\n  private firstAudioRtpTs: number | null = null;\n  /**\n   * Wall-clock (Date.now(), ms) at the FIRST packet sent on each track. The gap\n   * `firstAudioSendWallMs − firstVideoSendWallMs` is the constant startup offset\n   * between the two tracks — when video forwards with ~0 latency (passthrough)\n   * but audio is delayed by its decode/transcode pipeline, this gap ≈ the steady\n   * perceived audio-behind-video lag. Per-track media-vs-wall skew then tells us\n   * if it's a constant offset (no drift) vs a growing one. Diagnostic-only.\n   */\n  private firstVideoSendWallMs: number | null = null;\n  private firstAudioSendWallMs: number | null = null;\n  /**\n   * Most recent AUDIO source-frame capture timestamp (the incoming frame's\n   * `timestampMicros`, in µs) and the send-time wall-clock (Date.now(), ms)\n   * captured at the instant we wrote that frame's RTP. Their relationship is\n   * the transcode/buffer latency the av-sync diagnostic surfaces — the prime\n   * suspect for a constant audio lag (esp. the AAC→PCMU transcode path).\n   * Diagnostic-only; null until the first audio frame is written.\n   */\n  private lastAudioSrcCaptureMicros: number | null = null;\n  private lastAudioSendWallMs: number | null = null;\n  /**\n   * Cached last keyframe NALs (SPS + PPS + IDR slices) in Annex-B format,\n   * split and ready for RTP packetization. When the browser sends a PLI\n   * (Picture Loss Indication) because it lost reference frames, we\n   * immediately re-send this stored keyframe so the decoder can recover\n   * without waiting for the next natural keyframe from the encoder.\n   */\n  private lastKeyframeNals: Buffer[] | null = null;\n  /** Throttle: minimum interval between PLI-triggered keyframe re-sends (ms). */\n  private static readonly PLI_RESEND_COOLDOWN_MS = 500;\n  private lastPliResendAt = 0;\n\n  /**\n   * Per-session H.265 RTP repacketizer (lazy-init). The H.265 path\n   * forwards source RTP from the broker through this repacketizer so\n   * Chrome's HEVC depacketizer sees an RTP shape it actually accepts —\n   * the prior depacketize → AnnexB → re-packetize-from-scratch path\n   * produced `framesAssembledFromMultiplePackets = 0` regardless of\n   * codec metadata being correct. See `forwardSourceRtpVideo`.\n   */\n  private h265Repacketizer: H265Repacketizer | null = null;\n  /** RTP MTU for the repacketizer. 1100 keeps the full wire packet\n   *  (payload + RTP hdr + hdr-extensions + SRTP/GCM tag + UDP + IPv4/IPv6)\n   *  comfortably under the Tailscale/WireGuard overlay MTU (1280) — at the\n   *  old 1180/1200 a packet is ~1266 B, which fits a plain 1500-MTU LAN but\n   *  is at/over the 1280 overlay limit, so keyframe packets silently drop\n   *  over Tailscale → the keyframe never reassembles → endless PLI storm\n   *  (LAN-OK / remote-fail). 1100 leaves ~90 B of headroom for IPv6.\n   *  Both codecs share the live `repacketizerMtu` (default = the H264 static\n   *  below) so the adaptive ladder can shrink packets for either codec. */\n  /**\n   * Per-session H.264 RTP repacketizer (lazy-init). H.264 RTSP sources\n   * forward source RTP through this — same Tailscale-safe MTU and the same\n   * reason as H.265: iOS (WebKit) rejects the `writeVideoNals` synthesized\n   * RTP shape for Main/High passthrough; the repacketizer preserves the\n   * native RTP layout (STAP-A SPS/PPS + FU-A). See `forwardSourceRtpVideo`.\n   */\n  private h264Repacketizer: H264Repacketizer | null = null;\n  static readonly H264_REPACKETIZER_MTU = 1100;\n  /** Reduced RTP payload MTU engaged by the adaptive ladder at the lowest\n   *  tier when loss persists — smaller packets survive lossy links better.\n   *  Must stay below H264_REPACKETIZER_MTU (1100). */\n  static readonly ADAPTIVE_LOW_MTU = 900;\n  /** Broker source-RTP pre-buffer accessor + one-shot replay guard. */\n  private readonly rtpBootstrap?: () => readonly Buffer[];\n  private rtpBootstrapDone = false;\n  /**\n   * Trickle ICE: the server's locally-gathered candidates, buffered as werift\n   * surfaces them via `onIceCandidate`. The client polls `getIceCandidates`\n   * and adds each. Lets `handleOffer` return the answer IMMEDIATELY (no\n   * gathering wait) — candidates flow afterwards → ~0s connect.\n   */\n  private readonly localIceCandidates: IceCandidatePayload[] = [];\n  private iceGatheringComplete = false;\n  /** Source SSRC — captured on first source RTP, used to populate the\n   *  outbound packets so werift's send pipeline doesn't reject them. */\n  private sourceVideoSsrc: number | null = null;\n\n  /** Set by the broker server when a tier change needs a client re-offer.\n   *  Polled via the session state. null when no renegotiation is pending. */\n  pendingRenegotiation: { target: WebrtcStreamTarget; epoch: number } | null = null;\n\n  /** Sink for RTCP Receiver Report loss samples (adaptive controller). */\n  private rrSink: ((s: { packetsLost: number; highestSequence: number }) => void) | null = null;\n\n  /** Register a sink for RTCP Receiver Report loss samples (adaptive controller). */\n  onReceiverReport(cb: (s: { packetsLost: number; highestSequence: number }) => void): void {\n    this.rrSink = cb;\n  }\n\n  /** Live repacketizer MTU; read when (re)building the repacketizer so a\n   *  mid-session change applies on the next rebuild. Defaults to the static\n   *  H264 MTU so current behaviour is identical until an override is set. */\n  private repacketizerMtu: number = AdaptiveSession.H264_REPACKETIZER_MTU;\n\n  /** Override the repacketizer MTU at runtime (adaptive ladder lowest tier). */\n  setRepacketizerMtu(mtu: number): void {\n    this.repacketizerMtu = mtu;\n  }\n\n  /** Drop the repacketizer + bootstrap/sync state so a newly bound source\n   *  stream re-seeds its SPS/PPS and replays instant-start cleanly. */\n  resetRepacketizer(): void {\n    this.h264Repacketizer = null;\n    this.h265Repacketizer = null;\n    this.videoRtpSynced = false;\n    this.rtpBootstrapDone = false;\n    this.sourceVideoSsrc = null;\n  }\n\n  constructor(options: AdaptiveSessionOptions) {\n    this.sessionId = options.sessionId;\n    this.deviceId = options.deviceId;\n    this.source = options.source;\n    this.logger = options.logger;\n    this.intercom = options.intercom;\n    this.iceConfig = options.iceConfig;\n    this.onStats = options.onStats;\n    this.debug = options.debug ?? false;\n    this.sourceCodec = options.sourceCodec ?? 'H264';\n    this.transcodeToBaseline = options.transcodeToBaseline ?? false;\n    this.transcodeProfile = options.transcodeProfile ?? WEBRTC_FALLBACK_PROFILE;\n    this.transcodeDecodeHwAccel = options.transcodeDecodeHwAccel ?? null;\n    this.rtpBootstrap = options.rtpBootstrap;\n    this.createdAt = Date.now();\n  }\n\n  /**\n   * Force TURN-relay-only ICE for this session. MUST be called before\n   * `createOffer()` (which builds the PeerConnection via\n   * `buildPcOptions`) — once the PC exists the policy is fixed.\n   *\n   * Set by the broker for remote (non-LAN) viewers: with the patched\n   * werift, `iceTransportPolicy:'relay'` produces a genuinely relay-only\n   * SDP, so a CGNAT client (which can only offer a relay candidate) gets\n   * a clean relay↔relay media path instead of werift nominating a dead\n   * host/hairpin-srflx pair. The Alexa/WHEP `handleOffer` path sets this\n   * internally; this setter is the server-offer (browser live-view) path.\n   */\n  setForceRelayOnly(value: boolean): void {\n    this.forceRelayOnly = value;\n  }\n\n  /**\n   * Wait for ICE gathering to yield a usable candidate, then return — do NOT\n   * block on full gathering \"complete\". The direct LAN/Tailscale path only\n   * needs a host candidate (present in <1s); waiting for STUN/TURN gathering\n   * to finish cost many seconds of dead startup latency before the offer/\n   * answer could be sent. srflx/relay keep gathering into the local SDP; we\n   * just don't wait on them. A small floor (600ms) lets a fast srflx also\n   * land in the SDP. Hard cap (2.5s) bounds a stalled agent — e.g. a\n   * remote-only peer that genuinely needs a slow relay.\n   */\n  private async waitForIceGatheringFast(): Promise<void> {\n    if (!this.pc) return;\n    if (this.pc.iceGatheringState === \"complete\") return;\n    await new Promise<void>((resolve) => {\n      let settled = false;\n      let poll: ReturnType<typeof setInterval> | undefined;\n      const finish = (): void => {\n        if (settled) return;\n        settled = true;\n        if (poll) clearInterval(poll);\n        resolve();\n      };\n      this.pc?.iceGatheringStateChange.subscribe((state: string) => {\n        if (state === \"complete\") finish();\n      });\n      const start = Date.now();\n      poll = setInterval(() => {\n        const sdp = this.pc?.localDescription?.sdp ?? \"\";\n        if (/ typ host/.test(sdp) && Date.now() - start >= 600) finish();\n      }, 100);\n      setTimeout(finish, 2500);\n    });\n  }\n\n  /**\n   * Wait for ICE gathering to COMPLETE (not just yield a host candidate).\n   * Used by the NON-TRICKLE answer path (Alexa): unlike the browser path,\n   * a non-trickle peer never polls `getIceCandidates` nor trickles back, so\n   * the answer SDP must already carry the server's candidates. The Alexa\n   * path is `relayOnly` — there are NO host candidates, so the host-floor\n   * heuristic in `waitForIceGatheringFast` would never fire; we wait on the\n   * gathering-complete signal instead. werift fires `onIceCandidate(undefined)`\n   * on completion (which sets `iceGatheringComplete`), and also emits\n   * `iceGatheringStateChange('complete')`. A hard cap (`capMs`, default 5s)\n   * bounds a stalled relay so we never block forever — the answer is still\n   * returned with whatever candidates landed.\n   */\n  private async waitForIceGatheringComplete(capMs = 5000): Promise<void> {\n    if (!this.pc) return;\n    if (this.iceGatheringComplete || this.pc.iceGatheringState === \"complete\") return;\n    await new Promise<void>((resolve) => {\n      let settled = false;\n      let poll: ReturnType<typeof setInterval> | undefined;\n      const finish = (): void => {\n        if (settled) return;\n        settled = true;\n        if (poll) clearInterval(poll);\n        resolve();\n      };\n      this.pc?.iceGatheringStateChange.subscribe((state: string) => {\n        if (state === \"complete\") finish();\n      });\n      // The `onIceCandidate(undefined)` handler set up in `handleOffer` flips\n      // `iceGatheringComplete`; poll that flag so completion is observed even\n      // if the state-change event is missed.\n      poll = setInterval(() => {\n        if (this.iceGatheringComplete || this.pc?.iceGatheringState === \"complete\") finish();\n      }, 100);\n      setTimeout(finish, capMs);\n    });\n  }\n\n  /**\n   * Read the nominated (selected) ICE candidate pair off the live\n   * werift connection, via the video sender's DTLS → ICE transport.\n   * Returns null until ICE has nominated a pair (or if werift hasn't\n   * wired the transport chain yet). Used by the media diagnostic to\n   * confirm — on a remote retest — whether werift converged on a\n   * relay↔relay pair.\n   */\n  public readNominatedPair(): {\n    localType: string; localAddr: string;\n    remoteType: string; remoteAddr: string;\n  } | null {\n    const conn = this.videoSender?.dtlsTransport?.iceTransport?.connection;\n    const pair = conn?.nominated;\n    if (!pair) return null;\n    const lc = pair.localCandidate;\n    const rc = pair.remoteCandidate;\n    return {\n      localType: lc.type,\n      localAddr: `${lc.host}:${lc.port}`,\n      remoteType: rc.type,\n      remoteAddr: `${rc.host}:${rc.port}`,\n    };\n  }\n\n  /**\n   * Log the nominated ICE candidate pair once ICE connects. werift may\n   * populate `connection.nominated` a tick AFTER the \"connected\" event\n   * fires, so retry briefly before giving up. This is the decisive\n   * one-shot line for a remote retest: a `relay/…` ↔ `relay/…` pair\n   * means the relay-only fix took effect; anything else (or null) on a\n   * remote session is the dead-pair symptom.\n   */\n  private logNominatedPair(attempt = 0): void {\n    if (this.closed) return;\n    const pair = this.readNominatedPair();\n    if (!pair) {\n      if (attempt < 5) {\n        setTimeout(() => this.logNominatedPair(attempt + 1), 200);\n        return;\n      }\n      this.logger.info('ICE nominated pair', {\n        meta: {\n          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n          forceRelayOnly: this.forceRelayOnly, nominated: 'none',\n        },\n      });\n      return;\n    }\n    this.logger.info('ICE nominated pair', {\n      meta: {\n        phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n        forceRelayOnly: this.forceRelayOnly,\n        selectedLocalType: pair.localType,\n        selectedLocalAddr: pair.localAddr,\n        selectedRemoteType: pair.remoteType,\n        selectedRemoteAddr: pair.remoteAddr,\n      },\n    });\n  }\n\n  /** Build PeerConnection options including H.264 codec config. */\n  private async buildPcOptions(): Promise<{ werift: WeriftModule; pcOptions: WeriftPcOptions }> {\n    const werift = await loadWerift();\n\n    // See `shared-session.ts` — werift's WeriftIceServer carries one URL\n    // per entry, so array-shaped `entry.urls` expands into N entries that\n    // share the same credentials.\n    // ICE servers — STUN/TURN. werift's `setLocalDescription` blocks on\n    // `gatherCandidates()` until every STUN/TURN server responds or hits the\n    // per-server gather timeout — which the werift patch caps short (see\n    // patches/werift+0.22.9.patch) so a slow/unreachable server can't stall\n    // startup. We keep the full set so a mobile viewer (iOS Safari does NOT\n    // offer a Tailscale candidate → no direct path) can use a TURN relay.\n    const iceServers: WeriftIceServer[] = [];\n    for (const entry of this.iceConfig?.iceServers ?? []) {\n      const urlList = Array.isArray(entry.urls) ? entry.urls : [entry.urls];\n      for (const url of urlList) {\n        iceServers.push({\n          urls: url,\n          ...(entry.username !== undefined ? { username: entry.username } : {}),\n          ...(entry.credential !== undefined ? { credential: entry.credential } : {}),\n        });\n      }\n    }\n    // NOTE: we no longer add extra public STUN servers (stun.cloudflare /\n    // stun.l.google). From this hub they timed out (no srflx was ever\n    // produced) and — because werift only finalises the answer SDP when ICE\n    // gathering COMPLETES (peerConnection.js sets the candidates on the\n    // end-of-gathering signal, never incrementally) — every dead/slow server\n    // delays the answer. The turn-provider cap already supplies the servers\n    // we want (Cloudflare STUN+TURN); keeping the answer gated only on fast\n    // servers is what lets gathering complete (and the answer return) quickly.\n\n    const rtcpFeedback = [\n      { type: \"transport-cc\" },\n      { type: \"ccm\", parameter: \"fir\" },\n      { type: \"nack\" },\n      { type: \"nack\", parameter: \"pli\" },\n      { type: \"goog-remb\" },\n    ];\n\n    // Offer both codecs when source is H.265: browser picks H.265 if\n    // supported, falls back to H.264. When source is H.264, offer only H.264.\n    // After SDP negotiation, `negotiatedCodec` tells us which one won.\n    const h264Codec = new werift.RTCRtpCodecParameters({\n      mimeType: \"video/H264\",\n      clockRate: 90000,\n      payloadType: 96,\n      // Constrained Baseline 3.1. iOS Safari WebRTC ONLY negotiates Constrained\n      // Baseline for H.264 RECEIVE — offering Main (4d0033) or High (640034)\n      // makes werift's handleAnswer throw \"negotiate codecs failed\" because iOS\n      // answers Baseline and werift can't reconcile. So Baseline is the only\n      // value that negotiates with iOS; the catch is iOS then decodes ONLY\n      // Baseline, so the CAMERA must emit Baseline for the picture to render\n      // (set the camera/substream to a low/baseline profile). Desktop Chrome\n      // decodes any profile regardless. For a Main/High camera that can't be\n      // set to Baseline, the only werift-passthrough route to iOS is H.265\n      // (iOS negotiates + HW-decodes HEVC natively); H.264 needs transcoding.\n      parameters: \"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\",\n      rtcpFeedback,\n    });\n    // H.265 fmtp — using the H.264-shaped fmtp string for H.265 works\n    // in practice: Chrome rewrites the fmtp to proper H.265 syntax in\n    // the answer regardless. Switching the offer to RFC 7798 syntax did\n    // NOT change `framesDecoded=0` so the offer fmtp isn't the dominant\n    // factor — keep this known-good fmtp value.\n    const h265Codec = new werift.RTCRtpCodecParameters({\n      mimeType: \"video/H265\",\n      clockRate: 90000,\n      payloadType: 97,\n      parameters: 'level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f',\n      rtcpFeedback,\n    });\n\n    const videoCodecs = this.sourceCodec === 'H265'\n      ? [h265Codec, h264Codec]   // prefer H.265, fallback H.264\n      : [h264Codec];              // H.264 only\n\n    const pcOptions: WeriftPcOptions = {\n      codecs: {\n        video: videoCodecs,\n        audio: [\n          new werift.RTCRtpCodecParameters({\n            mimeType: \"audio/PCMU\",\n            clockRate: 8000,\n            payloadType: 0,\n            channels: 1,\n            parameters: \"\",\n          }),\n        ],\n      },\n      // RTP header extensions — KEEP `sdes:mid` only (required for BUNDLE\n      // demux; without it browsers can't route incoming RTP to the right\n      // m-line). DROP `transport-wide-cc` + `abs-send-time`: werift emits\n      // unreliable timing values on those, which a Chrome receiver feeds into\n      // its jitter buffer. On a ~zero-jitter LAN the bad timing is harmless,\n      // but over a real remote path (Tailscale/4G jitter+RTT) the jitter\n      // buffer discards frames as \"too late\" → never decodes → PLI storm,\n      // even though every packet arrives (browser sends RR, sends NO NACK).\n      // Dropping them lets Chrome time playout off its own receive clock.\n      // (Distinct from the earlier mistaken full-disable that also removed\n      // sdes:mid and regressed LAN demux.)\n      headerExtensions: {\n        video: [\n          { uri: \"urn:ietf:params:rtp-hdrext:sdes:mid\" },\n        ],\n        audio: [\n          { uri: \"urn:ietf:params:rtp-hdrext:sdes:mid\" },\n        ],\n      },\n    };\n    if (iceServers.length > 0) pcOptions.iceServers = iceServers;\n    if (this.forceRelayOnly) pcOptions.iceTransportPolicy = \"relay\";\n    // Amazon's Alexa.RTCSessionController spec: \"For ICE candidates, you can\n    // use either UDP or TCP but you must use IPv4.\" Echo's media gateway\n    // silently ignores IPv6 candidates AND counts them against the\n    // documented 6-second SDP processing budget — so emitting our 4 native\n    // GUA + 1 Tailscale fd7a:: candidates wastes the gateway's budget and\n    // can starve the host/srflx pair of validation time. With this set,\n    // werift skips IPv6 host gathering entirely (see\n    // node_modules/werift/lib/ice/src/iceBase.d.ts iceUseIpv6).\n    if (this.iceConfig?.disableIpv6) pcOptions.iceUseIpv6 = false;\n    // Per-candidate-pair logging is DEBUG-only — it floods (dozens of pairs\n    // per session). The nominated pair (the decisive one) is logged at info\n    // unconditionally via `logNominatedPair`. Deduped to avoid spam.\n    if (this.debug) {\n      const loggedPairs = new Set<string>();\n      pcOptions.iceFilterCandidatePair = (pair) => {\n        try {\n          const lc = pair.localCandidate;\n          const rc = pair.remoteCandidate;\n          const key = `${lc?.type}:${lc?.host}:${lc?.port}|${rc?.type}:${rc?.host}:${rc?.port}`;\n          if (!loggedPairs.has(key)) {\n            loggedPairs.add(key);\n            this.logger.info('ICE candidate pair', {\n              meta: {\n                phase: 'session',\n                sessionId: this.sessionId,\n                local: `${lc?.type ?? '?'}/${lc?.host ?? '?'}:${lc?.port ?? '?'}`,\n                remote: `${rc?.type ?? '?'}/${rc?.host ?? '?'}:${rc?.port ?? '?'}`,\n                proto: pair?.protocol?.type,\n              },\n            });\n          }\n        } catch {\n          // never break ICE on a logging error\n        }\n        return true;\n      };\n    }\n    if (this.iceConfig?.portRange) pcOptions.icePortRange = this.iceConfig.portRange;\n\n    // Merge any explicitly-configured additional host addresses with the\n    // dynamically-detected Tailscale overlay address(es). werift skips the\n    // point-to-point utun interface in its own gathering, so without this\n    // the broker never offers a Tailscale-reachable host candidate and a\n    // remote Tailscale viewer is stuck on the (broken) relay path. With it,\n    // both peers sit on the 100.x / fd7a:: overlay and a direct host↔host\n    // pair wins ICE — native H.264, no re-encode. De-dupe; never clobber.\n    const tailscaleHosts = getTailscaleHostAddresses();\n    let mergedHostAddrs = [\n      ...new Set<string>([\n        ...(this.iceConfig?.additionalHostAddresses ?? []),\n        ...tailscaleHosts,\n      ]),\n    ];\n    // Drop IPv6 addresses (anything containing a colon) when the caller\n    // banned IPv6 — Tailscale exposes both an IPv4 100.x and an IPv6\n    // fd7a:: overlay address; only the IPv4 one is useful for an\n    // IPv4-only peer like Alexa.\n    if (this.iceConfig?.disableIpv6) {\n      mergedHostAddrs = mergedHostAddrs.filter((addr) => !addr.includes(':'));\n    }\n    if (mergedHostAddrs.length > 0) {\n      pcOptions.iceAdditionalHostAddresses = mergedHostAddrs;\n      if (tailscaleHosts.length > 0) {\n        this.logger.info(\"offering Tailscale host candidate(s)\", {\n          meta: {\n            phase: \"session\",\n            sessionId: this.sessionId,\n            tailscaleHosts: tailscaleHosts.join(\",\"),\n            additionalHostAddresses: mergedHostAddrs.join(\",\"),\n          },\n        });\n      }\n    }\n\n    return { werift, pcOptions };\n  }\n\n  /** Create offer SDP (server → client). */\n  async createOffer(): Promise<{ sdp: string; type: \"offer\" }> {\n    const { werift, pcOptions } = await this.buildPcOptions();\n\n    this.pc = new werift.RTCPeerConnection(pcOptions);\n\n    // ICE state monitoring — promoted to info so the full lifecycle\n    // (new → checking → connected/failed) is visible without flipping a\n    // debug flag. Sessions stuck at \"checking\" are the canonical symptom\n    // of ICE candidate mismatch; hiding the transitions behind `debug`\n    // meant we couldn't tell \"stuck checking\" from \"never even started\".\n    this.pc.iceConnectionStateChange.subscribe((state: string) => {\n      this.logger.info('ICE state changed', { meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, state, forceRelayOnly: this.forceRelayOnly } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\n        this.logNominatedPair();\n        // Best-effort: the hook reads the nominated pair to tag the broker\n        // subscription. A throwing hook must never break the ICE path.\n        try { this.onConnected?.(); } catch { /* swallow — the session must not care */ }\n        this.startStatsCollection();\n      } else if (state === \"disconnected\" || state === \"failed\" || state === \"closed\") {\n        this.state = state === \"disconnected\" ? \"disconnected\" : \"closed\";\n        void this.close();\n      }\n    });\n\n    // Gathering state — surfaces whether the server is actually producing\n    // ICE candidates and how long it takes. \"gathering\" without a\n    // subsequent \"complete\" points at a local-network / binding issue.\n    this.pc.iceGatheringStateChange.subscribe((state: string) => {\n      this.logger.info('ICE gathering state changed', { meta: { phase: 'session', sessionId: this.sessionId, state } });\n    });\n\n    // Video track (sendonly) — save transceiver sender for sendRtp\n    this.videoTrack = new werift.MediaStreamTrack({ kind: \"video\" });\n    const videoTransceiver = this.pc.addTransceiver(this.videoTrack, { direction: \"sendonly\" });\n    this.videoSender = videoTransceiver.sender;\n\n    // PLI/FIR listener — when the browser reports picture loss (decoder\n    // can't decode P-frames because it missed the reference), re-send\n    // the last cached keyframe immediately. This is the fix for the\n    // \"stale frame with diffs\" symptom: without it, the viewer is stuck\n    // on an old I-frame until the encoder naturally emits the next one.\n    this.setupPliListener();\n\n    // Audio track (sendrecv if intercom, sendonly otherwise)\n    this.audioTrack = new werift.MediaStreamTrack({ kind: \"audio\" });\n    const audioDir = this.intercom ? \"sendrecv\" : \"sendonly\";\n    const audioTransceiver = this.pc.addTransceiver(this.audioTrack, { direction: audioDir });\n    this.audioSender = audioTransceiver.sender;\n\n    // Intercom: listen for incoming audio\n    if (this.intercom) {\n      const cb = this.intercom.onAudioReceived;\n      audioTransceiver.onTrack.subscribe((track) => {\n        track.onReceiveRtp.subscribe((pkt) => {\n          try {\n            const payload = pkt.payload;\n            if (payload?.length > 0) void cb(payload, \"Opus\");\n          } catch (err) {\n            this.logger.error('Intercom error', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) } });\n          }\n        });\n      });\n    }\n\n    const offer = await this.pc.createOffer();\n    await this.pc.setLocalDescription(offer);\n\n    // Include candidates in the offer, but don't block on full gathering —\n    // resolve as soon as a host candidate is present (see\n    // `waitForIceGatheringFast`). Cuts startup from ~5-10s to <1s on the\n    // LAN/Tailscale direct path.\n    await this.waitForIceGatheringFast();\n\n    // Use the local description which now includes gathered ICE candidates.\n    // Force a=setup:actpass (let browser choose role) — werift defaults may cause DTLS issues.\n    let finalSdp = this.pc.localDescription?.sdp ?? offer.sdp;\n    // Ensure actpass so browser can be either active or passive\n    finalSdp = finalSdp.replace(/a=setup:active\\r?\\n/g, \"a=setup:actpass\\r\\n\");\n    this.state = \"connecting\";\n\n    // Surface the ICE candidates the server ended up advertising. Printed\n    // at info level because this is the first thing to inspect when a\n    // session gets stuck at \"checking\" — if we only see a single\n    // `127.0.0.1 host` line, the browser may well be unable to pair from\n    // anywhere except a same-machine loopback, and anonymised mDNS\n    // candidates from Chrome won't bind.\n    const gatheredCandidates = finalSdp\n      .split('\\n')\n      .map(l => l.trim())\n      .filter(l => l.startsWith('a=candidate:'))\n    this.logger.info('Offer gathered ICE candidates', {\n      meta: { phase: 'session', sessionId: this.sessionId, count: gatheredCandidates.length, candidates: gatheredCandidates },\n    })\n\n    // Log SDP video codec lines for debugging\n    const videoCodecLines = finalSdp.split('\\n').filter((l: string) => l.includes('H264') || l.includes('rtpmap:96') || l.includes('fmtp:96') || l.includes('profile-level-id')).map((l: string) => l.trim());\n    if (this.debug) this.logger.info('Offer created — video SDP', {\n      meta: { phase: 'session', sessionId: this.sessionId, videoCodecLines },\n    });\n    return { sdp: finalSdp, type: \"offer\" };\n  }\n\n  /** Handle WHEP answer: client sends SDP answer, we set remote description and start feeding. */\n  async handleAnswer(answer: { sdp: string; type: \"answer\" }): Promise<void> {\n    if (!this.pc) throw new Error(\"Call createOffer() first\");\n    const werift = await loadWerift();\n\n    // Surface the browser's ICE candidates BEFORE setRemoteDescription so\n    // we can see them even if werift rejects one of them. Chrome's mDNS\n    // privacy layer emits candidates like `abcdef.local` on loopback; a\n    // Node.js ICE agent without mDNS resolution can't pair them, and the\n    // session stays at \"checking\" forever. If the only candidates shown\n    // here are `*.local`, that's the smoking gun.\n    const remoteCandidates = answer.sdp\n      .split('\\n')\n      .map(l => l.trim())\n      .filter(l => l.startsWith('a=candidate:'))\n    this.logger.info('Answer received ICE candidates', {\n      meta: { phase: 'session', sessionId: this.sessionId, count: remoteCandidates.length, candidates: remoteCandidates },\n    })\n\n    // Rewrite Chrome-anonymised `*.local` candidates into their resolved IPs\n    // before werift sees them; werift's ICE agent doesn't run mDNS itself\n    // and would otherwise stall at \"checking\" forever.\n    const resolvedSdp = await resolveMdnsCandidatesInSdp(answer.sdp, this.logger, `session:${this.sessionId}`)\n    const desc = new werift.RTCSessionDescription(resolvedSdp, answer.type);\n    await this.pc.setRemoteDescription(desc);\n\n    // Detect which video codec the browser selected from the answer SDP.\n    // If source is H.265 but browser only supports H.264, needsTranscode\n    // will be true and the feed loop will spawn ffmpeg. Thread `sourceCodec`\n    // as the preferred hint so an answer SDP that lists H.264 ahead of H.265\n    // doesn't wrongly downgrade an H.265-capable session to H.264 (see\n    // detectNegotiatedVideoCodec docstring).\n    const answerVideoCodec = detectNegotiatedVideoCodec(resolvedSdp, this.sourceCodec)\n    if (answerVideoCodec) {\n      this.negotiatedCodec = answerVideoCodec\n    }\n    this.logger.info('Codec negotiated', {\n      meta: { phase: 'session', sessionId: this.sessionId, source: this.sourceCodec, negotiated: this.negotiatedCodec, needsTranscode: this.needsTranscode },\n    })\n\n    if (this.debug) {\n      // Offered (our offer) vs negotiated (browser answer) profile-level-ids —\n      // the decisive evidence for an iOS H.264 passthrough that \"negotiates but\n      // shows black\" because the camera emits a higher profile than the\n      // browser will decode. See the iOS H.264 passthrough notes.\n      this.logger.info('Codec negotiation detail', {\n        meta: {\n          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n          offeredFmtp: extractFmtpLines(this.pc.localDescription?.sdp ?? ''),\n          negotiatedFmtp: extractFmtpLines(resolvedSdp),\n        },\n      })\n    }\n\n    // Wait for DTLS to connect before feeding — sendRtp silently drops if DTLS not ready\n    const dtlsTransport = this.videoSender?.dtlsTransport;\n    if (dtlsTransport && dtlsTransport.state !== \"connected\") {\n      this.logger.debug('Waiting for DTLS...', {\n        meta: { phase: 'session', sessionId: this.sessionId, current: dtlsTransport.state },\n      });\n      const deadline = Date.now() + 10_000;\n      // Track DTLS state transitions while we poll — werift's typed surface\n      // exposes only `state` (no onStateChange), so the wait loop doubles as\n      // the transition observer. Each change is logged when debug is on.\n      let prevDtls = dtlsTransport.state;\n      while (dtlsTransport.state !== \"connected\" && Date.now() < deadline) {\n        if (this.debug && dtlsTransport.state !== prevDtls) {\n          this.logger.info('DTLS state', {\n            meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, from: prevDtls, to: dtlsTransport.state },\n          });\n          prevDtls = dtlsTransport.state;\n        }\n        await new Promise<void>((r) => setTimeout(r, 100));\n      }\n      if (this.debug && dtlsTransport.state !== prevDtls) {\n        this.logger.info('DTLS state', {\n          meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, from: prevDtls, to: dtlsTransport.state },\n        });\n      }\n      this.logger.debug('DTLS wait complete', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          state: dtlsTransport.state,\n          waitedSec: (10000 - (deadline - Date.now())) / 1000,\n        },\n      });\n    }\n\n    this.logger.debug('Answer set, feeding started', {\n      meta: { phase: 'session', sessionId: this.sessionId, dtls: dtlsTransport?.state ?? 'unknown' },\n    });\n    this.startFeedingFrames();\n  }\n\n  /**\n   * Handle WHEP offer: client sends SDP offer, we create answer.\n   *\n   * Uses the server-creates-offer pattern internally: we create our own offer\n   * with sendonly tracks, then use the client's offer codecs to build a\n   * compatible answer. This avoids werift transceiver direction issues.\n   */\n  async handleOffer(\n    clientOffer: { sdp: string; type: \"offer\" },\n    opts?: { nonTrickle?: boolean },\n  ): Promise<{ sdp: string; type: \"answer\" }> {\n    // `forceRelayOnly` is set by the caller via `setForceRelayOnly` BEFORE\n    // this runs (the broker does it from the cap's server-injected\n    // `relayOnly` flag): true for a cloud peer like Alexa's\n    // RTCSessionController (never reachable via our host/srflx behind NAT),\n    // false for a LAN/Tailscale browser viewer doing client-offer\n    // passthrough (a direct host pair carries full native quality, and\n    // forcing relay would defeat it). No hardcoded assumption here.\n    const { werift, pcOptions } = await this.buildPcOptions();\n\n    this.pc = new werift.RTCPeerConnection(pcOptions);\n\n    // ICE state at INFO (the server-offer path does the same) — for the\n    // client-offer flow Alexa uses, \"stuck at checking\" is the canonical\n    // black-view symptom and we need to see the transition without a debug flag.\n    this.pc.iceConnectionStateChange.subscribe((state: string) => {\n      this.logger.info('ICE state', { meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, state, forceRelayOnly: this.forceRelayOnly } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\n        this.logNominatedPair();\n        // Best-effort: the hook reads the nominated pair to tag the broker\n        // subscription. A throwing hook must never break the ICE path.\n        try { this.onConnected?.(); } catch { /* swallow — the session must not care */ }\n        this.startStatsCollection();\n      } else if (state === \"disconnected\" || state === \"failed\" || state === \"closed\") {\n        this.state = state === \"disconnected\" ? \"disconnected\" : \"closed\";\n        void this.close();\n      }\n    });\n\n    // Trickle ICE: buffer each local candidate werift gathers so the client\n    // can poll it (`getIceCandidates`) AFTER we return the answer. werift\n    // fires `undefined` when gathering completes. This is what lets the answer\n    // go back immediately instead of blocking on full gathering.\n    this.pc.onIceCandidate.subscribe((c: WeriftIceCandidate | undefined) => {\n      if (!c || !c.candidate) { this.iceGatheringComplete = true; return; }\n      this.localIceCandidates.push({\n        candidate: c.candidate,\n        sdpMid: c.sdpMid ?? null,\n        sdpMLineIndex: c.sdpMLineIndex ?? null,\n      });\n    });\n\n    // Set the client's offer as remote description. mDNS-rewriting the\n    // offer for the same reason as `handleAnswer` — Chrome sends\n    // anonymised `*.local` host candidates even in the offer path when\n    // issuing WHEP-style requests.\n    const resolvedOfferSdp = await resolveMdnsCandidatesInSdp(clientOffer.sdp, this.logger, `session:${this.sessionId}`)\n    // Count of REMOTE candidates the peer offered (full list only at debug).\n    const offerCandidates = resolvedOfferSdp\n      .split('\\n')\n      .map((l) => l.trim())\n      .filter((l) => l.startsWith('a=candidate:'))\n    if (this.debug) {\n      this.logger.info('client offer REMOTE ICE candidates', {\n        meta: {\n          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n          count: offerCandidates.length,\n          candidateTypes: summarizeCandidates(resolvedOfferSdp),\n          candidates: offerCandidates,\n        },\n      })\n    }\n    const remoteDesc = new werift.RTCSessionDescription(resolvedOfferSdp, clientOffer.type);\n    await this.pc.setRemoteDescription(remoteDesc);\n\n    // Find the transceivers werift created from the offer and attach our\n    // tracks. CRITICAL: save the transceiver senders — the feed loop pushes\n    // RTP through `this.videoSender`/`this.audioSender`, and without these\n    // assignments every send path bails out with \"writeVideoNals: no sender\"\n    // (the server-offer path in `start()` does the same at line ~333).\n    const transceivers: WeriftTransceiver[] = this.pc.getTransceivers();\n    for (const t of transceivers) {\n      const kind = t.receiver?.track?.kind ?? t.kind;\n      if (kind === \"video\" && !this.videoTrack) {\n        // CRITICAL: force `sendonly`. The browser offers each transceiver\n        // `recvonly` (it only wants to receive). Without this, werift answers\n        // `inactive` and marks the m-line rejected (port 0), which empties\n        // the `a=group:BUNDLE` line → Chrome/iOS rejects the whole answer with\n        // \"A BUNDLE group contains a MID='' matching no m= section\" and the\n        // session never connects. The server-offer path adds its transceivers\n        // with `{direction:'sendonly'}` for the same reason.\n        t.setDirection(\"sendonly\");\n        this.videoTrack = new werift.MediaStreamTrack({ kind: \"video\" });\n        await t.sender.replaceTrack(this.videoTrack);\n        this.videoSender = t.sender;\n        // When the source is H.265, override the sender's codec with the\n        // H.265 entry from the transceiver's negotiated codec list.\n        // Without this, werift's createAnswer settles `sender.codec` on\n        // the FIRST codec the browser offered — and browsers list 10+\n        // H.264 PTs before the 2 H.265 PTs in their offer order, so the\n        // sender ends up tagged H.264 even though our pcOptions preferred\n        // H.265 and the answer SDP advertised both. The outgoing RTP\n        // packets then carry PT=H.264 in the header but payload=H.265\n        // NAL units; Chrome's depacketizer never initialises an HEVC\n        // decoder, and `hasReceiverReport` stays false even as packets\n        // keep flowing. Reassigning here before `createAnswer` propagates\n        // the right PT through werift's sender path. Symmetrical for\n        // H.264 sources is not required: pcOptions only advertises\n        // H.264 in that case, so there is no PT ambiguity.\n        if (this.sourceCodec === 'H265') {\n          const h265 = t.codecs?.find((c) => c.mimeType === 'video/H265')\n          if (h265) {\n            t.sender.codec = h265\n            this.logger.info('forced H.265 codec on video sender', {\n              meta: {\n                phase: 'session',\n                sessionId: this.sessionId,\n                payloadType: h265.payloadType,\n              },\n            })\n          } else {\n            this.logger.warn('sourceCodec=H265 but no H.265 codec in transceiver', {\n              meta: {\n                phase: 'session',\n                sessionId: this.sessionId,\n                availableMimeTypes: t.codecs?.map((c) => c.mimeType) ?? [],\n              },\n            })\n          }\n        }\n      } else if (kind === \"audio\" && !this.audioTrack) {\n        t.setDirection(\"sendonly\");\n        this.audioTrack = new werift.MediaStreamTrack({ kind: \"audio\" });\n        await t.sender.replaceTrack(this.audioTrack);\n        this.audioSender = t.sender;\n      }\n    }\n\n    // Fallback: if no transceivers matched (shouldn't happen with valid offer)\n    if (!this.videoTrack) {\n      this.logger.warn('No video transceiver found in offer, adding one', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n      this.videoTrack = new werift.MediaStreamTrack({ kind: \"video\" });\n      const videoTransceiver = this.pc.addTransceiver(this.videoTrack, { direction: \"sendonly\" });\n      this.videoSender = videoTransceiver.sender;\n    }\n    if (!this.audioTrack) {\n      this.logger.warn('No audio transceiver found in offer, adding one', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n      this.audioTrack = new werift.MediaStreamTrack({ kind: \"audio\" });\n      const audioTransceiver = this.pc.addTransceiver(this.audioTrack, { direction: \"sendonly\" });\n      this.audioSender = audioTransceiver.sender;\n    }\n\n    const answerDesc = await this.pc.createAnswer();\n    let finalSdp: string;\n    if (opts?.nonTrickle === true) {\n      // NON-TRICKLE ICE (Alexa): the peer never polls `getIceCandidates` nor\n      // trickles back, so the answer SDP must already carry the server's\n      // candidates. Await `setLocalDescription` then WAIT for ICE gathering to\n      // COMPLETE, then read the populated `localDescription.sdp` (now with\n      // `a=candidate` lines). The Alexa path is relayOnly (no host candidates),\n      // so we wait on gathering-complete — bounded to ~5s — rather than the\n      // host-floor heuristic of the trickle path; a relay/srflx candidate lands\n      // in the SDP without blocking forever.\n      await this.pc.setLocalDescription(answerDesc).catch((e) => {\n        this.logger.warn('setLocalDescription failed (non-trickle)', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(e) } });\n      });\n      await this.waitForIceGatheringComplete();\n      finalSdp = this.pc?.localDescription?.sdp ?? answerDesc.sdp;\n    } else {\n      // TRICKLE ICE (tempo-0 model): do NOT await setLocalDescription\n      // and do NOT wait for gathering. werift's setLocalDescription resolves only\n      // after FULL ICE gathering completes (~10s with TURN); awaiting it — or\n      // waiting for candidates to land in the answer SDP — is the entire startup\n      // latency. Instead fire it, return the BARE answer (ufrag/pwd/fingerprint/\n      // setup, no candidates) IMMEDIATELY, and let the server's candidates flow to\n      // the client afterwards via `getIceCandidates` (buffered by the\n      // onIceCandidate subscription above). The client likewise trickles its\n      // candidates back via `addIceCandidate`. Connection forms in ~0s.\n      void this.pc.setLocalDescription(answerDesc).catch((e) => {\n        this.logger.warn('setLocalDescription failed', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(e) } });\n      });\n\n      // Bare answer — candidates trickle separately. `localDescription` may not\n      // be populated yet (not awaited), so fall back to the createAnswer SDP.\n      finalSdp = this.pc?.localDescription?.sdp ?? answerDesc.sdp;\n    }\n    this.state = \"connecting\";\n\n    // Detect the codec werift settled on in OUR answer. In the client-offer\n    // flow the browser offers its full decode caps (incl. H.264 High on\n    // iOS 18) and werift echoes the matching codec — so `negotiatedCodec`\n    // must be read from the answer here, exactly as `handleAnswer` reads it\n    // from the browser's answer in the server-offer flow. Without this it\n    // stays at the 'H264' default and `needsTranscode` mis-decides for an\n    // H.265 source. Set BEFORE `startFeedingFrames()` so the feed picks the\n    // right path on the first frame. Thread `sourceCodec` as the preferred\n    // hint: werift orders H.264 ahead of H.265 in the answer (mirroring the\n    // browser offer), so the first-match regex would wrongly choose H.264 for\n    // an H.265-capable browser — see cam 99 freeze diagnosis.\n    const answerVideoCodec = detectNegotiatedVideoCodec(finalSdp, this.sourceCodec)\n    if (answerVideoCodec) {\n      this.negotiatedCodec = answerVideoCodec\n    }\n    this.logger.info('Codec negotiated (client-offer)', {\n      meta: { phase: 'session', sessionId: this.sessionId, source: this.sourceCodec, negotiated: this.negotiatedCodec, needsTranscode: this.needsTranscode },\n    })\n\n    if (this.debug) {\n      // Offered = the client's offer (Alexa/WHEP/browser); negotiated = OUR\n      // answer. Surfaces the profile-level-id the peer asked for vs the one we\n      // echoed back — the iOS/Alexa passthrough discriminator.\n      this.logger.info('Codec negotiation detail (client-offer)', {\n        meta: {\n          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n          offeredFmtp: extractFmtpLines(resolvedOfferSdp),\n          negotiatedFmtp: extractFmtpLines(finalSdp),\n        },\n      })\n    }\n\n    if (this.debug) {\n      // CRITICAL: SHOW the FULL set of LOCAL ICE candidates we put in the\n      // ANSWER — parsed into type (host/srflx/relay) + address — not just the\n      // count. On the Alexa (non-trickle, relay-only) path the answer must\n      // already carry the candidates; if it carries an unreachable\n      // Tailscale/private `host` candidate the peer can't pair with, this is\n      // the line that proves it. Raw lines are kept too for full fidelity.\n      const gatheredCandidates = finalSdp\n        .split('\\n')\n        .map((l) => l.trim())\n        .filter((l) => l.startsWith('a=candidate:'));\n      this.logger.info('WHEP answer LOCAL ICE candidates', {\n        meta: {\n          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n          nonTrickle: opts?.nonTrickle === true, forceRelayOnly: this.forceRelayOnly,\n          candidateCount: gatheredCandidates.length,\n          candidateTypes: summarizeCandidates(finalSdp),\n          candidates: gatheredCandidates,\n        },\n      });\n    }\n\n    // CRITICAL: do NOT feed synchronously here. In the client-offer flow the\n    // answer must be RETURNED before the browser can complete DTLS — so at\n    // this point the transport is never connected yet, and `sendRtp` silently\n    // drops everything until DTLS comes up (see `handleAnswer`'s DTLS wait).\n    // Feeding now blasts the FIRST keyframe into a dead transport; the browser\n    // then has only P-frames, can't initialise its decoder, and shows black\n    // (no loss/PLI on our side because nothing was actually transmitted).\n    // Defer feeding until DTLS connects — the dual of the server-offer path.\n    void this.startFeedingWhenDtlsReady();\n\n    return { sdp: finalSdp, type: \"answer\" };\n  }\n\n  /**\n   * Wait for the video sender's DTLS transport to connect, then start\n   * feeding. Used by the client-offer (`handleOffer`) path, which must\n   * return its SDP answer before the browser performs the DTLS handshake —\n   * so feeding can only safely begin once the transport is up. Mirrors the\n   * inline DTLS wait `handleAnswer` does for the server-offer path. Bounded\n   * (10s) so a never-connecting session doesn't leak the wait forever.\n   */\n  private async startFeedingWhenDtlsReady(): Promise<void> {\n    const dtlsTransport = this.videoSender?.dtlsTransport;\n    if (dtlsTransport && dtlsTransport.state !== \"connected\") {\n      const deadline = Date.now() + 10_000;\n      // Observe DTLS transitions while polling (werift exposes only `state`).\n      let prevDtls = dtlsTransport.state;\n      while (\n        dtlsTransport.state !== \"connected\" &&\n        Date.now() < deadline &&\n        !this.closed\n      ) {\n        if (this.debug && dtlsTransport.state !== prevDtls) {\n          this.logger.info('DTLS state (client-offer)', {\n            meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, from: prevDtls, to: dtlsTransport.state },\n          });\n          prevDtls = dtlsTransport.state;\n        }\n        await new Promise<void>((r) => setTimeout(r, 100));\n      }\n      if (this.debug && dtlsTransport.state !== prevDtls) {\n        this.logger.info('DTLS state (client-offer)', {\n          meta: { phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId, from: prevDtls, to: dtlsTransport.state },\n        });\n      }\n      this.logger.debug('client-offer DTLS wait complete', {\n        meta: { phase: 'session', sessionId: this.sessionId, state: dtlsTransport.state },\n      });\n    }\n    if (this.closed) return;\n    // If DTLS never reached \"connected\" within the deadline, sendRtp would\n    // silently drop every packet and the session would sit forever as a\n    // zombie — the broker keeps reading source RTP, the feed loop keeps\n    // pushing frames in, the transport drops them all, and nothing ever\n    // tears the session down. Observed today: a Hikvision session sat\n    // for 16+ minutes burning CPU on audio frames going nowhere because\n    // Echo's ICE check failed and there was no cleanup. Tear it down so\n    // the broker releases the source subscription too.\n    const dtlsState = this.videoSender?.dtlsTransport?.state;\n    if (dtlsState !== \"connected\") {\n      this.logger.warn('client-offer: DTLS never connected — closing session', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          deviceId: this.deviceId,\n          dtls: dtlsState ?? 'unknown',\n          forceRelayOnly: this.forceRelayOnly,\n        },\n      });\n      void this.close();\n      return;\n    }\n    this.startFeedingFrames();\n  }\n\n  /** Add a remote (client) ICE candidate — trickle ICE client→server. */\n  async addIceCandidate(candidate: unknown): Promise<void> {\n    if (!this.pc) throw new Error(\"Call createOffer() first\");\n    const werift = await loadWerift();\n    await this.pc.addIceCandidate(new werift.RTCIceCandidate(candidate));\n  }\n\n  /**\n   * Snapshot of the server's locally-gathered ICE candidates for trickle\n   * polling (server→client). The client polls this after receiving the bare\n   * answer and adds each candidate; `done` flips true when gathering finishes\n   * so the client can stop polling.\n   */\n  getIceCandidatesSnapshot(): { candidates: IceCandidatePayload[]; done: boolean } {\n    return { candidates: [...this.localIceCandidates], done: this.iceGatheringComplete };\n  }\n\n  /**\n   * Detach the frame source (for connection pooling).\n   * The session stays alive (ICE/DTLS connected) but stops feeding frames.\n   * Call replaceSource() later to reattach a camera.\n   */\n  detachSource(): void {\n    if (this.feedAbort) {\n      this.feedAbort.abort();\n      this.feedAbort = null;\n    }\n    this.logger.debug('Source detached (idle)', { meta: { phase: 'session', sessionId: this.sessionId } });\n  }\n\n  /** Whether the session has an active feed (vs idle/pooled). */\n  get isFeeding(): boolean {\n    return this.feedAbort !== null && !this.feedAbort.signal.aborted;\n  }\n\n  /**\n   * Replace the frame source (for seamless source switching).\n   * The new source will take effect at the next keyframe.\n   */\n  replaceSource(newSource: FrameSource): void {\n    this.source = newSource;\n    // Abort old feed — the feeding loop will restart with new source\n    if (this.feedAbort) {\n      this.feedAbort.abort();\n      this.feedAbort = null;\n    }\n    this.startFeedingFrames();\n  }\n\n  getInfo(): SessionInfo {\n    return { sessionId: this.sessionId, state: this.state, createdAt: this.createdAt };\n  }\n\n  async close(): Promise<void> {\n    if (this.closed) return;\n    this.closed = true;\n    this.state = \"closed\";\n    this.logger.info('Closing', { meta: { phase: 'session', sessionId: this.sessionId } });\n\n    if (this.statsTimer) {\n      clearInterval(this.statsTimer);\n      this.statsTimer = null;\n    }\n    if (this.feedAbort) {\n      this.feedAbort.abort();\n      this.feedAbort = null;\n    }\n    try { await this.source.return(undefined); } catch { /* */ }\n    if (this.pc) {\n      try { await this.pc.close(); } catch { /* */ }\n      this.pc = null;\n    }\n    this.videoTrack = null;\n    this.audioTrack = null;\n\n    // Notify the adaptive server so it can drop us from cam.sessions and\n    // trigger scheduleCameraAutoStop. Guarded null-call so the server is\n    // free to not wire it (tests, pooled sessions that manage cleanup\n    // themselves).\n    const hook = this.onClosed;\n    this.onClosed = null;\n    try { hook?.(); } catch { /* server-side bookkeeping shouldn't break close */ }\n  }\n\n  // -----------------------------------------------------------------------\n  // H.265 source-RTP forwarding (packet-level repacketization)\n  // -----------------------------------------------------------------------\n\n  /**\n   * Seed the H.265 repacketizer's codec info with VPS/SPS/PPS harvested\n   * from the camera SDP. Cameras that publish parameter sets only via\n   * SDP (e.g. Reolink high-profile streams) leave the wire bare of\n   * VPS/SPS/PPS — without this seed the repacketizer would never\n   * generate the AP codec packet that initialises the browser decoder.\n   */\n  seedH265CodecInfoFromSdp(parameterSets: ReadonlyArray<Buffer>): void {\n    // DO NOT gate on `this.negotiatedCodec === 'H265'` here. The broker calls\n    // this at session-create time (`broker-webrtc-server.ts:719`), BEFORE\n    // `handleOffer`/`handleAnswer` has had a chance to populate\n    // `negotiatedCodec` from the answer SDP — at that point the field is\n    // still the constructor default `'H264'` and the seed would be silently\n    // dropped. Without the seed, `H265Repacketizer.codecInfo` stays empty\n    // and `maybeSendAPCodecInfo` never emits the VPS+SPS+PPS aggregation\n    // packet that Chrome's HEVC depacketizer needs to initialise → freeze.\n    // The broker already gates the call by `sessionCodec === 'H265'`, so\n    // seeding here is a no-op for non-H.265 sessions.\n    if (!parameterSets.length) return;\n    this.ensureH265Repacketizer();\n    const rep = this.h265Repacketizer!;\n    for (const ps of parameterSets) {\n      const nalType = (ps[0]! & 0x7e) >> 1;\n      if (nalType === 32) rep.updateVps(Buffer.from(ps));\n      else if (nalType === 33) rep.updateSps(Buffer.from(ps));\n      else if (nalType === 34) rep.updatePps(Buffer.from(ps));\n    }\n    if (this.debug) {\n      this.logger.info('seeded H.265 repacketizer codecInfo from SDP', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          psCount: parameterSets.length,\n          hasVps: !!rep.codecInfo?.vps,\n          hasSps: !!rep.codecInfo?.sps,\n          hasPps: !!rep.codecInfo?.pps,\n        },\n      });\n    }\n  }\n\n  private ensureH265Repacketizer(): void {\n    if (this.h265Repacketizer) return;\n    const codecInfo: H265CodecInfo | undefined = undefined;\n    this.h265Repacketizer = new H265Repacketizer(\n      console,\n      this.repacketizerMtu,\n      codecInfo,\n    );\n  }\n\n  /**\n   * Seed the H.264 repacketizer's SPS/PPS from SDP-published parameter sets\n   * so the STAP-A codec packet can precede the first IDR even when the\n   * camera doesn't emit param sets in-band. H.264 sibling of\n   * `seedH265CodecInfoFromSdp`. No-op on non-H.264 sessions.\n   */\n  seedH264CodecInfoFromSdp(parameterSets: ReadonlyArray<Buffer>): void {\n    // Mirror of `seedH265CodecInfoFromSdp`: do NOT gate on negotiatedCodec.\n    // The broker discriminates by `sessionCodec === 'H264'` before calling\n    // (broker-webrtc-server.ts ~720), and seeding the H.264 repacketizer's\n    // SPS/PPS is a no-op for non-H.264 sessions because the repacketizer is\n    // only instantiated when `forwardSourceRtpVideo` runs with `codec='H264'`.\n    if (!parameterSets.length) return;\n    this.ensureH264Repacketizer();\n    const rep = this.h264Repacketizer!;\n    for (const ps of parameterSets) {\n      const nalType = ps[0]! & 0x1f;\n      if (nalType === 7) rep.updateSps(Buffer.from(ps));\n      else if (nalType === 8) rep.updatePps(Buffer.from(ps));\n    }\n    if (this.debug) {\n      this.logger.info('seeded H.264 repacketizer codecInfo from SDP', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          psCount: parameterSets.length,\n          hasSps: !!rep.codecInfo?.sps,\n          hasPps: !!rep.codecInfo?.pps,\n        },\n      });\n    }\n  }\n\n  private ensureH264Repacketizer(): void {\n    if (this.h264Repacketizer) return;\n    this.h264Repacketizer = new H264Repacketizer(\n      console,\n      this.repacketizerMtu,\n    );\n  }\n\n  /**\n   * Forward a source RTP video packet (raw on-wire bytes) through the\n   * codec's repacketizer to the browser. Used by the broker's RTSP\n   * direct-RTP subscription for BOTH H.265 and H.264 — bypasses the\n   * AnnexB→writeVideoNals path entirely so the native RTP layout (which\n   * iOS's strict depacketizer requires) is preserved.\n   *\n   * Drops everything until the SDP answer has been negotiated\n   * (`negotiatedCodec`/`videoSender` populated).\n   */\n  forwardSourceRtpVideo(rtpData: Buffer): void {\n    if (this.closed) return;\n    const codec = this.negotiatedCodec;\n    if (codec !== 'H265' && codec !== 'H264') return;\n    if (!this.videoSender || !_werift) return;\n\n    // Wait for DTLS. `sendRtp` silently drops before the transport is\n    // connected — and `handleOffer` returns its answer (so feeding can begin)\n    // BEFORE the browser performs DTLS. Without this gate the pre-buffer\n    // bootstrap fires the instant `videoSender` exists (mid-negotiation,\n    // pre-ICE), gets entirely dropped, and the viewer still waits a full GOP\n    // for the next live keyframe — the \"instant start\" never lands. Gating\n    // here means the bootstrap replays on the FIRST packet AFTER DTLS, when\n    // the pre-buffer = the current GOP and the packets actually transmit.\n    if (this.videoSender.dtlsTransport?.state !== 'connected') return;\n\n    // Instant start: on the FIRST packet forwarded after the session is\n    // ready, replay the broker's source-RTP pre-buffer (the current GOP from\n    // its keyframe) so the decoder initialises immediately instead of waiting\n    // for the camera's next IDR. The pre-buffer is captured in the broker's\n    // dispatch BEFORE fan-out, so `rtpData` is its tail — it's included in the\n    // replay and must not be forwarded again, hence the early return.\n    if (!this.rtpBootstrapDone) {\n      this.rtpBootstrapDone = true;\n      const bootstrap = this.rtpBootstrap?.() ?? [];\n      if (bootstrap.length > 0) {\n        let sent = 0;\n        for (const b of bootstrap) {\n          if (this.sendSourceRtpPacket(b, codec)) sent++;\n        }\n        this.logger.info('replayed source-RTP pre-buffer (instant start)', {\n          meta: { phase: 'session', sessionId: this.sessionId, codec, packets: bootstrap.length, sent },\n        });\n        return;\n      }\n    }\n\n    this.sendSourceRtpPacket(rtpData, codec);\n  }\n\n  /**\n   * Repacketize one source RTP packet and send it on the video sender.\n   * Shared by the live forward path and the pre-buffer bootstrap replay.\n   * Returns true when at least one output packet was sent.\n   */\n  private sendSourceRtpPacket(rtpData: Buffer, codec: 'H264' | 'H265'): boolean {\n    if (!this.videoSender || !_werift) return false;\n    const werift = _werift;\n\n    let srcPkt;\n    try {\n      srcPkt = werift.RtpPacket.deSerialize(rtpData);\n    } catch (err) {\n      this.logger.warn('source RTP deserialize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, codec, error: errMsg(err), len: rtpData.length },\n      });\n      return false;\n    }\n\n    if (this.sourceVideoSsrc === null) {\n      this.sourceVideoSsrc = srcPkt.header.ssrc;\n    }\n\n    const senderCodec = this.videoSender.codec;\n    const pt = senderCodec?.payloadType ?? (codec === 'H265' ? 97 : 96);\n\n    let outPkts;\n    try {\n      if (codec === 'H265') {\n        this.ensureH265Repacketizer();\n        outPkts = this.h265Repacketizer!.repacketize(srcPkt);\n      } else {\n        this.ensureH264Repacketizer();\n        outPkts = this.h264Repacketizer!.repacketize(srcPkt);\n      }\n    } catch (err) {\n      this.logger.warn('repacketize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, codec, error: errMsg(err) },\n      });\n      return false;\n    }\n\n    let sent = false;\n    for (const pkt of outPkts) {\n      pkt.header.payloadType = pt;\n      // Sync werift's sender SSRC/seq to our outbound stream on first\n      // packet — same fix the writeVideoNals path uses.\n      if (!this.videoRtpSynced) {\n        this.videoRtpSynced = true;\n        try {\n          this.videoSender.replaceRTP(\n            { sequenceNumber: pkt.header.sequenceNumber, timestamp: pkt.header.timestamp },\n            true,\n          );\n        } catch (e) {\n          this.logger.warn('replaceRTP failed (non-fatal)', {\n            meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(e) },\n          });\n        }\n      }\n      try {\n        this.videoSender.sendRtp(pkt);\n        // Record the forwarded packet's RTP timestamp (90 kHz) for the\n        // av-sync diagnostic. Diagnostic-only.\n        this.lastVideoRtpTs = pkt.header.timestamp >>> 0;\n        if (this.firstVideoRtpTs === null) {\n          this.firstVideoRtpTs = this.lastVideoRtpTs;\n          this.firstVideoSendWallMs = Date.now();\n        }\n        this.rtpPacketsSent++;\n        sent = true;\n        this.logFirstVideoRtp('source-rtp', codec);\n        if (this.debug && (this.rtpPacketsSent === 1 || this.rtpPacketsSent % 500 === 0)) {\n          this.logger.info('source-RTP forwarded', {\n            meta: { phase: 'session', sessionId: this.sessionId, codec, count: this.rtpPacketsSent },\n          });\n        }\n      } catch (err) {\n        if (this.rtpPacketsSent <= 10) {\n          this.logger.error('sendRtp (source forward) error', {\n            meta: { phase: 'session', sessionId: this.sessionId, codec, error: errMsg(err) },\n          });\n        }\n      }\n    }\n    return sent;\n  }\n\n  // -----------------------------------------------------------------------\n  // PLI handling — resend cached keyframe on picture loss\n  // -----------------------------------------------------------------------\n\n  private setupPliListener(): void {\n    const sender = this.videoSender;\n    if (!sender) return;\n\n    // werift exposes PLI via onPictureLossIndication (preferred) or the\n    // generic onRtcpFeedback. Not all builds surface these — degrade\n    // gracefully if neither is available.\n    const onPli = () => this.handlePli();\n\n    // Capture the remote receiver's RTCP Receiver Reports (jitter / loss).\n    // werift parses RR but never persists it, so the media-diag fields were\n    // always empty. Stash the latest RR matching our ssrc for diagnostics —\n    // jitter (90kHz units) and fractionLost tell us whether a remote viewer\n    // is seeing network jitter (timing) vs actual packet loss.\n    if (sender.onRtcp) {\n      sender.onRtcp.subscribe((rtcp) => {\n        const reports = rtcp?.reports;\n        if (!reports || reports.length === 0) return;\n        const mine = reports.find((r) => r.ssrc === sender.ssrc) ?? reports[0];\n        if (mine) this.lastRr = mine;\n        if (this.rrSink && mine) {\n          this.rrSink({\n            packetsLost: mine.packetsLost ?? mine.cumulativeLost ?? 0,\n            highestSequence: mine.highestSequence ?? 0,\n          });\n        }\n      });\n    }\n\n    if (sender.onPictureLossIndication) {\n      sender.onPictureLossIndication.subscribe(onPli);\n      this.logger.debug('PLI listener attached (onPictureLossIndication)', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n    } else if (sender.onRtcpFeedback) {\n      sender.onRtcpFeedback.subscribe((fb) => {\n        if (fb.type === 'nack' && fb.parameter === 'pli') onPli();\n        if (fb.type === 'ccm' && fb.parameter === 'fir') onPli();\n      });\n      this.logger.debug('PLI listener attached (onRtcpFeedback)', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n    } else {\n      this.logger.debug('PLI listener not available — werift version may not expose RTCP feedback', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n    }\n  }\n\n  private handlePli(): void {\n    const now = Date.now();\n    if (now - this.lastPliResendAt < AdaptiveSession.PLI_RESEND_COOLDOWN_MS) return;\n    this.lastPliResendAt = now;\n\n    // Source-RTP path (RTSP H.264 / H.265 — current cam 127, cam 99 H.265):\n    // replay the broker's source-RTP pre-buffer. The ring buffer contains\n    // the current GOP starting from the most recent keyframe — exactly\n    // what the browser is asking for after losing reference frames. This\n    // is the only recovery path that doesn't require an out-of-band\n    // keyframe-request to the camera; the keyframe is already in our\n    // hands, we just need to retransmit. Same mechanism as the first-\n    // packet instant-start replay, but on-demand for PLI recovery.\n    // Without this, an on-Wi-Fi camera that loses a packet from the\n    // initial IDR (or mid-GOP) freezes the viewer with no recovery —\n    // the browser sends PLI, our cached-keyframe path bails because\n    // `lastKeyframeNals` is only populated by `writeVideoNals` (the\n    // AnnexB path), not by `forwardSourceRtpVideo` (the source-RTP path).\n    if (this.rtpBootstrap) {\n      const codec = this.negotiatedCodec;\n      if (codec === 'H264' || codec === 'H265') {\n        const bootstrap = this.rtpBootstrap();\n        if (bootstrap.length > 0) {\n          let sent = 0;\n          for (const b of bootstrap) {\n            if (this.sendSourceRtpPacket(b, codec)) sent++;\n          }\n          this.logger.info('PLI received — replayed source-RTP pre-buffer', {\n            meta: { phase: 'session', sessionId: this.sessionId, codec, packets: bootstrap.length, sent },\n          });\n          return;\n        }\n      }\n      // Falls through to the AnnexB cached-keyframe path when the\n      // pre-buffer is empty (broker just rebuilt the reader, hasn't seen\n      // a keyframe yet) AND the AnnexB feed has populated lastKeyframeNals.\n    }\n\n    // AnnexB push-mode path (Reolink Baichuan, Frigate, push-rtp): re-send\n    // the last cached keyframe via writeVideoNals. Same packetization the\n    // feed loop uses for keyframes (single-NAL packets for VPS/SPS/PPS,\n    // FU fragmentation for IDR slice).\n    if (!this.lastKeyframeNals || this.lastKeyframeNals.length === 0) {\n      this.logger.debug('PLI received but no cached keyframe yet', {\n        meta: { phase: 'session', sessionId: this.sessionId },\n      });\n      return;\n    }\n\n    this.logger.info('PLI received — re-sending cached keyframe', {\n      meta: {\n        phase: 'session',\n        sessionId: this.sessionId,\n        nals: this.lastKeyframeNals.length,\n        totalBytes: this.lastKeyframeNals.reduce((s, n) => s + n.length, 0),\n      },\n    });\n    // Stamp with a CURRENT send-time RTP timestamp (not the cached\n    // keyframe's old ts): a resend carrying a past timestamp arrives\n    // out-of-order and the receiver discards it, which would perpetuate\n    // the PLI storm we're trying to break.\n    this.writeVideoNals(this.lastKeyframeNals, this.sendRtpTimestamp(), this.negotiatedCodec);\n  }\n\n  // -----------------------------------------------------------------------\n  // Frame feeding\n  // -----------------------------------------------------------------------\n\n  private startFeedingFrames(): void {\n    this.feedAbort = new AbortController();\n    const { signal } = this.feedAbort;\n\n    this.logger.info('Feed loop starting', {\n      meta: { phase: 'session', sessionId: this.sessionId, dtls: this.videoSender?.dtlsTransport?.state ?? 'unknown', needsTranscode: this.needsTranscode },\n    });\n\n    if (this.needsTranscode) {\n      this.startTranscodeFeed(signal);\n    } else {\n      this.startDirectFeed(signal);\n    }\n  }\n\n  /**\n   * Direct feed — source codec matches negotiated codec. Zero transcode.\n   * H.264 and H.265 both go through this path with codec-specific RTP\n   * packetization handled by writeVideoNals.\n   */\n  private startDirectFeed(signal: AbortSignal): void {\n\n    void (async () => {\n      let gotKeyframe = false;\n      // Sample-count accumulator for PCMU/PCMA audio RTP timestamps:\n      // 1 byte = 1 sample @ codec clock (PCMU 8 kHz mono). The counter\n      // advances exactly with the audio data we've forwarded, so RTP\n      // timestamps stay clock-coherent with the codec even when the\n      // source delivers below real-time. (A µs-derived alternative was\n      // tried but the broker doesn't seem to populate\n      // `frame.timestampMicros` with a usable source-clock value for\n      // audio — debug log gated on `this.debug` below dumps the raw\n      // value so we can confirm what the broker is passing through.)\n      let audioTimestampBase: number | null = null;\n      let frameCount = 0;\n\n      try {\n        for await (const mediaFrame of this.source) {\n          if (signal.aborted || this.closed) break;\n          frameCount++;\n          if (frameCount <= 5 || frameCount % 100 === 0) {\n            this.logger.debug('Frame received', {\n              meta: {\n                phase: 'session',\n                sessionId: this.sessionId,\n                frameCount,\n                type: mediaFrame.type,\n                size: mediaFrame.frame.data.length,\n                isKeyframe: mediaFrame.type === 'video' && 'isKeyframe' in mediaFrame.frame\n                  ? mediaFrame.frame.isKeyframe\n                  : undefined,\n              },\n            });\n          }\n\n          if (mediaFrame.type === \"video\") {\n            const frame = mediaFrame.frame;\n            const annexB = frame.codec === \"H264\"\n              ? convertH264ToAnnexB(frame.data)\n              : convertH265ToAnnexB(frame.data);\n\n            // Cache parameter sets BEFORE the keyframe gate. Cameras\n            // that ship VPS/SPS/PPS in their own non-keyframe access\n            // unit (or via the broker's synthetic SDP-replay packet —\n            // see StreamBroker.emitSdpParamSetsTo) deliver them in a\n            // frame whose `isKeyframe` is false. The previous code\n            // skipped such frames entirely with `continue`, so\n            // `lastVps/Sps/Pps` were never populated and HEVC decoding\n            // failed on the first IDR.\n            const __nalsForCaching = splitAnnexBToNals(annexB);\n            if (frame.codec === 'H265') {\n              for (const n of __nalsForCaching) {\n                const nalType = (n[0]! & 0x7e) >> 1;\n                if (nalType === 32) this.lastVps = Buffer.from(n);\n                if (nalType === 33) this.lastSps = Buffer.from(n);\n                if (nalType === 34) this.lastPps = Buffer.from(n);\n              }\n            } else {\n              for (const n of __nalsForCaching) {\n                const nalType = n[0]! & 0x1f;\n                if (nalType === 7) this.lastSps = Buffer.from(n);\n                if (nalType === 8) this.lastPps = Buffer.from(n);\n              }\n            }\n\n            if (!gotKeyframe) {\n              const isKey = frame.codec === \"H264\"\n                ? isH264IdrAccessUnit(annexB)\n                : isH265IrapAccessUnit(annexB);\n              if (!isKey) {\n                if (frameCount <= 3) {\n                  this.logger.debug('Skipping frame (waiting for keyframe)', {\n                    meta: { phase: 'session', sessionId: this.sessionId, frameCount, size: annexB.length },\n                  });\n                }\n                continue;\n              }\n              gotKeyframe = true;\n              const iceState = this.pc?.iceConnectionState ?? \"unknown\";\n              this.logger.info('First keyframe', {\n                meta: { phase: 'session', sessionId: this.sessionId, frameCount, size: annexB.length, ice: iceState },\n              });\n            }\n\n            // Send-time RTP timestamp (see sendRtpTimestamp): capture-time\n            // stamps caused ~80ms receiver jitter over Tailscale → PLI storm.\n            const rtpTs = this.sendRtpTimestamp();\n\n            // Filter out AUD and SEI — not needed in RTP.\n            // H.264: AUD=9, SEI=6 (mask 0x1f). H.265: AUD=35, SEI_PREFIX=39, SEI_SUFFIX=40 (mask (b>>1)&0x3f).\n            const isH265 = frame.codec === 'H265';\n            const allNals = splitAnnexBToNals(annexB);\n            const nals = allNals.filter((n: Buffer) => {\n              if (isH265) {\n                const t = (n[0]! & 0x7e) >> 1;\n                return t !== 35 && t !== 39 && t !== 40; // AUD, SEI_PREFIX, SEI_SUFFIX\n              }\n              const t = n[0]! & 0x1f;\n              return t !== 9 && t !== 6; // AUD, SEI\n            });\n\n            // Debug: dump NAL info for first few frames (only when debug enabled)\n            if (this.debug && (frameCount <= 5 || (gotKeyframe && frameCount <= (this._firstKeyFrame ?? 0) + 2))) {\n              const nalInfo = allNals.map((n: Buffer) => {\n                if (isH265) {\n                  const t = (n[0]! & 0x7e) >> 1;\n                  const names: Record<number, string> = {0:'TRAIL_N',1:'TRAIL_R',19:'IDR_W_RADL',20:'IDR_N_LP',21:'CRA',32:'VPS',33:'SPS',34:'PPS',35:'AUD',39:'SEI_P',40:'SEI_S'};\n                  return `${names[t] ?? `t${t}`}(${n.length})`;\n                }\n                const t = n[0]! & 0x1f;\n                const names: Record<number, string> = {1:'SLICE',5:'IDR',6:'SEI',7:'SPS',8:'PPS',9:'AUD'};\n                return `${names[t] ?? `t${t}`}(${n.length})`;\n              }).join(' + ');\n              this.logger.info('Frame NAL breakdown', {\n                meta: { phase: 'session', sessionId: this.sessionId, frameCount, codec: frame.codec, nals: nalInfo, sending: nals.length },\n              });\n              if (gotKeyframe && !this._firstKeyFrame) this._firstKeyFrame = frameCount;\n            }\n\n            if (nals.length > 0 && this.videoTrack) {\n              // For H.265 IRAP access units, send VPS+SPS+PPS as a\n              // single Aggregation Packet (RFC 7798 §4.4.2) BEFORE the\n              // IDR fragments. Chrome's HEVC\n              // depacketizer relies on this shape to initialise\n              // VideoToolbox properly. Sending the param sets as 3\n              // separate single-NAL packets caused `framesDecoded` to\n              // stay at 0 with PLI flooding even with valid SDP fmtp.\n              if (isH265) {\n                // ── Param set tracking + AP+IDR for keyframes ──\n                for (const n of nals) {\n                  const nalType = (n[0]! & 0x7e) >> 1;\n                  if (nalType === 32) this.lastVps = Buffer.from(n);\n                  if (nalType === 33) this.lastSps = Buffer.from(n);\n                  if (nalType === 34) this.lastPps = Buffer.from(n);\n                }\n\n                if (isH265IrapAccessUnit(annexB)) {\n                  // Resolve param sets — prefer inline, fall back to cache.\n                  const inlineVps = nals.find((n) => ((n[0]! & 0x7e) >> 1) === 32) ?? this.lastVps\n                  const inlineSps = nals.find((n) => ((n[0]! & 0x7e) >> 1) === 33) ?? this.lastSps\n                  const inlinePps = nals.find((n) => ((n[0]! & 0x7e) >> 1) === 34) ?? this.lastPps\n                  // VCL NALs (IRAP slices) — everything that isn't a param set.\n                  const vclNals = nals.filter((n) => {\n                    const t = (n[0]! & 0x7e) >> 1\n                    return t !== 32 && t !== 33 && t !== 34\n                  })\n                  if (inlineVps && inlineSps && inlinePps && vclNals.length > 0) {\n                    // Send VPS+SPS+PPS+IDR all via writeVideoNals — same\n                    // shape as the H.264 path that Chrome decodes\n                    // correctly. The previous Aggregation Packet (AP,\n                    // RFC 7798 §4.4.2) bundling produced a packet shape\n                    // that Chrome's HEVC depacketizer never accepted —\n                    // `framesAssembledFromMultiplePackets` stayed 0 and\n                    // `framesDecoded` stayed 0 even with valid SDP and\n                    // fmtp. Single-NAL packets for the param sets\n                    // followed by FU-fragmented IDR is what the H.264\n                    // path does for SPS/PPS/IDR and it works.\n                    const auNals = [inlineVps, inlineSps, inlinePps, ...vclNals]\n                    this.writeVideoNals(auNals, rtpTs, frame.codec)\n                    this.lastKeyframeNals = auNals.map((n) => Buffer.from(n))\n                  } else {\n                    // Missing param sets — fall back to \"send as-is\".\n                    this.writeVideoNals(nals, rtpTs, frame.codec)\n                  }\n                } else {\n                  // Non-IRAP frames (TRAIL_R / RADL_R / etc.) — send as-is.\n                  this.writeVideoNals(nals, rtpTs, frame.codec)\n                }\n              } else {\n                // ── H.264 param set tracking ──\n                let auNals: Buffer[] = nals\n                for (const n of nals) {\n                  const nalType = n[0]! & 0x1f;\n                  if (nalType === 7) {\n                    this.lastSps = Buffer.from(n);\n                    if (!this.profileLogged && n.length >= 4) {\n                      this.profileLogged = true;\n                      const plid = `${n[1]!.toString(16).padStart(2, '0')}${n[2]!.toString(16).padStart(2, '0')}${n[3]!.toString(16).padStart(2, '0')}`;\n                      this.logger.info('camera H.264 SPS profile-level-id', {\n                        meta: {\n                          phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n                          actualProfileLevelId: plid, advertisedProfileLevelId: '42e01f',\n                          match: plid === '42e01f',\n                          profileIdc: n[1], levelIdc: n[3],\n                        },\n                      });\n                    }\n                  }\n                  if (nalType === 8) this.lastPps = Buffer.from(n);\n                }\n                if (isH264IdrAccessUnit(annexB)) {\n                  const hasInlineSps = nals.some((n: Buffer) => (n[0]! & 0x1f) === 7);\n                  if (!hasInlineSps && this.lastSps && this.lastPps) {\n                    auNals = [this.lastSps, this.lastPps, ...nals]\n                  }\n                  this.lastKeyframeNals = auNals.map((n) => Buffer.from(n))\n                }\n                this.writeVideoNals(auNals, rtpTs, frame.codec);\n              }\n              if (this.debug && frameCount % 250 === 0) {\n                this.logger.info('Feed progress', {\n                  meta: {\n                    phase: 'session',\n                    sessionId: this.sessionId,\n                    frames: frameCount,\n                    rtpPackets: this.rtpPacketsSent,\n                    ice: this.pc?.iceConnectionState ?? \"?\",\n                    conn: this.pc?.connectionState ?? \"?\",\n                  },\n                });\n              }\n            }\n          } else if (mediaFrame.type === \"audio\") {\n            const frame = mediaFrame.frame;\n            if (!this.audioSender) continue;\n\n            // Derive RTP timestamp from the SOURCE frame's capture\n            // time (now in honest µs after the broker-side packet.pts\n            // ms→µs conversion in broker-webrtc-server.ts). Lets\n            // werift's RTCP Sender Report carry an honest NTP↔RTP\n            // mapping anchored on the source clock — the browser can\n            // apply its internal drift compensation instead of sync-\n            // gating the MediaStream into a freeze. PCMU clock is\n            // 8 kHz: 1 µs = 0.008 ticks → `µs × 8 / 1000`, u32-wrapped.\n            const rtpTs = (Math.floor(frame.timestampMicros * AdaptiveSession.PCMU_RTP_CLOCK_HZ / 1_000_000)) >>> 0;\n            // `audioTimestampBase` reference kept satisfied for tsc —\n            // see the loop-local declaration above; not used now.\n            void audioTimestampBase;\n\n            if (this.debug) {\n              this.logger.info('audio frame in (direct-feed)', {\n                meta: {\n                  phase: 'session',\n                  sessionId: this.sessionId,\n                  timestampMicros: frame.timestampMicros,\n                  dataLen: frame.data.length,\n                  computedRtpTs: rtpTs,\n                  codec: frame.codec,\n                },\n              });\n            }\n\n            // Audio pipeline latency: stash the SOURCE frame's capture\n            // timestamp (`timestampMicros`) and the send-time wall-clock at the\n            // instant we write it. The av-sync diagnostic compares them to\n            // surface transcode/buffer latency — the prime suspect for the\n            // constant ~0.5s audio lag (esp. the AAC→PCMU transcode path).\n            // Diagnostic-only; no effect on the RTP we send.\n            this.recordAudioSourceLatency(frame.timestampMicros);\n\n            this.writeAudio(frame.data, rtpTs, frame.codec);\n          }\n        }\n      } catch (err) {\n        if (!signal.aborted && !this.closed) {\n          this.logger.error('Feed error', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) } });\n        }\n      } finally {\n        if (!this.closed) {\n          if (this.debug) this.logger.info('Feed ended', { meta: { phase: 'session', sessionId: this.sessionId } });\n          void this.close();\n        }\n      }\n    })();\n  }\n\n  /**\n   * Transcode feed — re-encode the source to Constrained Baseline H.264:\n   * H.265→H.264 (browser lacks HEVC) or H.264 Main/High→Baseline (iOS only\n   * decodes Baseline). Pipes the source Annex-B to ffmpeg stdin, reads\n   * Baseline H.264 from stdout, and re-frames it into access units so each\n   * picture is sent with one RTP timestamp (see the stdout handler).\n   */\n  private startTranscodeFeed(signal: AbortSignal): void {\n    const { spawn } = require('node:child_process') as typeof import('node:child_process')\n\n    // Sample-count accumulator for the audio passthrough branch\n    // below — same role as `audioTimestampBase` in the direct-feed\n    // loop. Kept loop-local so it resets cleanly on every transcode\n    // restart.\n    let transcodeAudioTimestampBase: number | null = null\n\n    const inputFormat = transcodeInputFormat(this.sourceCodec)\n    this.logger.info(`Starting ${this.sourceCodec}→H.264 Baseline transcode feed`, {\n      meta: { phase: 'session', sessionId: this.sessionId, sourceCodec: this.sourceCodec, inputFormat },\n    })\n\n    // Hardware decode is tried first (CPU win) but is fragile for some HEVC\n    // streams (VideoToolbox can fail to decode). If ffmpeg dies before\n    // producing any output and a decode backend was set, we respawn once with\n    // software decode. The source keeps flowing into the new child, which\n    // re-syncs at the next keyframe (the broker drives a PLI). `proc.ff` is the\n    // current child so the stdin loop can retarget without restarting the\n    // source iterator.\n    let producedOutput = false\n    let fellBackToSoftware = false\n\n    const spawnFfmpeg = (decodeHwAccel: string | null): ReturnType<typeof spawn> => {\n      const ffmpegArgs = buildTranscodeFfmpegArgs(inputFormat, this.transcodeProfile, decodeHwAccel)\n      const child = spawn('ffmpeg', ffmpegArgs, { stdio: ['pipe', 'pipe', 'pipe'] })\n      if (decodeHwAccel) {\n        this.logger.info('Transcode feed using hardware decode', {\n          meta: { phase: 'session', sessionId: this.sessionId, decodeHwAccel },\n        })\n      }\n\n      child.on('exit', (code) => {\n        this.logger.info('Transcode ffmpeg exited', { meta: { phase: 'session', sessionId: this.sessionId, code } })\n        if (!producedOutput && decodeHwAccel && !fellBackToSoftware && !signal.aborted && !this.closed) {\n          fellBackToSoftware = true\n          this.logger.warn('Transcode hw decode produced no output — retrying software decode', {\n            meta: { phase: 'session', sessionId: this.sessionId, decodeHwAccel },\n          })\n          proc.ff = spawnFfmpeg(null)\n        }\n      })\n      child.stderr?.on('data', (chunk: Buffer) => {\n        const msg = chunk.toString().trim()\n        if (msg.length > 0) this.logger.warn('Transcode ffmpeg stderr', { meta: { sessionId: this.sessionId, msg } })\n      })\n\n      // Re-frame ffmpeg's raw-H.264 bytestream into access units and send each\n      // as ONE RTP-timestamped picture. ffmpeg stdout carries no frame\n      // boundaries, so we split into NALs up to the last complete start code,\n      // carry the trailing (still-growing) access unit across chunks, and emit\n      // every COMPLETE access unit with a fresh send-time RTP timestamp. The\n      // reframing state is per-child so a software-fallback respawn starts clean.\n      let pendingBuf = Buffer.alloc(0)\n      let carryNals: Buffer[] = []\n\n      child.stdout?.on('data', (chunk: Buffer) => {\n        producedOutput = true\n        pendingBuf = Buffer.concat([pendingBuf, chunk])\n        const lastSc = findLastStartCode(pendingBuf)\n        if (lastSc <= 0) return\n        const complete = pendingBuf.subarray(0, lastSc)\n        pendingBuf = Buffer.from(pendingBuf.subarray(lastSc))\n\n        const nals = carryNals.concat(splitAnnexBToNals(complete))\n        const accessUnits = groupNalsIntoAccessUnits(nals)\n        // The last access unit may still be growing — hold it back until the\n        // next picture boundary proves it complete.\n        if (accessUnits.length <= 1) { carryNals = nals; return }\n        carryNals = accessUnits[accessUnits.length - 1]!\n        for (let i = 0; i < accessUnits.length - 1; i++) {\n          // Drop SEI; AUDs were already removed by the grouping. Keep SPS/PPS/IDR/slices.\n          const auNals = accessUnits[i]!.filter((n) => (n[0]! & 0x1f) !== 6)\n          if (auNals.length > 0) this.writeVideoNals(auNals, this.sendRtpTimestamp(), 'H264')\n        }\n      })\n      return child\n    }\n\n    const proc = { ff: spawnFfmpeg(this.transcodeDecodeHwAccel) }\n\n    signal.addEventListener('abort', () => {\n      if (!proc.ff.killed) proc.ff.kill('SIGTERM')\n    })\n\n    // Feed the source Annex-B to ffmpeg stdin\n    void (async () => {\n      try {\n        for await (const mediaFrame of this.source) {\n          if (signal.aborted || this.closed) break\n          if (mediaFrame.type === 'video') {\n            if (proc.ff.stdin && !proc.ff.stdin.destroyed) {\n              proc.ff.stdin.write(mediaFrame.frame.data)\n            }\n          } else if (mediaFrame.type === 'audio') {\n            // Audio passthrough (no transcode needed). Same µs-derived\n            // RTP timestamp as the direct-feed branch — see comment\n            // there for the RTCP SR / NTP↔RTP rationale.\n            const frame = mediaFrame.frame\n            const rtpTs = (Math.floor(frame.timestampMicros * AdaptiveSession.PCMU_RTP_CLOCK_HZ / 1_000_000)) >>> 0\n            void transcodeAudioTimestampBase;\n\n            if (this.debug) {\n              this.logger.info('audio frame in (transcode passthrough)', {\n                meta: {\n                  phase: 'session',\n                  sessionId: this.sessionId,\n                  timestampMicros: frame.timestampMicros,\n                  dataLen: frame.data.length,\n                  computedRtpTs: rtpTs,\n                  codec: frame.codec,\n                },\n              })\n            }\n            // Audio pipeline latency probe for the av-sync diagnostic — see the\n            // direct-feed audio branch. Diagnostic-only.\n            this.recordAudioSourceLatency(frame.timestampMicros)\n            this.writeAudio(frame.data, rtpTs, frame.codec)\n          }\n        }\n      } catch (err) {\n        if (!signal.aborted && !this.closed) {\n          this.logger.error('Transcode feed error', { meta: { sessionId: this.sessionId, error: errMsg(err) } })\n        }\n      } finally {\n        if (proc.ff.stdin && !proc.ff.stdin.destroyed) proc.ff.stdin.end()\n        if (!this.closed) void this.close()\n      }\n    })()\n  }\n\n  // 1100 (not 1200): keep the full wire packet under the Tailscale/WireGuard\n  // overlay MTU (1280). See H264_REPACKETIZER_MTU for the rationale — at 1200\n  // a keyframe's RTP packets are ~1266 B and silently drop over the overlay,\n  // causing a never-decoding PLI storm that works fine on a 1500-MTU LAN.\n  private static readonly MAX_RTP_PAYLOAD = 1100;\n\n  private rtpPacketsSent = 0;\n  private rtpLogCounter = 0;\n  /** One-shot guards for the \"first video/audio RTP\" debug markers. */\n  private firstVideoRtpLogged = false;\n  private firstAudioRtpLogged = false;\n  /**\n   * One-shot guard for the audio-silence pre-stream. When the source audio\n   * pipeline (e.g. AAC → PCMU transcode) lags video by a CONSTANT startup\n   * delay, the first real audio frame arrives N ms AFTER the first video\n   * frame went out → audio `firstSendWallMs` − video `firstSendWallMs` is a\n   * persistent positive gap, the browser inflates its audio jitter buffer\n   * over that gap, and the user perceives ~1 s of audio-late lag.\n   *\n   * Right before the very first real audio packet is written we emit N PCMU\n   * silence packets (0xFF bytes, 160 samples each = 20 ms) so the audio\n   * sender's first wall-clock instant aligns with the video sender's first\n   * wall-clock instant, then the real packet continues seamlessly. See\n   * `prestreamAudioSilenceIfNeeded` + `computeAudioSilencePadPackets`.\n   */\n  private firstAudioFramePadded = false;\n  /** PCMU frame size in samples per packet — 160 = 20 ms at 8 kHz. */\n  private static readonly PCMU_FRAME_SAMPLES = 160;\n  /** PCMU silence byte. PCMU's µ-law representation of digital zero is 0xFF. */\n  private static readonly PCMU_SILENCE_BYTE = 0xff;\n  /** Upper bound on the audio silence pre-stream duration (ms) — guards a\n   *  runaway from blasting tens of seconds of silence onto the wire. */\n  private static readonly AUDIO_SILENCE_MAX_PAD_MS = 2_000;\n\n  /**\n   * Emit the one-shot \"first video RTP sent\" debug marker. Both egress\n   * paths (`writeVideoNals` synthesis + `sendSourceRtpPacket` repacketizer\n   * forward) funnel here so the marker fires exactly once regardless of\n   * which path the session took. Gated on `debug`.\n   */\n  private logFirstVideoRtp(path: 'writeVideoNals' | 'source-rtp', codec: string): void {\n    if (!this.debug || this.firstVideoRtpLogged) return;\n    this.firstVideoRtpLogged = true;\n    this.logger.info('first video RTP sent', {\n      meta: {\n        phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n        path, codec, negotiated: this.negotiatedCodec,\n        sinceCreatedMs: Date.now() - this.createdAt,\n      },\n    });\n  }\n\n  /**\n   * Emit the one-shot \"first audio RTP sent\" debug marker, including the\n   * source audio codec and the negotiated sender clock rate — the data\n   * needed to diagnose A/V sync and \"no audio\" cases. Gated on `debug`.\n   */\n  private logFirstAudioRtp(sourceCodec: string | undefined, senderPt: number): void {\n    if (!this.debug || this.firstAudioRtpLogged) return;\n    this.firstAudioRtpLogged = true;\n    const senderCodec = this.audioSender?.codec;\n    this.logger.info('first audio RTP sent', {\n      meta: {\n        phase: 'session', sessionId: this.sessionId, deviceId: this.deviceId,\n        sourceCodec: sourceCodec ?? 'unknown',\n        senderMime: senderCodec?.mimeType ?? 'unknown',\n        senderPayloadType: senderPt,\n        sinceCreatedMs: Date.now() - this.createdAt,\n      },\n    });\n  }\n\n  /**\n   * RTP timestamp (90 kHz) derived from the hub's MONOTONIC clock at the\n   * moment of sending — NOT the camera capture time.\n   *\n   * Forwarding live media with capture-time RTP timestamps creates a cadence\n   * mismatch vs the actual send time: the pipeline/decoder adds variable\n   * latency between capture (`frame.timestampMicros`) and the instant we push\n   * the packets out. A remote receiver measures that mismatch as high RTCP\n   * jitter (~80 ms observed over Tailscale, with ZERO packet loss) — its\n   * jitter buffer can't lock onto a stable clock, so frames are discarded and\n   * it PLI-storms. A ~0-RTT LAN absorbs it; a real remote path does not.\n   *\n   * Stamping off send time makes the RTP-timestamp cadence equal the actual\n   * transmit cadence, so arrival cadence matches and the receiver's jitter\n   * collapses to true network jitter. Same effect ffmpeg/libwebrtc achieve by\n   * pacing output. All video send paths (live feed + PLI keyframe resend)\n   * share this one clock so timestamps stay monotonic.\n   */\n  private sendRtpTimestamp(): number {\n    const now = performance.now();\n    if (this.videoRtpClockBaseMs === null) this.videoRtpClockBaseMs = now;\n    return Math.floor((now - this.videoRtpClockBaseMs) * 90) >>> 0;\n  }\n\n  /**\n   * Record the latest audio SOURCE frame's capture timestamp and the send-time\n   * wall-clock at the moment we write it. Diagnostic-only — read by the\n   * debug \"av-sync diag\" log to estimate audio transcode/buffer latency. Does\n   * not touch any send path. A non-finite capture timestamp is ignored so a\n   * source that doesn't stamp frames can't poison the figure.\n   */\n  private recordAudioSourceLatency(captureMicros: number): void {\n    if (!Number.isFinite(captureMicros)) return;\n    this.lastAudioSrcCaptureMicros = captureMicros;\n    this.lastAudioSendWallMs = Date.now();\n  }\n\n  private writeVideoNals(nals: Buffer[], rtpTs: number, codec: VideoCodec): void {\n    if (!this.videoSender || !_werift) {\n      if (this.rtpLogCounter === 0) {\n        this.logger.warn('writeVideoNals: no sender', {\n          meta: {\n            phase: 'session',\n            sessionId: this.sessionId,\n            hasVideoSender: !!this.videoSender,\n            hasWerift: !!_werift,\n          },\n        });\n        this.rtpLogCounter++;\n      }\n      return;\n    }\n    const werift = _werift;\n    // Record the last video RTP timestamp for the av-sync diagnostic\n    // (90 kHz). Diagnostic-only — does not influence packetization.\n    this.lastVideoRtpTs = rtpTs >>> 0;\n    if (this.firstVideoRtpTs === null) {\n      this.firstVideoRtpTs = this.lastVideoRtpTs;\n      this.firstVideoSendWallMs = Date.now();\n    }\n    // Use the payload type from the negotiated codec on the sender,\n    // not a hardcoded value. After SDP negotiation, werift assigns the\n    // correct PT to the sender's codec. Using a different PT causes\n    // the browser to ignore the packets.\n    const senderCodec = this.videoSender.codec;\n    const pt = senderCodec?.payloadType ?? (codec === \"H264\" ? 96 : 97);\n\n    const sendPkt = (payload: Buffer, marker: boolean) => {\n      try {\n        const header = new werift.RtpHeader();\n        header.payloadType = pt;\n        header.timestamp = rtpTs;\n        header.marker = marker;\n        header.sequenceNumber = (this.videoSeqNum = (this.videoSeqNum + 1) & 0xffff);\n        const pkt = new werift.RtpPacket(header, payload);\n\n        // FIX: Sync SSRC/seq on first packet via werift's replaceRTP.\n        // Without this, werift's internal SSRC and the RTP header SSRC\n        // diverge, and the browser ignores the packets.\n        if (!this.videoRtpSynced) {\n          this.videoRtpSynced = true;\n          try {\n            this.videoSender!.replaceRTP(header, true);\n          } catch (e) {\n            this.logger.warn('replaceRTP failed (non-fatal)', {\n              meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(e) },\n            });\n          }\n        }\n\n        // Debug: check sender state for first few packets\n        if (this.debug && this.rtpPacketsSent < 3) {\n          const dtls = this.videoSender?.dtlsTransport;\n          const senderCodec = this.videoSender?.codec;\n          const ssrc = this.videoSender?.ssrc;\n          this.logger.info('sendRtp debug', {\n            meta: {\n              phase: 'session',\n              sessionId: this.sessionId,\n              packetIndex: this.rtpPacketsSent,\n              dtls: dtls?.state ?? 'none',\n              codec: senderCodec?.mimeType ?? 'NULL',\n              payloadType: senderCodec?.payloadType ?? '?',\n              ssrc: String(ssrc),\n              payloadSize: payload.length,\n            },\n          });\n        }\n        this.videoSender!.sendRtp(pkt);\n\n        this.rtpPacketsSent++;\n        this.logFirstVideoRtp('writeVideoNals', codec);\n        if (this.debug && (this.rtpPacketsSent === 1 || this.rtpPacketsSent % 500 === 0)) {\n          this.logger.info('RTP packets sent', {\n            meta: { phase: 'session', sessionId: this.sessionId, count: this.rtpPacketsSent, payload: payload.length },\n          });\n        }\n      } catch (err) {\n        if (this.rtpPacketsSent <= 10) {\n          this.logger.error('sendRtp error', {\n            meta: { phase: 'session', sessionId: this.sessionId, packetIndex: this.rtpPacketsSent, error: errMsg(err) },\n          });\n        }\n      }\n    };\n\n    for (let i = 0; i < nals.length; i++) {\n      const nal = nals[i]!;\n      const isLastNal = i === nals.length - 1;\n\n      if (nal.length <= AdaptiveSession.MAX_RTP_PAYLOAD) {\n        sendPkt(nal, isLastNal);\n      } else if (codec === 'H265') {\n        // H.265 FU fragmentation (RFC 7798 §4.4.3)\n        // NAL header is 2 bytes: [F|type(6)|layerId_hi(1)] [layerId_lo(5)|tid(3)]\n        const nalType = (nal[0]! & 0x7e) >> 1;\n        const layerId = ((nal[0]! & 0x01) << 5) | ((nal[1]! & 0xf8) >> 3);\n        const tid = nal[1]! & 0x07;\n        // FU NAL header: type=49 (FU), keep layerId+tid\n        const fuNalHeader0 = (49 << 1) | (layerId >> 5);\n        const fuNalHeader1 = ((layerId & 0x1f) << 3) | tid;\n        const nalBody = nal.subarray(2); // skip 2-byte NAL header\n\n        let offset = 0;\n        let isFirst = true;\n        while (offset < nalBody.length) {\n          const end = Math.min(offset + AdaptiveSession.MAX_RTP_PAYLOAD - 3, nalBody.length);\n          const isLast = end >= nalBody.length;\n\n          let fuHeader = nalType & 0x3f;\n          if (isFirst) fuHeader |= 0x80;\n          if (isLast) fuHeader |= 0x40;\n\n          const fragment = Buffer.alloc(3 + (end - offset));\n          fragment[0] = fuNalHeader0;\n          fragment[1] = fuNalHeader1;\n          fragment[2] = fuHeader;\n          nalBody.copy(fragment, 3, offset, end);\n\n          sendPkt(fragment, isLastNal && isLast);\n          offset = end;\n          isFirst = false;\n        }\n      } else {\n        // H.264 FU-A fragmentation (RFC 6184 §5.8)\n        const nalHeader = nal[0]!;\n        const fnri = nalHeader & 0xe0;\n        const nalType = nalHeader & 0x1f;\n        const fuIndicator = fnri | 28;\n        const nalBody = nal.subarray(1);\n\n        let offset = 0;\n        let isFirst = true;\n        while (offset < nalBody.length) {\n          const end = Math.min(offset + AdaptiveSession.MAX_RTP_PAYLOAD - 2, nalBody.length);\n          const isLast = end >= nalBody.length;\n\n          let fuHeader = nalType;\n          if (isFirst) fuHeader |= 0x80;\n          if (isLast) fuHeader |= 0x40;\n\n          const fragment = Buffer.alloc(2 + (end - offset));\n          fragment[0] = fuIndicator;\n          fragment[1] = fuHeader;\n          nalBody.copy(fragment, 2, offset, end);\n\n          sendPkt(fragment, isLastNal && isLast);\n          offset = end;\n          isFirst = false;\n        }\n      }\n    }\n  }\n\n  private writeAudio(data: Buffer, rtpTs: number, codec?: string): void {\n    if (!this.audioSender || !_werift) return;\n    const werift = _werift;\n    // Use the payload type from the negotiated codec on the sender —\n    // SAME pattern as `writeVideoNals`. werift assigns the negotiated\n    // codec to the sender after SDP exchange completes; reading\n    // `senderCodec.payloadType` keeps us in sync with whatever the\n    // browser actually accepted in its answer. Fallback to the\n    // static OFFER-side PT only if werift hasn't bound the codec yet\n    // (timing race during the very first packet).\n    // NOTE: NOT calling `audioSender.replaceRTP(header, true)`. Video\n    // does it (and works), but audio's SSRC binding is more brittle\n    // in werift — overriding the sender's SSRC with our header's\n    // default (zero) leaves the SRTP context out of sync with the\n    // SSRC the browser expects from the answer SDP, so the browser\n    // gets RTP with the right PT but can't decrypt → silent drop.\n    // werift assigns the audio sender's SSRC during transceiver\n    // setup; let `sendRtp` use it directly.\n    const senderCodec = this.audioSender.codec;\n    const fallbackPt = codec === \"Pcmu\" || codec === \"Pcma\" ? 0 : 111;\n    const pt = senderCodec?.payloadType ?? fallbackPt;\n    // ONE-SHOT: pre-stream PCMU silence so audio first-send wall-clock\n    // aligns to video first-send wall-clock. Runs only on the very first\n    // real audio packet; idempotent on every subsequent call.\n    this.prestreamAudioSilenceIfNeeded(rtpTs >>> 0, pt);\n    try {\n      const header = new werift.RtpHeader();\n      header.payloadType = pt;\n      header.timestamp = rtpTs;\n      // marker=true on every audio packet — browsers tolerate it on\n      // G.711 codecs, and some receive paths stall the jitter buffer\n      // waiting for marker=true to start playback. Setting marker\n      // on the first packet only (RFC 3551 talkspurt convention)\n      // produced silent receivers in practice.\n      header.marker = true;\n      header.sequenceNumber = (this.audioSeqNum = (this.audioSeqNum + 1) & 0xffff);\n      const pkt = new werift.RtpPacket(header, data);\n      this.audioSender.sendRtp(pkt);\n      // Record the last audio RTP timestamp (PCMU 8 kHz) for the av-sync\n      // diagnostic. Diagnostic-only.\n      this.lastAudioRtpTs = rtpTs >>> 0;\n      if (this.firstAudioRtpTs === null) {\n        this.firstAudioRtpTs = this.lastAudioRtpTs;\n        this.firstAudioSendWallMs = Date.now();\n      }\n      this.logFirstAudioRtp(codec, pt);\n    } catch (err) {\n      this.logger.debug('Audio write error', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) } });\n    }\n  }\n\n  /**\n   * Pre-stream N PCMU silence packets (160 bytes of 0xFF each, 20 ms / 8 kHz)\n   * BEFORE the first real audio packet so the audio sender's first wall-clock\n   * instant aligns with the video sender's first wall-clock instant. One-shot\n   * per session, gated on `firstAudioFramePadded`. The packet count comes from\n   * the pure `computeAudioSilencePadPackets` helper (gap clamp, sub-frame\n   * floor, maxPad cap). Silence packets occupy RTP timestamps\n   * `[realRtpTs − N·160, realRtpTs)` so the real packet's own timestamp stays\n   * exactly what the caller computed and continuity holds across the boundary\n   * (no gap, no overlap). Sequence numbers continue from `audioSeqNum`.\n   *\n   * After emitting the silence, `firstAudioRtpTs` / `firstAudioSendWallMs`\n   * reference the FIRST silence packet (so `firstSendGapMs` in the av-sync\n   * diag reads ~0). Any failure on a single `sendRtp` is logged and the loop\n   * continues — partial alignment is better than aborting.\n   */\n  private prestreamAudioSilenceIfNeeded(realRtpTs: number, payloadType: number): void {\n    if (this.firstAudioFramePadded) return;\n    this.firstAudioFramePadded = true;\n    if (!this.audioSender || !_werift) return;\n    const werift = _werift;\n    const nowMs = Date.now();\n    const plan = planAudioSilencePadPackets({\n      nowMs,\n      firstVideoSendWallMs: this.firstVideoSendWallMs,\n      framesizeSamples: AdaptiveSession.PCMU_FRAME_SAMPLES,\n      rateHz: AdaptiveSession.PCMU_RTP_CLOCK_HZ,\n      maxPadMs: AdaptiveSession.AUDIO_SILENCE_MAX_PAD_MS,\n      realRtpTs,\n      audioSeqNum: this.audioSeqNum,\n    });\n    if (plan.packets.length === 0) return;\n    const silencePayload = Buffer.alloc(\n      AdaptiveSession.PCMU_FRAME_SAMPLES,\n      AdaptiveSession.PCMU_SILENCE_BYTE,\n    );\n    for (let i = 0; i < plan.packets.length; i++) {\n      const planned = plan.packets[i]!;\n      try {\n        const header = new werift.RtpHeader();\n        header.payloadType = payloadType;\n        header.timestamp = planned.rtpTs;\n        header.marker = true;\n        header.sequenceNumber = planned.seq;\n        const pkt = new werift.RtpPacket(header, silencePayload);\n        this.audioSender.sendRtp(pkt);\n        this.lastAudioRtpTs = planned.rtpTs;\n        if (this.firstAudioRtpTs === null) {\n          this.firstAudioRtpTs = planned.rtpTs;\n          // Anchor to the VIDEO first-send wall-clock so the av-sync diag's\n          // `firstSendGapMs = firstAudioSendWallMs − firstVideoSendWallMs`\n          // reads ~0 after this fix. Falls back to `nowMs` defensively.\n          this.firstAudioSendWallMs = this.firstVideoSendWallMs ?? nowMs;\n        }\n      } catch (err) {\n        this.logger.debug('Audio silence pre-stream error', {\n          meta: {\n            phase: 'session',\n            sessionId: this.sessionId,\n            error: errMsg(err),\n            packetIndex: i,\n            totalPackets: plan.packets.length,\n          },\n        });\n      }\n    }\n    // Carry forward the seqNum past the silence packets so the real packet\n    // (which does `this.audioSeqNum = (this.audioSeqNum + 1) & 0xffff`) lands\n    // on the next contiguous seq, matching the plan.\n    this.audioSeqNum = plan.nextAudioSeqNum;\n    if (this.debug) {\n      this.logger.info('audio silence pre-stream emitted', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          deviceId: this.deviceId,\n          packets: plan.packets.length,\n          gapMs: this.firstVideoSendWallMs === null ? null : nowMs - this.firstVideoSendWallMs,\n          firstSilenceRtpTs: plan.packets[0]?.rtpTs ?? null,\n          realRtpTs,\n          frameSize: AdaptiveSession.PCMU_FRAME_SAMPLES,\n        },\n      });\n    }\n  }\n\n  // -----------------------------------------------------------------------\n  // RTCP stats collection\n  // -----------------------------------------------------------------------\n\n  private startStatsCollection(): void {\n    // Runs regardless of `onStats` so the diagnostic log below always fires —\n    // it's the decisive evidence for \"hub sends RTP but browser loses it\" vs\n    // \"hub never sends\" on a remote (WAN) session.\n    if (this.statsTimer) return;\n\n    this.statsTimer = setInterval(() => {\n      if (!this.pc || this.closed) return;\n      this.collectStats();\n    }, 3_000);\n  }\n\n  private collectStats(): void {\n    if (!this.pc) return;\n\n    // Receiver Report captured live from sender.onRtcp (werift never persists\n    // lastReceiverReport, so the old getSenders()-based read was always empty).\n    const report: WeriftReceiverReport | undefined = this.lastRr ?? undefined;\n\n    const fractionLost = report?.fractionLost ?? 0;\n    const packetsLost = report?.packetsLost ?? report?.cumulativeLost ?? 0;\n    const jitter = report?.jitter ?? 0;\n    const rtt = report?.roundTripTime ?? report?.rtt ?? 0;\n\n    // Decisive diagnostic: is the hub actually emitting RTP, and does the\n    // browser's RTCP receiver-report show loss? `rtpPacketsSent` climbing +\n    // a receiver-report with loss ⇒ packets sent but dropped on hub→browser\n    // (path/relay loss). `rtpPacketsSent` flat ⇒ werift isn't sending at all.\n    //\n    // The nominated-pair fields are the smoking gun for the remote-NAT\n    // case: on a 4G/CGNAT retest we expect `selectedLocalType:'relay'` +\n    // `selectedRemoteType:'relay'` once relay-only is forced. A\n    // `host`/`srflx` local with a `relay` remote (or no pair) on a\n    // remote session is the dead-pair symptom this fix targets.\n    const nominated = this.readNominatedPair();\n    if (this.debug) {\n      this.logger.info('webrtc media diag', {\n        meta: {\n          phase: 'session',\n          sessionId: this.sessionId,\n          deviceId: this.deviceId,\n          rtpPacketsSent: this.rtpPacketsSent,\n          dtls: this.videoSender?.dtlsTransport?.state ?? 'unknown',\n          ice: this.state,\n          forceRelayOnly: this.forceRelayOnly,\n          hasReceiverReport: !!report,\n          fractionLostPct: report ? Math.round((fractionLost / 256) * 100) : -1,\n          packetsLost,\n          // video RTP clock is 90 kHz → jitter ticks / 90 = milliseconds.\n          jitterMs: report ? Math.round(jitter / 90) : -1,\n          rttMs: Math.round(rtt * 1000),\n          selectedLocalType: nominated?.localType ?? 'none',\n          selectedLocalAddr: nominated?.localAddr ?? 'none',\n          selectedRemoteType: nominated?.remoteType ?? 'none',\n          selectedRemoteAddr: nominated?.remoteAddr ?? 'none',\n        },\n      });\n    }\n\n    if (report && this.onStats) {\n      this.onStats({\n        sessionId: this.sessionId,\n        packetLoss: fractionLost / 256, // Fraction lost is 0–255\n        jitterMs: jitter,\n        rttMs: rtt * 1000,\n        packetsReceived: 0,\n        packetsLost,\n        timestamp: Date.now(),\n      });\n    }\n\n    // Debug-only A/V lip-sync diagnostic — shares this 3s tick.\n    this.emitAvSyncDiag();\n  }\n\n  /** Negotiated audio RTP clock rate in Hz (PCMU is always 8 kHz here; falls\n   *  back to the sender's clockRate if werift reports one). Video is 90 kHz. */\n  private static readonly VIDEO_RTP_CLOCK_HZ = 90_000;\n  private static readonly PCMU_RTP_CLOCK_HZ = 8_000;\n\n  /**\n   * Emit the periodic \"av-sync diag\" log (debug only, ~every 3s) to gather the\n   * evidence for the steady \"audio plays ~0.5s behind video\" lag — WITHOUT\n   * changing any audio/video behaviour. For BOTH tracks it reports the last\n   * sent RTP timestamp, the clock rate, the media-time (rtpTs/rate), the\n   * wall-clock elapsed since session create, and the per-track skew\n   * (media-time − wall-clock-elapsed); plus the INTER-TRACK skew (audio\n   * media-seconds − video media-seconds) — negative means audio's RTP clock\n   * runs behind video's, and by how much. It also surfaces the audio pipeline\n   * latency (source capture timestamp vs send wall-clock) — the prime suspect\n   * for the constant offset — and documents the werift RTCP-SR limitation.\n   */\n  private emitAvSyncDiag(): void {\n    if (!this.debug) return;\n\n    const elapsedMs = Date.now() - this.createdAt;\n    // The session only ever offers/negotiates PCMU for audio (8 kHz) — the\n    // typed werift sender surface exposes mimeType/payloadType but no\n    // clockRate, so the codec-fixed 8 kHz constant is the honest rate here.\n    const audioRate = AdaptiveSession.PCMU_RTP_CLOCK_HZ;\n\n    // Feed SESSION-RELATIVE rtp timestamps (last − first, 32-bit-wrap-safe via\n    // `>>> 0`) so video's huge source clock and audio's near-zero sample counter\n    // share a session-zero base — otherwise videoMediaSec is hours and\n    // interTrackSkew is garbage.\n    const skew = computeAvSyncSkew({\n      video:\n        this.lastVideoRtpTs === null || this.firstVideoRtpTs === null\n          ? null\n          : {\n              rtpTs: (this.lastVideoRtpTs - this.firstVideoRtpTs) >>> 0,\n              rate: AdaptiveSession.VIDEO_RTP_CLOCK_HZ,\n              elapsedMs,\n            },\n      audio:\n        this.lastAudioRtpTs === null || this.firstAudioRtpTs === null\n          ? null\n          : {\n              rtpTs: (this.lastAudioRtpTs - this.firstAudioRtpTs) >>> 0,\n              rate: audioRate,\n              elapsedMs,\n            },\n    });\n\n    // Audio pipeline latency: source-frame capture (µs) vs the send wall-clock\n    // captured when we wrote it. We can't subtract a µs capture clock from a\n    // ms wall clock to a single absolute latency (the two clocks have unknown\n    // offsets) — but the DRIFT of (sendWall − capture) across ticks is the\n    // transcode/buffer latency growth. Report both raw anchors so the operator\n    // can watch the delta evolve; a steady gap that matches the inter-track\n    // skew points the finger at the audio pipeline (esp. AAC→PCMU transcode).\n    const audioCaptureMs =\n      this.lastAudioSrcCaptureMicros === null ? null : this.lastAudioSrcCaptureMicros / 1000;\n    const audioSendWallMs = this.lastAudioSendWallMs;\n    const audioPipelineDeltaMs =\n      audioCaptureMs !== null && audioSendWallMs !== null\n        ? Math.round(audioSendWallMs - audioCaptureMs)\n        : null;\n\n    // Constant startup offset between tracks: how much LATER (wall-clock) audio\n    // first flowed vs video. Positive ⇒ audio started later ⇒ steady audio-late\n    // lag of roughly this magnitude (video passthrough ~0 latency, audio behind\n    // its decode/transcode pipeline). The decisive number for a CONSTANT lag.\n    const firstSendGapMs =\n      this.firstAudioSendWallMs !== null && this.firstVideoSendWallMs !== null\n        ? this.firstAudioSendWallMs - this.firstVideoSendWallMs\n        : null;\n\n    // RTCP Sender Report (ntpTime, rtpTimestamp) — NOT reachable from our\n    // typed werift surface. werift builds the SR inside `RTCRtpSender.runRtcp`\n    // from the PRIVATE fields `ntpTimestamp` / `rtpTimestamp` (set in\n    // `sendRtp`); the `.d.ts` declares them `private`, and `onRtcp` only fires\n    // on INCOMING RTCP (the remote's RR/SR), never the SR we emit. Surfacing\n    // the pair would require an `as any` reach into werift internals, which the\n    // project forbids — so we report the limitation instead of the value.\n    this.logger.info('av-sync diag', {\n      meta: {\n        phase: 'session',\n        sessionId: this.sessionId,\n        deviceId: this.deviceId,\n        elapsedMs,\n        // Video track.\n        videoRtpTs: this.lastVideoRtpTs,\n        videoClockRate: AdaptiveSession.VIDEO_RTP_CLOCK_HZ,\n        videoMediaSec: skew.video ? Number(skew.video.mediaSeconds.toFixed(3)) : null,\n        videoSkewSec: skew.video ? Number(skew.video.skewSeconds.toFixed(3)) : null,\n        // Constant startup offset audio-vs-video (ms); ≈ steady perceived lag.\n        firstSendGapMs,\n        // Audio track (PCMU).\n        audioRtpTs: this.lastAudioRtpTs,\n        audioClockRate: audioRate,\n        audioMediaSec: skew.audio ? Number(skew.audio.mediaSeconds.toFixed(3)) : null,\n        audioSkewSec: skew.audio ? Number(skew.audio.skewSeconds.toFixed(3)) : null,\n        // The decisive figure: negative ⇒ audio media clock lags video.\n        interTrackSkewSec:\n          skew.interTrackSkewSeconds === null\n            ? null\n            : Number(skew.interTrackSkewSeconds.toFixed(3)),\n        // Audio pipeline latency anchors (see comment above).\n        audioSrcCaptureMs: audioCaptureMs === null ? null : Number(audioCaptureMs.toFixed(1)),\n        audioSendWallMs,\n        audioPipelineDeltaMs,\n        // RTCP SR limitation (see comment above).\n        rtcpSr: 'unavailable: werift sender ntpTimestamp/rtpTimestamp are private; onRtcp is incoming-only',\n      },\n    });\n  }\n}\n\n\n/** Find the byte offset of the last 0x00000001 start code in a buffer. */\nfunction findLastStartCode(buf: Buffer): number {\n  for (let i = buf.length - 4; i >= 0; i--) {\n    if (buf[i] === 0 && buf[i + 1] === 0 && buf[i + 2] === 0 && buf[i + 3] === 1) return i\n  }\n  return -1\n}\n","export interface LossSample {\n  readonly packetsLost: number\n  readonly highestSequence: number\n}\n\nexport interface LossTrackerConfig {\n  readonly analyzeWindowMs: number\n  readonly minFailPercent: number\n  readonly waitAfterResetMs: number\n}\n\ninterface StampedSample extends LossSample {\n  readonly time: number\n}\n\n/** Delta between two RTCP extended (32-bit) highest-sequence values, with\n *  32-bit wraparound handling. */\nfunction seqDelta(from: number, to: number): number {\n  const d = to - from\n  return d < 0 ? d + 0x1_0000_0000 : d\n}\n\n/**\n * Windowed packet-loss tracker. Mirrors the adaptive tracker used by\n * reference NVR implementations: loss ratio = ΔpacketsLost / Δsequence\n * across the oldest and newest in-window samples; `isFailing` when that\n * exceeds `minFailPercent`. After a tier change (`markChanged`), samples\n * are ignored for `waitAfterResetMs` so the renegotiation's own transient\n * loss doesn't cascade a second downgrade.\n */\nexport class LossTracker {\n  private samples: StampedSample[] = []\n  private resetUntil = 0\n\n  constructor(private readonly config: LossTrackerConfig) {}\n\n  markChanged(now: number): void {\n    this.samples = []\n    this.resetUntil = now + this.config.waitAfterResetMs\n  }\n\n  track(sample: LossSample, now: number): void {\n    if (now < this.resetUntil) return\n    this.samples.push({ ...sample, time: now })\n    const cutoff = now - this.config.analyzeWindowMs\n    this.samples = this.samples.filter((s) => s.time > cutoff)\n  }\n\n  /** Current loss ratio over the window, or 0 when fewer than two samples. */\n  lossRatio(): number {\n    const first = this.samples[0]\n    const last = this.samples[this.samples.length - 1]\n    if (!first || !last || first === last) return 0\n    const seq = seqDelta(first.highestSequence, last.highestSequence)\n    if (seq <= 0) return 0\n    return (last.packetsLost - first.packetsLost) / seq\n  }\n\n  isFailing(now: number): boolean {\n    if (now < this.resetUntil) return false\n    return this.lossRatio() > this.config.minFailPercent\n  }\n}\n","import { CAM_PROFILE_ORDER, type CamProfile } from '@camstack/types'\nimport { LossTracker, type LossSample, type LossTrackerConfig } from './loss-tracker.js'\n\n/** ABR tier == the broker profile tier. */\nexport type Tier = CamProfile\nconst TIER_ORDER: readonly Tier[] = CAM_PROFILE_ORDER\n\nexport interface AdaptiveControllerConfig extends LossTrackerConfig {\n  readonly upgradeHealthyMs: number\n  readonly minTimeBetweenChangesMs: number\n}\n\nexport interface AdaptiveControllerOptions {\n  readonly initialTier: Tier\n  readonly config: AdaptiveControllerConfig\n  readonly onSwitchTier: (tier: Tier) => void\n  readonly onReduceMtu: () => void\n  readonly onTranscodeDown: () => void\n}\n\n/**\n * Ladder state machine for an adaptive WebRTC session. Fed loss samples\n * (RTCP RR or client-reported) via `onLoss`; `tick` evaluates the ladder.\n * Downgrade order: tier high→mid→low, then MTU reduction at low, then\n * transcode-down. Upgrade one tier after sustained health + hysteresis.\n * Side effects run through injected callbacks (testable without werift).\n */\nexport class AdaptiveController {\n  private readonly tracker: LossTracker\n  private tier: Tier\n  private mtuReduced = false\n  private transcodeEngaged = false\n  private lastChangeAt = 0\n  private healthySince = 0\n\n  constructor(private readonly opts: AdaptiveControllerOptions) {\n    this.tracker = new LossTracker(opts.config)\n    this.tier = opts.initialTier\n  }\n\n  get currentTier(): Tier { return this.tier }\n\n  onLoss(sample: LossSample, now: number): void {\n    this.tracker.track(sample, now)\n  }\n\n  tick(now: number): void {\n    const failing = this.tracker.isFailing(now)\n    if (failing) { this.healthySince = now; this.handleFailing(now); return }\n    this.handleHealthy(now)\n  }\n\n  private canUpgrade(now: number): boolean {\n    return now - this.lastChangeAt >= this.opts.config.minTimeBetweenChangesMs\n      || this.lastChangeAt === 0\n  }\n\n  private commitChange(now: number): void {\n    this.lastChangeAt = now\n    this.tracker.markChanged(now)\n  }\n\n  private handleFailing(now: number): void {\n    const idx = TIER_ORDER.indexOf(this.tier)\n    if (idx < TIER_ORDER.length - 1) {\n      this.tier = TIER_ORDER[idx + 1]!\n      this.opts.onSwitchTier(this.tier)\n      this.commitChange(now)\n      return\n    }\n    // at lowest tier\n    if (!this.mtuReduced) {\n      this.mtuReduced = true\n      this.opts.onReduceMtu()\n      this.commitChange(now)\n      return\n    }\n    if (!this.transcodeEngaged) {\n      this.transcodeEngaged = true\n      this.opts.onTranscodeDown()\n      this.commitChange(now)\n    }\n  }\n\n  private handleHealthy(now: number): void {\n    if (now - this.healthySince < this.opts.config.upgradeHealthyMs) return\n    if (!this.canUpgrade(now)) return\n    const idx = TIER_ORDER.indexOf(this.tier)\n    if (idx > 0) {\n      this.tier = TIER_ORDER[idx - 1]!\n      this.mtuReduced = false\n      this.transcodeEngaged = false\n      this.opts.onSwitchTier(this.tier)\n      this.commitChange(now)\n      this.healthySince = now\n    }\n  }\n}\n","/**\n * Broker-integrated WebRTC server.\n *\n * Unlike the old `addon-webrtc-adaptive` which spawned ffmpeg to re-encode\n * the broker's RTSP restream (Camera → Broker → RTSP → ffmpeg libx264 → RTP),\n * this server subscribes directly to `IStreamBroker.onEncodedData()` and\n * forward the raw H.264 packets to WebRTC sessions as-is — zero transcode,\n * zero extra RTSP connection, zero latency from re-encoding.\n *\n * Flow: Camera → RTSP → Broker → EncodedPacket → RTP packetization → WebRTC\n */\n\nimport crypto from 'node:crypto'\nimport type { EncodedPacket, IStreamBroker, IScopedLogger, Unsubscribe, EncodeProfile, BrokerConsumerAttribution, WebrtcStreamTarget, CamProfile } from '@camstack/types'\nimport { errMsg } from '@camstack/types'\nimport { AdaptiveSession, type IceServerEntry, type IceCandidatePayload } from './session.js'\nimport { AdaptiveController, type Tier } from './adaptive-controller.js'\nimport type { FrameSource, MediaFrame } from './types.js'\nimport { convertH264ToAnnexB, extractH264ParamSets, isBaselineProfileLevelId } from './h264-utils.js'\nimport { WEBRTC_FALLBACK_PROFILE } from './webrtc-fallback-profile.js'\n\n// ---------------------------------------------------------------------------\n// Types\n// ---------------------------------------------------------------------------\n\nexport interface BrokerWebrtcServerOptions {\n  readonly logger: IScopedLogger\n  readonly getIceServers?: () => Promise<readonly IceServerEntry[]>\n  readonly iceServers?: readonly IceServerEntry[]\n  readonly icePortRange?: [number, number]\n  readonly iceAdditionalHostAddresses?: readonly string[]\n  /**\n   * Optional injected async accessor for a device's LATEST client-reported\n   * packet-loss percentage (0–100), or null when no recent client report\n   * exists. Polled by the adaptive controller's tick so a downgrade can fire\n   * from client telemetry (e.g. the benchmark \"Network Simulation\" presets)\n   * even when no RTCP Receiver Report has arrived. The broker synthesizes a\n   * monotonic {@link LossSample} from this percentage locally (see\n   * `startAdaptive`). Only consulted for sessions created via the\n   * `/adaptive` alias.\n   */\n  readonly clientLossPct?: (deviceId: number) => Promise<number | null>\n  /**\n   * Optional accessor for the decode `-hwaccel` backend the in-process\n   * transcode feed should use (probe-gated upstream). Resolved once and cached.\n   * Absent ⇒ software decode (current default).\n   */\n  readonly resolveTranscodeDecodeHwAccel?: () => Promise<string | null>\n}\n\n// ---------------------------------------------------------------------------\n// Adaptive controller tuning\n// ---------------------------------------------------------------------------\n\n/** How often the adaptive controller re-evaluates the ladder (ms). */\nconst ADAPTIVE_TICK_MS = 2_000\n/** Ladder tuning shared by every adaptive session. */\nconst ADAPTIVE_CONFIG = {\n  analyzeWindowMs: 15_000, minFailPercent: 0.05, waitAfterResetMs: 8_000,\n  upgradeHealthyMs: 60_000, minTimeBetweenChangesMs: 60_000,\n} as const\n\n/**\n * The adaptive controller's desired tier, carried across a close-and-recreate\n * re-offer so the recreated session resolves to the chosen tier AND re-arms\n * its controller. `forceTranscodeDown` additionally pins the recreated\n * session to a reduced transcode profile (the transcode-down rung).\n */\ninterface AdaptiveIntent {\n  readonly prefersTier: Tier\n  readonly forceTranscodeDown: boolean\n}\n\ninterface SessionEntry {\n  readonly session: AdaptiveSession\n  /**\n   * Current broker this session is fed from. Fixed for the session's\n   * lifetime: an adaptive tier switch is a close-and-recreate re-offer\n   * (a NEW SessionEntry on the chosen-tier broker), never an in-place\n   * rebind of this field.\n   */\n  readonly brokerId: string\n  /** Unsubscribe from broker's onEncodedData. */\n  readonly unsubBroker: Unsubscribe\n  /**\n   * Unsubscribe from broker's `onVideoRtp` for sessions that forward\n   * source RTP through the codec repacketizer (RTP-bearing sources, H.264\n   * AND H.265). `null` for AnnexB-only push sessions (which use the\n   * AnnexB feed-loop path). Fixed for the session's lifetime — an adaptive\n   * tier switch recreates the session rather than rebinding this feed.\n   */\n  readonly unsubRtp: Unsubscribe | null\n  /**\n   * Unsubscribe from broker's `onSdpParameterSets` for H.265 sessions\n   * that need late delivery of VPS/SPS/PPS. Sessions created during a\n   * battery-cam wake-up window register this so the repacketizer is\n   * seeded as soon as the broker's first `onVideoTrack` fires —\n   * without it the AP codec packet is never emitted and Chrome's HEVC\n   * decoder never initialises. `null` for H.264 sessions and for H.265\n   * sessions that found the SDP params already populated at create\n   * time.\n   */\n  readonly unsubSdpParams: Unsubscribe | null\n  /** Close the push source. */\n  readonly closeSource: () => void\n  /** Adaptive ladder controller — present only for sessions created via\n   *  the `/adaptive` alias. Drives tier switches / MTU / transcode-down. */\n  adaptiveController?: AdaptiveController\n  /** Interval that polls client loss + ticks the adaptive controller.\n   *  Cleared in `cleanupSession`. Present only for adaptive sessions. */\n  adaptiveTimer?: ReturnType<typeof setInterval>\n  /**\n   * Remote-address tagging is fire-at-most-once per session. The ICE\n   * `connected` transition fires on the INITIAL connect AND on every\n   * reconnect after a `disconnected`; without this latch each reconnect\n   * would re-arm a fresh retry chain, leaving overlapping untracked\n   * timers under ICE flap. Set true once the nominated pair is read +\n   * tagged OR the retry budget is exhausted; subsequent `connected`\n   * transitions are then no-ops.\n   */\n  attributionTagged?: boolean\n  /** Pending retry-timer handle for the remote-address tagging chain.\n   *  Cleared in `cleanupSession` so no timer fires after teardown. */\n  attributionTimer?: ReturnType<typeof setTimeout>\n}\n\n// ---------------------------------------------------------------------------\n// iOS Baseline transcode decision\n// ---------------------------------------------------------------------------\n\n/**\n * Whether a WebRTC session for this broker must re-encode the source to\n * Constrained Baseline H.264 so iOS (WebKit) can decode it. True only for\n * an H.264 source whose profile-level-id is known AND non-Baseline\n * (Main/High). Unknown profile (e.g. a battery cam before its first\n * keyframe) ⇒ passthrough — desktop still decodes, and a reconnect once the\n * profile is known gets it right. H.265 sources are driven by codec\n * negotiation (`resolveVideoTranscode`), never this flag.\n */\nexport function resolveBrokerTranscodeToBaseline(\n  sessionCodec: 'H264' | 'H265',\n  sourceProfileLevelId: string | undefined,\n): boolean {\n  if (sessionCodec !== 'H264') return false\n  if (!sourceProfileLevelId) return false\n  return !isBaselineProfileLevelId(sourceProfileLevelId)\n}\n\n/** profile_idc (first byte of a profile-level-id) as a number, e.g.\n *  `640c34` → 0x64 (100, High), `4d0033` → 0x4d (77, Main),\n *  `42e01f` → 0x42 (66, Constrained Baseline). 0 for malformed input. */\nfunction h264ProfileIdc(profileLevelId: string): number {\n  if (profileLevelId.length < 2) return 0\n  const idc = Number.parseInt(profileLevelId.slice(0, 2), 16)\n  return Number.isNaN(idc) ? 0 : idc\n}\n\n\n/** All H.264 `profile-level-id` values advertised in an SDP (offer or\n *  answer), lower-cased. Browsers list one fmtp line per offered H.264\n *  profile (e.g. Constrained Baseline + Main + High), so this is the set\n *  of profiles the client's decoder can handle. Empty when none is given —\n *  per RFC 6184 that implies Constrained Baseline. */\nexport function extractOfferedH264ProfileLevelIds(sdp: string): string[] {\n  const ids: string[] = []\n  const re = /profile-level-id=([0-9a-fA-F]{6})/g\n  let m: RegExpExecArray | null\n  while ((m = re.exec(sdp)) !== null) ids.push(m[1]!.toLowerCase())\n  return ids\n}\n\n/**\n * Capability-driven transcode decision for the CLIENT-offer flow.\n *\n * Unlike the server-offer path — where the server can only assume iOS\n * decodes Constrained Baseline and so transcodes any Main/High source\n * (`resolveBrokerTranscodeToBaseline`) — a client offer carries the\n * browser's real decode capabilities. iOS 18 (WebKit) offers H.264 High,\n * so a 4K Main/High source passes through untouched (werift echoes the\n * client codec; the native decoder handles it). Only fall back to a\n * Baseline re-encode when the source is non-Baseline AND none of the\n * client's offered profiles can decode it (an older Baseline-only client,\n * or an offer that advertised no H.264 profile at all).\n *\n * \"Can decode\" uses profile_idc ordering: a decoder advertising High\n * (idc 100) decodes Main (77) and Baseline (66); Constrained Baseline\n * (66) decodes only Baseline. So passthrough iff `max(offered idc) >=\n * source idc`. Unknown source profile ⇒ passthrough (desktop still\n * decodes; a reconnect once the profile is known re-decides).\n */\nexport function resolveClientOfferTranscodeToBaseline(\n  sessionCodec: 'H264' | 'H265',\n  sourceProfileLevelId: string | undefined,\n  offeredProfileLevelIds: readonly string[],\n): boolean {\n  if (sessionCodec !== 'H264') return false\n  if (!sourceProfileLevelId) return false\n  if (isBaselineProfileLevelId(sourceProfileLevelId)) return false\n  // Capability-driven only: passthrough iff the client offered a\n  // profile that decodes the source. Earlier we added a level ≥ 5.0\n  // force-transcode (cam 127 main case) on the theory that broadcast\n  // Wi-Fi sources lose packets too aggressively for passthrough; but\n  // ffmpeg `ultrafast` at the `-maxrate` cap didn't actually shrink\n  // the output volumetrically (it just dropped framerate), and the\n  // re-encoded keyframes turned out larger and more loss-prone than\n  // the source ones. The audio drift that originally appeared to be\n  // a passthrough symptom was actually fix #25 (ms → µs broker-side\n  // EncodedPacket.pts conversion). Reverted to capability-only.\n  const sourceIdc = h264ProfileIdc(sourceProfileLevelId)\n  const maxOfferedIdc = offeredProfileLevelIds.reduce(\n    (max, plid) => Math.max(max, h264ProfileIdc(plid)),\n    0,\n  )\n  return maxOfferedIdc < sourceIdc\n}\n\n/**\n * Best-effort extraction of the source H.264 profile-level-id (e.g.\n * `4d0033` Main 5.1, `42e01f` Constrained Baseline) from the camera's SPS.\n * Checks SDP-published parameter sets first (RTSP cameras publish the SPS\n * as a bare NAL in the SDP), then scans the pre-buffer (push/Baichuan\n * cameras emit the SPS inline in keyframes). Returns undefined when no SPS\n * is available yet (e.g. a battery cam before its first keyframe).\n */\nexport function deriveH264ProfileLevelId(\n  sdpParameterSets: ReadonlyArray<Buffer> | null,\n  preBuffer: readonly EncodedPacket[],\n): string | undefined {\n  if (sdpParameterSets) {\n    for (const ps of sdpParameterSets) {\n      // Bare NAL (no start code); H.264 SPS = type 7. profile_idc /\n      // constraints / level_idc are the 3 bytes after the NAL header.\n      if (ps.length >= 4 && (ps[0]! & 0x1f) === 7) {\n        return Buffer.from([ps[1]!, ps[2]!, ps[3]!]).toString('hex')\n      }\n    }\n  }\n  for (const pkt of preBuffer) {\n    if (pkt.type !== 'video') continue\n    const { profileLevelId } = extractH264ParamSets(convertH264ToAnnexB(pkt.data))\n    if (profileLevelId) return profileLevelId\n  }\n  return undefined\n}\n\n// ---------------------------------------------------------------------------\n// Adaptive source selection\n// ---------------------------------------------------------------------------\n\n/** Fallback resolution/bitrate per label when the broker hasn't reported stats yet. */\nconst LABEL_DEFAULTS: Readonly<Record<string, { pixels: number; bitrateKbps: number }>> = {\n  high: { pixels: 1920 * 1080, bitrateKbps: 4000 },\n  mid:  { pixels: 1280 * 720,  bitrateKbps: 1200 },\n  low:  { pixels: 640 * 360,   bitrateKbps: 400 },\n}\nconst TIER_PREFERENCE: readonly string[] = ['high', 'mid', 'low']\n\n/** Transcode-down ceiling: cap egress height to 360p. */\nconst TRANSCODE_DOWN_MAX_HEIGHT = 360\n/** Transcode-down ceiling: cap egress bitrate to 400 kbps (the `low` LADDER default). */\nconst TRANSCODE_DOWN_MAX_BITRATE_KBPS = 400\n\n/**\n * Derive a reduced transcode profile for the transcode-down rung. Caps the\n * egress to 360p (preserving aspect ratio when both source dims are known)\n * and the bitrate to a low ceiling, forcing an H.264 re-encode so the\n * reduced shape actually reaches the wire. Pure — returns a new profile,\n * never mutates `base`.\n */\nfunction reduceProfileForTranscodeDown(base: EncodeProfile): EncodeProfile {\n  const srcWidth = base.video.width\n  const srcHeight = base.video.height\n  let width = srcWidth\n  let height = srcHeight\n  if (srcWidth !== undefined && srcHeight !== undefined && srcHeight > TRANSCODE_DOWN_MAX_HEIGHT) {\n    height = TRANSCODE_DOWN_MAX_HEIGHT\n    // Preserve aspect ratio; force an even width for the encoder.\n    const scaled = Math.round((srcWidth * TRANSCODE_DOWN_MAX_HEIGHT) / srcHeight)\n    width = scaled % 2 === 0 ? scaled : scaled + 1\n  } else if (srcHeight === undefined) {\n    height = TRANSCODE_DOWN_MAX_HEIGHT\n  }\n  const bitrateKbps = base.video.bitrateKbps !== undefined\n    ? Math.min(base.video.bitrateKbps, TRANSCODE_DOWN_MAX_BITRATE_KBPS)\n    : TRANSCODE_DOWN_MAX_BITRATE_KBPS\n  return {\n    ...base,\n    video: {\n      ...base.video,\n      // Force a real re-encode so the reduced shape reaches the wire.\n      codec: base.video.codec === 'copy' ? 'h264' : base.video.codec,\n      ...(width !== undefined ? { width } : {}),\n      height,\n      bitrateKbps,\n    },\n  }\n}\n\ninterface ClientHints {\n  readonly viewportWidth?: number\n  readonly viewportHeight?: number\n  readonly devicePixelRatio?: number\n  readonly downlinkMbps?: number\n  readonly prefersTier?: string\n}\n\nfunction scoreBroker(\n  tier: CamProfile | null,\n  broker: IStreamBroker,\n  hints: ClientHints,\n): number {\n  const stats = broker.getStats()\n  const fallback = tier ? LABEL_DEFAULTS[tier] : undefined\n  const bitrateKbps = stats.bitrateKbps > 0 ? stats.bitrateKbps : (fallback?.bitrateKbps ?? 2000)\n  // Approximate pixel count from tier defaults (broker doesn't track resolution)\n  const srcPixels = fallback?.pixels ?? 1920 * 1080\n\n  const dpr = hints.devicePixelRatio ?? 1\n  const targetW = (hints.viewportWidth ?? 1920) * dpr\n  const targetH = (hints.viewportHeight ?? 1080) * dpr\n  const targetPixels = targetW * targetH\n\n  const pixelDistance = Math.abs(srcPixels - targetPixels) / 1_000_000\n\n  let bandwidthPenalty = 0\n  if (hints.downlinkMbps && hints.downlinkMbps > 0 && bitrateKbps > 0) {\n    const availableKbps = hints.downlinkMbps * 1000 * 0.6\n    if (bitrateKbps > availableKbps) {\n      bandwidthPenalty = ((bitrateKbps - availableKbps) / availableKbps) * 100\n    }\n  }\n  return pixelDistance + bandwidthPenalty\n}\n\n// ---------------------------------------------------------------------------\n// BrokerWebrtcServer\n// ---------------------------------------------------------------------------\n\nexport class BrokerWebrtcServer {\n  private readonly logger: IScopedLogger\n  private readonly getIceServersFn: BrokerWebrtcServerOptions['getIceServers']\n  private readonly staticIceServers: readonly IceServerEntry[] | undefined\n  private readonly icePortRange: [number, number] | undefined\n  private readonly iceAdditionalHostAddresses: readonly string[] | undefined\n  private readonly clientLossPct: BrokerWebrtcServerOptions['clientLossPct']\n  private readonly resolveTranscodeDecodeHwAccelFn: BrokerWebrtcServerOptions['resolveTranscodeDecodeHwAccel']\n  /** Cached decode-hwaccel backend for transcode feeds (resolved once). */\n  private transcodeDecodeHwAccel: string | null | undefined = undefined\n\n  /** brokerId → IStreamBroker reference. Set by the broker manager on registration. */\n  private readonly brokers = new Map<string, IStreamBroker>()\n  /**\n   * Resolves a brokerId to its current profile tier (`CamProfile`)\n   * via the manager's `assignments` map, or `null` for brokers that\n   * are only kept alive by a manual activation. Injected by the\n   * StreamBrokerManager since this server doesn't own the assignment\n   * map directly. Defaults to a no-op resolver — adaptive bitrate logic\n   * still works (falls back to \"mid\" defaults) but tier-explicit\n   * `prefersTier` requests always pick the first matching candidate.\n   */\n  private profileTierResolver: ((brokerId: string) => CamProfile | null) | null = null\n  /** sessionId → session entry. */\n  private readonly sessions = new Map<string, SessionEntry>()\n  /**\n   * Carries the adaptive controller's desired tier (+ forced transcode-down)\n   * across a close-and-recreate re-offer, so the recreated session resolves\n   * to the chosen tier AND re-arms its controller. Written by\n   * `switchSessionTier`, consumed once by `setupSessionForBroker`, cleared on\n   * close.\n   */\n  private readonly adaptiveIntent = new Map<string, AdaptiveIntent>()\n  private stopped = false\n\n  constructor(options: BrokerWebrtcServerOptions) {\n    this.logger = options.logger\n    this.getIceServersFn = options.getIceServers\n    this.staticIceServers = options.iceServers\n    this.icePortRange = options.icePortRange\n    this.iceAdditionalHostAddresses = options.iceAdditionalHostAddresses\n    this.clientLossPct = options.clientLossPct\n    this.resolveTranscodeDecodeHwAccelFn = options.resolveTranscodeDecodeHwAccel\n  }\n\n  /**\n   * Resolve (once, cached) the decode-hwaccel backend for transcode feeds.\n   * Never throws — a resolver failure means software decode.\n   */\n  private async getTranscodeDecodeHwAccel(): Promise<string | null> {\n    if (this.transcodeDecodeHwAccel !== undefined) return this.transcodeDecodeHwAccel\n    if (!this.resolveTranscodeDecodeHwAccelFn) {\n      this.transcodeDecodeHwAccel = null\n      return null\n    }\n    try {\n      this.transcodeDecodeHwAccel = await this.resolveTranscodeDecodeHwAccelFn()\n    } catch (err) {\n      this.logger.warn('resolveTranscodeDecodeHwAccel failed — software decode', {\n        meta: { error: err instanceof Error ? err.message : String(err) },\n      })\n      this.transcodeDecodeHwAccel = null\n    }\n    return this.transcodeDecodeHwAccel\n  }\n\n  // ── Broker registration (called by StreamBrokerManager) ─────────────\n\n  registerBroker(brokerId: string, broker: IStreamBroker): void {\n    this.brokers.set(brokerId, broker)\n    this.logger.info('WebRTC broker registered', { meta: { brokerId } })\n  }\n\n  /**\n   * Provide a resolver that maps brokerId → current profile tier. The\n   * manager injects one wired to its `assignments` map. Without it the\n   * server still serves brokers but can't honour client `prefersTier`\n   * hints precisely (it falls through to the order-based pick).\n   */\n  setProfileTierResolver(\n    resolver: (brokerId: string) => CamProfile | null,\n  ): void {\n    this.profileTierResolver = resolver\n  }\n\n  unregisterBroker(brokerId: string): void {\n    this.brokers.delete(brokerId)\n    // Close all sessions on this broker\n    for (const [sid, entry] of this.sessions) {\n      if (entry.brokerId === brokerId) {\n        this.cleanupSession(sid)\n        void entry.session.close().catch(() => {})\n      }\n    }\n    this.logger.info('WebRTC broker unregistered', { meta: { brokerId } })\n  }\n\n  supportsStream(streamId: string): boolean {\n    // streamId is brokerId (e.g. \"6/high\") or bare deviceKey\n    // Accept if any registered broker matches\n    if (this.brokers.has(streamId)) return true\n    // Check for adaptive alias: \"6/adaptive\" → any broker for device \"6\"\n    const slash = streamId.lastIndexOf('/')\n    if (slash > 0) {\n      const tail = streamId.slice(slash + 1)\n      if (tail === 'adaptive') {\n        const deviceKey = streamId.slice(0, slash)\n        for (const bid of this.brokers.keys()) {\n          if (bid.startsWith(`${deviceKey}/`)) return true\n        }\n      }\n    }\n    return false\n  }\n\n  // ── Session lifecycle ───────────────────────────────────────────────\n\n  async createSession(\n    streamId: string,\n    hints: ClientHints = {},\n    opts: { streamingDebug?: boolean; relayOnly?: boolean; transcodeProfile?: EncodeProfile; consumerAttribution?: BrokerConsumerAttribution } = {},\n  ): Promise<{ sessionId: string; sdpOffer: string }> {\n    const setup = await this.setupSessionForBroker(streamId, hints, opts)\n    try {\n      // Force TURN-relay-only ICE for remote (non-LAN) viewers BEFORE\n      // `createOffer()` builds the PeerConnection — the patched werift\n      // emits a genuinely relay-only SDP under `iceTransportPolicy:'relay'`,\n      // so a CGNAT client gets a clean relay↔relay path instead of\n      // werift nominating a dead host/hairpin-srflx pair. LAN viewers\n      // (`relayOnly` absent/false) keep the direct host/srflx path.\n      if (opts.relayOnly === true) {\n        setup.session.setForceRelayOnly(true)\n      }\n      const offer = await setup.session.createOffer()\n      setup.sessionLogger.info('WebRTC session created', {\n        meta: {\n          sessionId: setup.sessionId,\n          deviceId: setup.deviceId,\n          brokerId: setup.brokerId,\n          iceServers: setup.iceServerCount,\n          codec: setup.sessionCodec,\n          useRtpRepacketizer: setup.useRtpRepacketizer,\n          relayOnly: opts.relayOnly === true,\n        },\n      })\n      return { sessionId: setup.sessionId, sdpOffer: offer.sdp }\n    } catch (err) {\n      this.cleanupSession(setup.sessionId)\n      await setup.session.close().catch(() => {})\n      throw err\n    }\n  }\n\n  /**\n   * WHEP-style client-offer signaling: caller has already constructed\n   * an SDP offer (Alexa / WHEP / mobile clients that do\n   * `createOffer()` locally) and posts it here. We build the same\n   * broker subscription + AdaptiveSession plumbing as `createSession`,\n   * then call `session.handleOffer(clientOffer)` to produce the\n   * server's SDP answer.\n   *\n   * The session lifecycle is identical to `createSession` — broker\n   * unsubscribe and `closeSession(sessionId)` still apply — so the\n   * RTC controller / Alexa handler closes the session the same way.\n   */\n  async handleOffer(\n    streamId: string,\n    clientOfferSdp: string,\n    hints: ClientHints = {},\n    opts: { streamingDebug?: boolean; sessionId?: string; relayOnly?: boolean; nonTrickle?: boolean; disableTurn?: boolean; disableIpv6?: boolean; transcodeProfile?: EncodeProfile; consumerAttribution?: BrokerConsumerAttribution } = {},\n  ): Promise<{ sessionId: string; sdpAnswer: string }> {\n    // Pass the client's offer SDP into setup so the transcode decision is\n    // capability-driven: the offered H.264 profiles decide passthrough vs\n    // Baseline re-encode (iOS 18 offers High → 4K Main/High passes through).\n    const setup = await this.setupSessionForBroker(streamId, hints, { ...opts, clientOfferSdp })\n    try {\n      // Relay policy must be set BEFORE handleOffer builds the PeerConnection.\n      // Cloud peers (Alexa) force TURN-relay-only; LAN/Tailscale browser\n      // viewers (relayOnly absent/false) keep the direct host/srflx path.\n      if (opts.relayOnly === true) {\n        setup.session.setForceRelayOnly(true)\n      }\n      const answer = await setup.session.handleOffer(\n        { sdp: clientOfferSdp, type: 'offer' },\n        { nonTrickle: opts.nonTrickle === true },\n      )\n      setup.sessionLogger.info('WebRTC session created (client-offer)', {\n        meta: {\n          sessionId: setup.sessionId,\n          deviceId: setup.deviceId,\n          brokerId: setup.brokerId,\n          iceServers: setup.iceServerCount,\n          codec: setup.sessionCodec,\n          useRtpRepacketizer: setup.useRtpRepacketizer,\n          relayOnly: opts.relayOnly === true,\n        },\n      })\n      return { sessionId: setup.sessionId, sdpAnswer: answer.sdp }\n    } catch (err) {\n      this.cleanupSession(setup.sessionId)\n      await setup.session.close().catch(() => {})\n      throw err\n    }\n  }\n\n  /**\n   * Shared session bring-up: resolve broker → subscribe to encoded\n   * data + RTP → flush pre-buffer → seed H.265 codec info → register\n   * the SessionEntry. Returns the `AdaptiveSession` ready for SDP\n   * negotiation (either `createOffer()` for the server-offer flow or\n   * `handleOffer(clientOffer)` for the WHEP / Alexa flow).\n   *\n   * The returned session has its `onClosed` hook already wired to\n   * cleanup, and the SessionEntry is already in `this.sessions` keyed\n   * by `sessionId` — callers MUST cleanup + close on negotiation\n   * failure (both `createSession` and `handleOffer` do this in their\n   * catch blocks).\n   */\n  private async setupSessionForBroker(\n    streamId: string,\n    hints: ClientHints,\n    opts: { streamingDebug?: boolean; sessionId?: string; relayOnly?: boolean; disableTurn?: boolean; disableIpv6?: boolean; clientOfferSdp?: string; transcodeProfile?: EncodeProfile; consumerAttribution?: BrokerConsumerAttribution },\n  ): Promise<{\n    sessionId: string\n    session: AdaptiveSession\n    sessionLogger: IScopedLogger\n    deviceId: number\n    brokerId: string\n    sessionCodec: 'H264' | 'H265'\n    useRtpRepacketizer: boolean\n    iceServerCount: number\n  }> {\n    if (this.stopped) throw new Error('Server stopped')\n\n    // Adaptive re-offer: a close-and-recreate re-offer carries the same\n    // sessionId, so an `adaptiveIntent` entry keyed by it pins the recreated\n    // session to the controller's chosen tier (and applies the reduced\n    // transcode profile when transcode-down was requested). Only adaptive\n    // streams ever record an intent; non-adaptive (profile / cam-stream)\n    // sessions never have one and resolve exactly as before.\n    const isAdaptiveStream = streamId.endsWith('/adaptive')\n    const intent = isAdaptiveStream && opts.sessionId !== undefined\n      ? this.adaptiveIntent.get(opts.sessionId)\n      : undefined\n    // Honour the controller's chosen tier on the recreated broker resolution\n    // so `startAdaptive` reads `initialTier` from that broker and resumes at\n    // the right rung (BUG #1: upgrade-back was dead because the re-offer\n    // landed on a raw cam-stream and never re-armed the controller).\n    const resolveHints: ClientHints = intent !== undefined\n      ? { ...hints, prefersTier: intent.prefersTier }\n      : hints\n    const brokerId = this.resolveBrokerId(streamId, resolveHints)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) throw new Error(`No broker for stream \"${streamId}\" (resolved: \"${brokerId}\")`)\n\n    // Transcode-down rung (BUG #4): when the controller forced a transcode-down,\n    // the recreated session re-encodes its egress to a reduced (≤360p, low\n    // bitrate) profile derived from whatever transcode profile it would\n    // otherwise use. The baked default is used when no profile was supplied.\n    const baseTranscodeProfile = opts.transcodeProfile ?? WEBRTC_FALLBACK_PROFILE\n    const effectiveTranscodeProfile = intent?.forceTranscodeDown === true\n      ? reduceProfileForTranscodeDown(baseTranscodeProfile)\n      : opts.transcodeProfile\n    // Intent is single-use: the recreated session's fresh controller now owns\n    // the tier state and writes a new intent on its next switch. Delete to\n    // prevent leaks across subsequent re-offers.\n    if (intent !== undefined && opts.sessionId !== undefined) {\n      this.adaptiveIntent.delete(opts.sessionId)\n    }\n\n    // Parse the brokerId so every session log gets `deviceId` /\n    // `camStreamId` tags. Without these tags, operator log filters\n    // scoped to a single device (the common debugging shape) drop\n    // every WebRTC session lifecycle line — and we end up flying\n    // blind on whether the session was created, on which codec path,\n    // and whether the H.265 SDP seed / first-RTP forward fired.\n    const slashIdx = brokerId.indexOf('/')\n    const deviceId = deviceIdFromBrokerId(brokerId)\n    const deviceTags = {\n      deviceId,\n      camStreamId: slashIdx > 0 ? brokerId.slice(slashIdx + 1) : brokerId,\n    }\n    const sessionLogger = this.logger.withTags(deviceTags)\n\n    // Wake the broker if it's still cold (RTSP on-demand pull hasn't dialed\n    // the camera yet) and wait briefly for the camera SDP. Without this, an\n    // on-demand session created against a cold broker reads\n    // `getStats().codec === undefined`, defaults to H.264, and locks every\n    // downstream decision (PT, codec offer list, repacketizer choice,\n    // SDP-seed dispatch) into the wrong codec — every H.265 source then\n    // goes black on the viewer (cam 99 freeze diagnosis 2026-05-28). The\n    // warm-up `onEncodedData` subscription triggers `checkDemand`→dial and\n    // stays alive until the real session subscribers attach below, so the\n    // broker can't see zero consumers and teardown mid-wake. Timeout\n    // bounded so we never block forever on an unreachable camera; on\n    // timeout we fall back to the prior default-H.264 behaviour rather\n    // than failing the session entirely.\n    let warmUpUnsubscribe: Unsubscribe | null = null\n    if (!broker.getStats().codec) {\n      warmUpUnsubscribe = broker.onEncodedData(() => { /* warm-up no-op */ }, {\n        kind: 'warmup',\n        media: 'both',\n      })\n      await new Promise<void>((resolve) => {\n        const timer = setTimeout(() => {\n          sessionLogger.warn('broker SDP wait timed out — falling back to default codec', {\n            meta: { brokerId, currentCodec: broker.getStats().codec ?? null, timeoutMs: 2500 },\n          })\n          resolve()\n        }, 2500)\n        const unsubSdp = broker.onSdpParameterSets(() => {\n          clearTimeout(timer)\n          unsubSdp()\n          resolve()\n        })\n      })\n      sessionLogger.info('broker warm-up complete', {\n        meta: { brokerId, detectedCodec: broker.getStats().codec ?? null },\n      })\n    }\n\n    const brokerCodec = broker.getStats().codec ?? 'h264'\n    const isHevc = brokerCodec === 'h265' || brokerCodec === 'hevc'\n    const sessionCodec = isHevc ? 'H265' : 'H264' as const\n    // RTP-bearing brokers expose source RTP via `onVideoRtp` — that's\n    // RTSP (pull) and `push-rtp` (provider-synthesised RTP). AnnexB-\n    // only push brokers (Reolink Baichuan today, Frigate event streams)\n    // and the future RTMP path have no source RTP and fall back to\n    // AnnexB → writeVideoNals even on H.265.\n    const sourceType = broker.getSourceType()\n    const isRtp = broker.isRtpSource()\n    // Forward source RTP through the codec's repacketizer for ALL RTP-bearing\n    // sources (RTSP), H.264 AND H.265 — instead of the AnnexB→writeVideoNals\n    // synthesis path. iOS (WebKit) has a strict H.264 depacketizer (like its\n    // HEVC one) that rejects the synthesized shape for Main/High passthrough;\n    // the repacketizer preserves the native RTP layout. AnnexB-only push\n    // brokers (Reolink Baichuan, Frigate) stay on writeVideoNals.\n    const useRtpRepacketizer = isRtp\n    // Surface the codec/path resolution in the message itself so it\n    // shows up in operator log filters that strip the structured meta\n    // — without this, \"WebRTC session created\" alone makes it\n    // impossible to tell whether a session went down the H.265 RTP\n    // repacketizer path (where the late-seed fix matters) or the\n    // AnnexB fallback (where it doesn't), which is the first\n    // discriminator any wake-up debugging needs.\n    sessionLogger.info(\n      `WebRTC session: codec=${sessionCodec} brokerCodec=${brokerCodec} sourceType=${sourceType ?? 'null'} isRtp=${isRtp} repacketizer=${useRtpRepacketizer}`,\n      { meta: { brokerId, sessionCodec, brokerCodec, sourceType, isRtp, useRtpRepacketizer } },\n    )\n\n    // Create push-based FrameSource from broker's encoded data\n    const { source, pushFrame, close: closeSource } = createPushFrameSource()\n\n    // Subscribe to broker's encoded packets. Both H.264 and H.265 use\n    // the same direct path — the session handles codec-specific RTP\n    // packetization (FU-A for H.264, FU for H.265).\n    //\n    // Parameter-set NALs (SPS/PPS for H.264, VPS/SPS/PPS for H.265)\n    // arrive as separate EncodedPackets from the broker. We buffer them\n    // and prepend to the next VCL NAL so the decoder gets a complete\n    // access unit.\n    const pendingParamNals: Buffer[] = []\n    /**\n     * Sticky SDP-derived VPS/SPS/PPS Annex-B blob. Cameras that publish\n     * parameter sets only in SDP (Reolink high-profile streams) emit\n     * raw IDR access units on the wire — without this sticky prepend,\n     * every keyframe after the first one would arrive at the WebRTC\n     * session as a bare IDR and Chrome's HEVC decoder never advances\n     * past zero.\n     *\n     * Captured the first time `pendingParamNals` is populated by a\n     * paramSet packet (the synthetic SDP-replay emitted by\n     * `StreamBroker.emitSdpParamSetsTo` when this subscriber registered).\n     * After that, every keyframe missing inline VPS/SPS/PPS gets the\n     * sticky blob prepended.\n     */\n    let stickyParamSets: Buffer | null = null\n    /**\n     * Whether we've seen a non-placeholder video packet on this\n     * session yet. The placeholder pump emits a 1280×720 SPS/PPS+IDR\n     * blob at 1Hz while the broker waits for real RTP. If those\n     * param sets land in `stickyParamSets`, the camera's first real\n     * keyframe (a different resolution) would get the placeholder\n     * SPS prepended and Chrome's decoder would freeze on the last\n     * placeholder frame until the session is recreated. Flipping\n     * `seenRealPacket` on the first real video packet flushes the\n     * placeholder-derived sticky state.\n     */\n    let seenRealPacket = false\n    // The real consumer is now attached — release the warm-up ghost\n    // subscription. `checkDemand` runs on add+remove; because we add\n    // BEFORE removing, the broker never sees zero consumers and won't\n    // teardown the dial mid-startup.\n    //\n    // Allow callers (WHEP / Alexa) to override the server-generated\n    // sessionId so the protocol response can echo the directive's\n    // sessionId verbatim. When the same sessionId arrives twice (Alexa\n    // retries InitiateSessionWithOffer with the SAME sessionId when the\n    // first Lambda relay didn't deliver the answer back to Echo in time),\n    // tear down the stale session first so the retry creates a fresh\n    // PeerConnection and Echo receives a usable answer instead of an\n    // \"already in use\" error. Resolved BEFORE the subscriptions are\n    // created so it can be stamped into the attribution — the\n    // `onConnected` hook re-tags subscriptions by this sessionId once ICE\n    // discovers the nominated remote address.\n    const requestedSessionId = opts.sessionId\n    if (requestedSessionId !== undefined && this.sessions.has(requestedSessionId)) {\n      const stale = this.sessions.get(requestedSessionId)!\n      sessionLogger.warn('Session id reused by client retry — closing stale session before re-creating', {\n        meta: { sessionId: requestedSessionId, brokerId: stale.brokerId },\n      })\n      this.cleanupSession(requestedSessionId)\n      await stale.session.close().catch(() => {})\n    }\n    const sessionId = requestedSessionId ?? crypto.randomUUID()\n    const baseAttribution: BrokerConsumerAttribution = {\n      ...(opts.consumerAttribution ?? { kind: 'webrtc-browser' }),\n      sessionId,\n    }\n    // Audio path always rides AnnexB; downstream codec depends on the\n    // source: when the broker codec is one Echo/HomeKit/browsers can\n    // copy (PCMU, Opus) we passthrough, otherwise libav transcodes.\n    // The session has the truthy view of the negotiated audio codec\n    // in the `audio frame in` log line — we mirror it best-effort here\n    // for the widget chip. Source audio codec is whatever the broker's\n    // `audioTrackInfo` advertised.\n    const sourceAudioCodec = broker.getStats().audio?.codec ?? null\n    const targetAudioCodec: string = sourceAudioCodec === 'PCMU' || sourceAudioCodec === 'PCMA' || sourceAudioCodec === 'OPUS'\n      ? sourceAudioCodec\n      : 'Pcmu'\n    const audioTranscoded = sourceAudioCodec !== null && targetAudioCodec.toLowerCase() !== sourceAudioCodec.toLowerCase()\n    // The AnnexB subscription serves audio always; for RTP-bearing sources\n    // (`useRtpRepacketizer === true`) the video branch early-returns and\n    // the actual video runs over the source-RTP fan-out below. Tagging\n    // each subscription with its consumed media keeps the widget honest:\n    // operators see ONE row per channel (\"audio\" / \"video\"), not two\n    // rows that look duplicated.\n    const annexbAttribution: BrokerConsumerAttribution = {\n      ...baseAttribution,\n      media: useRtpRepacketizer ? 'audio' : 'both',\n      // When media === 'both' (push-AnnexB source) the dominant channel\n      // chip is video: report the video codec as targetCodec for the\n      // widget. Audio detail is captured by the audio transport.\n      targetCodec: useRtpRepacketizer ? targetAudioCodec : sessionCodec,\n      transport: audioTranscoded ? 'transcode' : 'passthrough',\n    }\n    const rtpAttribution: BrokerConsumerAttribution = {\n      ...baseAttribution,\n      media: 'video',\n      targetCodec: sessionCodec,\n      // Source-RTP path through repacketizer preserves frames, only\n      // rebuilds headers — that's `'repacketize'`. Baseline transcode\n      // (set elsewhere when the cam ships Main/High to a Baseline-only\n      // viewer) flips this to `'transcode'`.\n      transport: effectiveTranscodeProfile !== undefined ? 'transcode' : 'repacketize',\n    }\n    const unsubBroker = broker.onEncodedData((packet: EncodedPacket) => {\n      if (packet.type === 'video') {\n        // RTSP+H.265 sessions forward source RTP directly through the\n        // repacketizer — the AnnexB → writeVideoNals path is\n        // bypassed entirely. The repacketizer is seeded at\n        // session-create time with the camera's VPS/SPS/PPS via\n        // `seedH265CodecInfoFromSdp`. Pushing placeholder frames\n        // through writeVideoNals here would re-configure Chrome's\n        // HEVC decoder with the placeholder's parameter sets\n        // (1280×720) and the transition back to the camera's real\n        // SPS through the repacketizer's AP codec packet doesn't\n        // recover cleanly — the viewer ends up frozen on the\n        // placeholder. So H.265 sessions show no labelled\n        // placeholder during wake-up; H.264 paths still get them.\n        if (useRtpRepacketizer) return\n\n        // First real packet after a placeholder window — drop any\n        // sticky state captured from placeholder param sets. From\n        // this point forward `pendingParamNals` / `stickyParamSets`\n        // track the live camera's actual SPS/PPS only.\n        if (!packet.isPlaceholder && !seenRealPacket) {\n          seenRealPacket = true\n          pendingParamNals.length = 0\n          stickyParamSets = null\n        }\n\n        const annexB = convertH264ToAnnexB(packet.data)\n\n        // Detect first NAL type\n        const nalTypeInfo = detectFirstNalType(annexB, isHevc)\n        if (nalTypeInfo.isParamSet) {\n          // Placeholder packets are self-contained access units —\n          // VPS+SPS+PPS+IDR concatenated in a single buffer. The\n          // first NAL is a parameter set, but the buffer also\n          // contains the IDR slice that produces the visible frame.\n          // `detectFirstNalType` only inspects the leading NAL, so\n          // the live-stream branch (which assumes \"param sets arrive\n          // separately, then a VCL\") would discard the IDR portion\n          // and the operator never sees the labelled placeholder.\n          // Push the whole bundle as a keyframe and skip the sticky-\n          // param caching (corrupts the camera's real keyframe later\n          // — different resolution/profile).\n          if (packet.isPlaceholder) {\n            pushFrame({\n              type: 'video',\n              frame: {\n                data: annexB,\n                codec: sessionCodec,\n                isKeyframe: true,\n                // Broker `EncodedPacket.pts` is wall-clock-relative MS\n                // by convention (see stream-broker.ts:2031 et al. and\n                // rtsp-restreamer.ts:246 — `pts * 90` to convert to\n                // 90kHz). Convert to µs for the frame contract.\n                timestampMicros: packet.pts * 1000,\n              },\n            })\n            return\n          }\n          pendingParamNals.push(Buffer.from(annexB))\n          // Keep a sticky copy so future keyframes that ship without\n          // inline param sets still get configured decoder.\n          stickyParamSets = Buffer.concat(pendingParamNals)\n          return\n        }\n\n        // VCL NAL — prepend any buffered parameter sets, OR the sticky\n        // SDP-derived blob if the keyframe lacks them inline.\n        let finalAnnexB = annexB\n        if (pendingParamNals.length > 0) {\n          pendingParamNals.push(annexB)\n          finalAnnexB = Buffer.concat(pendingParamNals)\n          pendingParamNals.length = 0\n        } else if (nalTypeInfo.isKeyframe && stickyParamSets) {\n          finalAnnexB = Buffer.concat([stickyParamSets, annexB])\n        }\n\n        pushFrame({\n          type: 'video',\n          frame: {\n            data: finalAnnexB,\n            codec: sessionCodec,\n            isKeyframe: packet.keyframe || nalTypeInfo.isKeyframe,\n            // Broker `EncodedPacket.pts` is MS-relative — convert to µs\n            // for the frame contract. See the placeholder-branch above.\n            timestampMicros: packet.pts * 1000,\n          },\n        })\n      } else if (packet.type === 'audio') {\n        pushFrame({\n          type: 'audio',\n          frame: {\n            data: packet.data,\n            codec: packet.codec?.toLowerCase() === 'pcma' ? 'Pcma' : 'Pcmu',\n            sampleRate: 8000,\n            channels: 1,\n            // Same MS → µs conversion as the video branch above.\n            timestampMicros: packet.pts * 1000,\n          },\n        })\n      }\n    }, annexbAttribution)\n    // Real consumer attached → release the warm-up ghost. The add-then-\n    // remove ordering keeps `checkDemand` from ever seeing zero\n    // subscribers, so the dial we just woke stays up.\n    warmUpUnsubscribe?.()\n\n    // Flush pre-buffer. Seed `preParamNals` with the SDP-derived\n    // VPS/SPS/PPS captured in the live callback's `stickyParamSets`\n    // closure variable above. Without this seed, pre-buffer flush\n    // starts with empty param sets and the first VCL flushed is a\n    // bare IDR — Chrome's HEVC decoder fails on cameras that ship\n    // param sets only via SDP (Reolink high-profile streams).\n    const preBuffer = broker.getPreBuffer()\n    const preParamNals: Buffer[] = stickyParamSets\n      ? [Buffer.from(stickyParamSets)]\n      : pendingParamNals.length > 0\n        ? pendingParamNals.map((b) => Buffer.from(b))\n        : []\n    for (const pkt of preBuffer) {\n      if (pkt.type === 'video') {\n        // RTSP+H.265 pre-buffer holds AnnexB packets — feeding them\n        // into the writeVideoNals path conflicts with the\n        // repacketizer's outbound RTP stream. Skip pre-buffer for\n        // video on the repacketizer path and rely on PLI to trigger\n        // a fresh keyframe from the camera. Push-mode H.265 keeps\n        // pre-buffer flush.\n        if (useRtpRepacketizer) continue\n\n        const annexB = convertH264ToAnnexB(pkt.data)\n        const nalTypeInfo = detectFirstNalType(annexB, isHevc)\n        if (nalTypeInfo.isParamSet) { preParamNals.push(Buffer.from(annexB)); continue }\n\n        let finalAnnexB = annexB\n        if (preParamNals.length > 0) {\n          preParamNals.push(annexB)\n          finalAnnexB = Buffer.concat(preParamNals)\n          preParamNals.length = 0\n        } else if (nalTypeInfo.isKeyframe && stickyParamSets) {\n          finalAnnexB = Buffer.concat([stickyParamSets, annexB])\n        }\n        pushFrame({\n          type: 'video',\n          frame: {\n            data: finalAnnexB,\n            codec: sessionCodec,\n            isKeyframe: pkt.keyframe || nalTypeInfo.isKeyframe,\n            // Pre-buffer pkt.pts is the same MS convention as the\n            // live `packet.pts` above — see the comment there.\n            timestampMicros: pkt.pts * 1000,\n          },\n        })\n      }\n    }\n\n    const allIceServers = await this.resolveIceServers()\n    // When the caller declares the peer is publicly reachable and TURN is\n    // counterproductive (Alexa's cloud media gateway), strip relay entries\n    // — keep only STUN. Filter URL-by-URL: a single IceServerEntry may\n    // expand into both `stun:` and `turn:` URLs sharing one credential\n    // record, and dropping the whole entry would discard the STUN URL too.\n    const iceServers: typeof allIceServers = opts.disableTurn === true\n      ? allIceServers\n          .map((entry) => {\n            const urls = Array.isArray(entry.urls) ? entry.urls : [entry.urls]\n            const stunOnly = urls.filter((u) => !u.startsWith('turn:') && !u.startsWith('turns:'))\n            if (stunOnly.length === 0) return null\n            return { ...entry, urls: stunOnly.length === 1 ? stunOnly[0]! : stunOnly }\n          })\n          .filter((e): e is NonNullable<typeof e> => e !== null)\n      : allIceServers\n\n    // iOS fix: an H.264 source emitting Main/High can't be decoded by iOS\n    // (WebKit) WebRTC, which negotiates Constrained Baseline and decodes\n    // ONLY Baseline. Detect the source profile from the camera SPS (SDP\n    // params first, then the pre-buffer) and re-encode the egress to\n    // Baseline when it isn't already. Baseline H.264 and all H.265 stay\n    // passthrough (zero re-encode). See\n    // docs/superpowers/specs/2026-05-26-webrtc-ios-baseline-transcode-design.md\n    const sourceProfileLevelId = sessionCodec === 'H264'\n      ? deriveH264ProfileLevelId(broker.getSdpParameterSets(), broker.getPreBuffer())\n      : undefined\n    // Client-offer flow: decide from the browser's advertised decode caps\n    // (iOS 18 offers H.264 High → 4K Main/High passes through). Server-offer\n    // flow: no client offer yet, so fall back to the conservative\n    // source-profile-only rule (transcode any Main/High to Baseline for iOS).\n    const offeredProfileLevelIds = opts.clientOfferSdp\n      ? extractOfferedH264ProfileLevelIds(opts.clientOfferSdp)\n      : undefined\n    const transcodeToBaseline = offeredProfileLevelIds\n      ? resolveClientOfferTranscodeToBaseline(sessionCodec, sourceProfileLevelId, offeredProfileLevelIds)\n      : resolveBrokerTranscodeToBaseline(sessionCodec, sourceProfileLevelId)\n    if (transcodeToBaseline) {\n      sessionLogger.info(\n        'WebRTC: source H.264 is non-Baseline and the client cannot decode it — re-encoding egress to Constrained Baseline',\n        { meta: { brokerId, sessionId, sourceProfileLevelId, offeredProfileLevelIds, clientOffer: opts.clientOfferSdp !== undefined } },\n      )\n    } else if (offeredProfileLevelIds && sourceProfileLevelId && !isBaselineProfileLevelId(sourceProfileLevelId)) {\n      sessionLogger.info(\n        'WebRTC: client offered a profile that decodes the non-Baseline H.264 source — passthrough (no re-encode)',\n        { meta: { brokerId, sessionId, sourceProfileLevelId, offeredProfileLevelIds } },\n      )\n    }\n\n    const transcodeDecodeHwAccel = await this.getTranscodeDecodeHwAccel()\n\n    const session = new AdaptiveSession({\n      sessionId,\n      source,\n      sourceCodec: sessionCodec,\n      transcodeToBaseline,\n      transcodeProfile: effectiveTranscodeProfile,\n      transcodeDecodeHwAccel,\n      iceConfig: {\n        iceServers,\n        portRange: this.icePortRange,\n        additionalHostAddresses: this.iceAdditionalHostAddresses,\n        disableIpv6: opts.disableIpv6 === true,\n      },\n      // Device-tagged logger so every WebRTC session lifecycle line\n      // (ICE state, candidate pairs, nominated pair, feed progress)\n      // carries `deviceId` — operator log filters scoped to one device\n      // no longer drop session logs as `dev=?`.\n      logger: sessionLogger,\n      deviceId,\n      // Source-RTP pre-buffer accessor for instant start on the repacketizer\n      // path — the session replays the current GOP on its first forwarded\n      // packet. Only meaningful for RTP sources (returns [] otherwise).\n      rtpBootstrap: useRtpRepacketizer ? (() => broker.getRtpPreBuffer()) : undefined,\n      debug: opts.streamingDebug === true,\n    })\n\n    // For RTSP (RTP-bearing) sessions, H.264 AND H.265: subscribe to\n    // source-RTP and forward through the codec's repacketizer in the\n    // session. Also seed the repacketizer codec info from the SDP-derived\n    // param sets (H.264 SPS/PPS, H.265 VPS/SPS/PPS) so the STAP-A/AP codec\n    // packet can precede the first IDR even when the camera doesn't emit\n    // param sets in-band. Push-mode brokers (Reolink Baichuan, etc.) skip\n    // this and use the AnnexB feed path.\n    const seedFromSdp = (ps: ReadonlyArray<Buffer>): void => {\n      if (sessionCodec === 'H265') session.seedH265CodecInfoFromSdp(ps)\n      else session.seedH264CodecInfoFromSdp(ps)\n    }\n    let unsubRtp: Unsubscribe | null = null\n    let unsubSdpParams: Unsubscribe | null = null\n    if (useRtpRepacketizer) {\n      const sdpPs = broker.getSdpParameterSets()\n      if (sdpPs && sdpPs.length > 0) {\n        seedFromSdp(sdpPs)\n        sessionLogger.info('RTP session seeded from SDP at create-time', {\n          meta: { sessionId, brokerId, codec: sessionCodec, psCount: sdpPs.length },\n        })\n      } else {\n        // Battery-cam wake-up: broker is still dialing rfc4571, so\n        // `getSdpParameterSets()` is null at this point. Subscribe so\n        // the seed lands the moment `onVideoTrack` fires — before\n        // that, the repacketizer can't emit its codec packet, so the\n        // browser receives RTP it can't initialise a decoder against.\n        // `onSdpParameterSets` delivers synchronously when params are\n        // already known (no-op for the wake-up case), and on every\n        // future reader rebuild.\n        sessionLogger.info('RTP session deferred: SDP params not ready, subscribing for late delivery', {\n          meta: { sessionId, brokerId, codec: sessionCodec },\n        })\n        unsubSdpParams = broker.onSdpParameterSets((ps) => {\n          try {\n            seedFromSdp(ps)\n            sessionLogger.info('RTP session seeded from SDP late delivery', {\n              meta: { sessionId, brokerId, codec: sessionCodec, psCount: ps.length },\n            })\n          } catch (err) {\n            sessionLogger.warn('seedCodecInfoFromSdp threw', {\n              meta: { sessionId, codec: sessionCodec, error: errMsg(err) },\n            })\n          }\n        })\n      }\n      let firstRtpForwarded = false\n      unsubRtp = broker.onVideoRtp((rtpData) => {\n        if (!firstRtpForwarded) {\n          firstRtpForwarded = true\n          sessionLogger.info('RTP session: first source RTP forwarded to repacketizer', {\n            meta: { sessionId, brokerId, codec: sessionCodec, bytes: rtpData.length },\n          })\n        }\n        try {\n          session.forwardSourceRtpVideo(rtpData)\n        } catch (err) {\n          sessionLogger.warn('forwardSourceRtpVideo threw', {\n            meta: { sessionId, codec: sessionCodec, error: errMsg(err) },\n          })\n        }\n      }, rtpAttribution)\n    }\n\n    const entry: SessionEntry = { session, brokerId, unsubBroker, unsubRtp, unsubSdpParams, closeSource }\n    this.sessions.set(sessionId, entry)\n\n    session.onClosed = () => this.cleanupSession(sessionId)\n\n    // Once ICE connects, read the nominated pair and tag this session's broker\n    // subscriptions (audio + video) with the remote peer address so operators\n    // can identify the viewer. Best-effort: werift may populate `nominated` up\n    // to ~1s after the \"connected\" event, so retry briefly; any failure is\n    // swallowed — this never affects the live session/ICE path. When the\n    // selected pair is a relay/srflx candidate the displayed `@addr` is the\n    // TURN/reflexive address (the relay's or the NAT-reflexive one), NOT the\n    // viewer's real LAN IP — appending the type disambiguates for the operator.\n    //\n    // Fire-at-most-once: the ICE `connected` transition also fires on every\n    // reconnect after a `disconnected`. The `attributionTagged` latch on the\n    // session entry ensures a reconnect does NOT re-arm a new retry chain; the\n    // pending `attributionTimer` is cleared on teardown (`cleanupSession`) so\n    // no timer fires after the session is gone.\n    session.onConnected = () => {\n      if (entry.attributionTagged) return\n      let attempt = 0\n      const tag = (): void => {\n        // Bail if the session was torn down while a retry was pending — the\n        // entry is removed from `this.sessions` in `cleanupSession`. Belt-and-\n        // suspenders alongside the timer-clear there.\n        if (this.sessions.get(sessionId) !== entry) return\n        try {\n          const pair = session.readNominatedPair()\n          if (!pair) {\n            if (attempt++ < 5) {\n              entry.attributionTimer = setTimeout(tag, 200)\n            } else {\n              // Retry budget exhausted — latch so a later reconnect does not\n              // re-arm a fresh chain for the same (likely unreadable) pair.\n              entry.attributionTagged = true\n            }\n            return\n          }\n          const remoteAddr = pair.remoteType === 'host'\n            ? pair.remoteAddr\n            : `${pair.remoteAddr} (${pair.remoteType})`\n          broker.updateConsumerAttributionBySession(sessionId, { remoteAddr })\n          entry.attributionTagged = true\n        } catch {\n          /* best-effort — never break the session path */\n        }\n      }\n      tag()\n    }\n\n    // Adaptive sessions (created via the `${deviceId}/adaptive` alias) get a\n    // ladder controller that can switch the session to a different-tier broker\n    // mid-stream. Non-adaptive sessions (pinned profile / cam-stream targets)\n    // are left UNCHANGED — they keep the single broker they were created on.\n    if (streamId.endsWith('/adaptive')) {\n      const initialTier: Tier = this.profileTierResolver?.(brokerId) ?? 'high'\n      this.startAdaptive(entry, deviceId, initialTier, sessionId)\n    }\n\n    return {\n      sessionId,\n      session,\n      sessionLogger,\n      deviceId,\n      brokerId,\n      sessionCodec,\n      useRtpRepacketizer,\n      iceServerCount: iceServers.length,\n    }\n  }\n\n  async handleAnswer(sessionId: string, sdpAnswer: string): Promise<void> {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) throw new Error(`Session not found: ${sessionId}`)\n    await entry.session.handleAnswer({ sdp: sdpAnswer, type: 'answer' })\n  }\n\n  /** Trickle ICE — add a remote (client) candidate to a live session. */\n  async addIceCandidate(sessionId: string, candidate: IceCandidatePayload): Promise<void> {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    await entry.session.addIceCandidate(candidate).catch((err) => {\n      this.logger.warn('addIceCandidate threw', { meta: { sessionId, error: errMsg(err) } })\n    })\n  }\n\n  /** Trickle ICE — the server's gathered candidates for a session (poll). */\n  getIceCandidates(sessionId: string): { candidates: IceCandidatePayload[]; done: boolean } {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return { candidates: [], done: true }\n    return entry.session.getIceCandidatesSnapshot()\n  }\n\n  async closeSession(sessionId: string): Promise<void> {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.cleanupSession(sessionId)\n    await entry.session.close().catch(() => {})\n    this.logger.info('WebRTC session closed', {\n      meta: { sessionId, deviceId: deviceIdFromBrokerId(entry.brokerId) },\n    })\n  }\n\n  async stop(): Promise<void> {\n    if (this.stopped) return\n    this.stopped = true\n    const closes: Promise<void>[] = []\n    for (const [sid, entry] of this.sessions) {\n      this.cleanupSession(sid)\n      closes.push(entry.session.close().catch(() => {}))\n    }\n    await Promise.all(closes)\n    this.brokers.clear()\n    this.logger.info('WebRTC server stopped')\n  }\n\n  getSessionCount(): number {\n    return this.sessions.size\n  }\n\n  /** Live signaling state for a session — used by the webrtc-session\n   *  provider's getSessionState. Returns a null pending when unknown. */\n  getSessionRenegotiationState(sessionId: string): { pendingRenegotiation: { target: WebrtcStreamTarget; epoch: number } | null } {\n    const entry = this.sessions.get(sessionId)\n    return { pendingRenegotiation: entry?.session.pendingRenegotiation ?? null }\n  }\n\n  // ── Private ─────────────────────────────────────────────────────────\n\n  /**\n   * Wire an {@link AdaptiveController} to a freshly-created adaptive session.\n   * The controller is fed loss from two sources — RTCP Receiver Reports\n   * (`onReceiverReport`) and, when injected, the device's client-reported\n   * packet-loss % polled every {@link ADAPTIVE_TICK_MS} — and drives the\n   * ladder via the timer's `tick`. Side effects (tier switch / MTU reduce /\n   * transcode-down) run through the controller's callbacks back into this\n   * server.\n   *\n   * The client-loss % is turned into a monotonic {@link LossSample} LOCALLY:\n   * a per-controller counter `clientLossN` advances each tick that carries a\n   * positive loss %, so consecutive samples have a clean delta whose ratio\n   * equals `pct/100` — exactly what the controller's `LossTracker` expects.\n   * Both paths feed the same controller; the tracker naturally honours\n   * whichever produces the higher loss ratio across its window. The async\n   * poll is guarded against overlapping ticks via `tickInFlight`.\n   */\n  private startAdaptive(entry: SessionEntry, deviceId: number, initialTier: Tier, sessionId: string): void {\n    const controller = new AdaptiveController({\n      initialTier,\n      config: ADAPTIVE_CONFIG,\n      onSwitchTier: (tier) => { this.switchSessionTier(entry, deviceId, tier, sessionId) },\n      onReduceMtu: () => entry.session.setRepacketizerMtu(AdaptiveSession.ADAPTIVE_LOW_MTU),\n      onTranscodeDown: () => { this.switchSessionTier(entry, deviceId, 'low', sessionId, true) },\n    })\n    entry.session.onReceiverReport((s) => controller.onLoss(s, Date.now()))\n    entry.adaptiveController = controller\n\n    let clientLossN = 0\n    let tickInFlight = false\n    const CLIENT_LOSS_SPAN = 1000\n    entry.adaptiveTimer = setInterval(() => {\n      if (tickInFlight) return\n      tickInFlight = true\n      void (async () => {\n        try {\n          const pct = this.clientLossPct ? await this.clientLossPct(deviceId) : null\n          const now = Date.now()\n          if (pct !== null && pct !== undefined && pct > 0) {\n            clientLossN += 1\n            controller.onLoss({\n              packetsLost: Math.round((pct / 100) * CLIENT_LOSS_SPAN * clientLossN),\n              highestSequence: CLIENT_LOSS_SPAN * clientLossN,\n            }, now)\n          }\n          controller.tick(now)\n        } finally {\n          tickInFlight = false\n        }\n      })()\n    }, ADAPTIVE_TICK_MS)\n  }\n\n  /**\n   * Schedule an adaptive tier switch via a close-and-recreate re-offer.\n   *\n   * Records the controller's desired tier (+ forced transcode-down) in\n   * `adaptiveIntent`, then sets `pendingRenegotiation` with an ADAPTIVE\n   * target so the client re-offers under the same sessionId. The re-offer\n   * runs through `setupSessionForBroker`, which tears the old session down\n   * completely and rebinds whichever feed type the session uses — so we no\n   * longer do a live broker rebind here (that produced a dual-feed for the\n   * AnnexB path and couldn't recreate the controller). `setupSessionForBroker`\n   * consumes the recorded intent to pin the recreated session to `tier` and\n   * apply the reduced transcode profile when `forceTranscodeDown` is set.\n   */\n  private switchSessionTier(entry: SessionEntry, deviceId: number, tier: Tier, sessionId: string, forceTranscodeDown = false): void {\n    const prev = this.adaptiveIntent.get(sessionId)\n    this.adaptiveIntent.set(sessionId, {\n      prefersTier: tier,\n      forceTranscodeDown: forceTranscodeDown || (prev?.forceTranscodeDown ?? false),\n    })\n    const epoch = (entry.session.pendingRenegotiation?.epoch ?? 0) + 1\n    entry.session.pendingRenegotiation = { target: { kind: 'adaptive' }, epoch }\n    this.logger.info('adaptive: scheduling renegotiation', { meta: { sessionId, deviceId, tier, forceTranscodeDown, epoch } })\n  }\n\n  private cleanupSession(sessionId: string): void {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.sessions.delete(sessionId)\n    // Clear any pending adaptive intent as a leak safety. On the close-and-\n    // recreate re-offer the intent is already consumed (read + deleted at the\n    // top of `setupSessionForBroker`) before this runs for the stale session,\n    // so this is a no-op there; it only catches the permanent-teardown case\n    // where a switch was scheduled but the client disconnected before\n    // re-offering.\n    this.adaptiveIntent.delete(sessionId)\n    if (entry.adaptiveTimer) clearInterval(entry.adaptiveTimer)\n    // Clear any pending remote-address tagging retry so no timer fires after\n    // teardown (and latch it so a stale fire is a guaranteed no-op).\n    if (entry.attributionTimer) clearTimeout(entry.attributionTimer)\n    entry.attributionTagged = true\n    entry.unsubBroker()\n    entry.unsubRtp?.()\n    entry.unsubSdpParams?.()\n    entry.closeSource()\n  }\n\n  private resolveBrokerId(streamId: string, hints: ClientHints = {}): string {\n    // Direct match — `webrtc-session.createSession` resolves the\n    // discriminated `WebrtcStreamTarget` to a canonical brokerId\n    // (`${deviceId}/${camStreamId}`) before invoking the server, so\n    // every legitimate client request hits this branch.\n    if (this.brokers.has(streamId)) return streamId\n\n    // Adaptive fallthrough — `${deviceId}/adaptive` is a sentinel\n    // that means \"pick the best broker for the device\". Bare\n    // deviceKey (no slash) is treated the same way for legacy\n    // callers that only know the device id.\n    const slash = streamId.indexOf('/')\n    if (slash > 0) {\n      const tail = streamId.slice(slash + 1)\n      if (tail === 'adaptive') {\n        return this.selectBestBroker(streamId.slice(0, slash), hints)\n      }\n    } else {\n      return this.selectBestBroker(streamId, hints)\n    }\n    // Unknown shape — let the lookup fail in the caller so the\n    // 'No broker for stream' message points at the actual input.\n    return streamId\n  }\n\n  /**\n   * Adaptive source selection: score registered brokers for a device\n   * against client hints (viewport, downlink). Returns the broker ID\n   * with the best fit. Falls back to tier preference when no hints.\n   */\n  private selectBestBroker(deviceKey: string, hints: ClientHints): string {\n    // Collect all brokers for this device, attaching the resolver-\n    // derived tier so candidates can be ranked irrespective of brokerId\n    // shape (post-refactor brokerId carries the camStreamId, not the\n    // tier — tiers are looked up in the assignments map at request\n    // time via `profileTierResolver`).\n    type Candidate = {\n      brokerId: string\n      tier: CamProfile | null\n      broker: IStreamBroker\n    }\n    const candidates: Candidate[] = []\n    for (const [bid, broker] of this.brokers) {\n      if (!bid.startsWith(`${deviceKey}/`)) continue\n      const tier = this.profileTierResolver?.(bid) ?? null\n      candidates.push({ brokerId: bid, tier, broker })\n    }\n    if (candidates.length === 0) return `${deviceKey}/high` // will fail at broker lookup\n    if (candidates.length === 1) return candidates[0]!.brokerId\n\n    // Explicit tier override from client\n    if (hints.prefersTier === 'high' || hints.prefersTier === 'mid' || hints.prefersTier === 'low') {\n      const explicit = candidates.find((c) => c.tier === hints.prefersTier)\n      if (explicit) return explicit.brokerId\n    }\n\n    // No hints — use tier preference order\n    const hasHints = hints.viewportWidth !== undefined\n      || hints.viewportHeight !== undefined\n      || hints.downlinkMbps !== undefined\n    if (!hasHints) {\n      for (const tier of TIER_PREFERENCE) {\n        const c = candidates.find((c) => c.tier === tier)\n        if (c) return c.brokerId\n      }\n      return candidates[0]!.brokerId\n    }\n\n    // Score each candidate\n    let bestId = candidates[0]!.brokerId\n    let bestTier: CamProfile | null = candidates[0]!.tier\n    let bestScore = Infinity\n    for (const c of candidates) {\n      const score = scoreBroker(c.tier, c.broker, hints)\n      const cRank = c.tier ? TIER_PREFERENCE.indexOf(c.tier) : TIER_PREFERENCE.length\n      const bestRank = bestTier ? TIER_PREFERENCE.indexOf(bestTier) : TIER_PREFERENCE.length\n      if (score < bestScore || (score === bestScore && cRank < bestRank)) {\n        bestId = c.brokerId\n        bestTier = c.tier\n        bestScore = score\n      }\n    }\n\n    this.logger.debug('Adaptive source selected', {\n      meta: { deviceKey, selected: bestId, tier: bestTier, score: Math.round(bestScore * 100) / 100, hints },\n    })\n    return bestId\n  }\n\n  private async resolveIceServers(): Promise<readonly IceServerEntry[]> {\n    if (this.getIceServersFn) {\n      try { return await this.getIceServersFn() } catch (err) {\n        this.logger.warn('getIceServers failed', { meta: { error: errMsg(err) } })\n      }\n    }\n    return this.staticIceServers ?? []\n  }\n}\n\n// ---------------------------------------------------------------------------\n// Push-based FrameSource factory\n// ---------------------------------------------------------------------------\n// NAL type detection (codec-agnostic)\n// ---------------------------------------------------------------------------\n\n/**\n * Parse the numeric deviceId out of a canonical brokerId\n * (`${deviceId}/${camStreamId}`). Returns -1 for the bare-deviceKey\n * shape with no slash. Used so every session/broker log line can carry\n * `deviceId` (operator log filters scoped to one device otherwise drop\n * them as `dev=?`).\n */\nfunction deviceIdFromBrokerId(brokerId: string): number {\n  const slashIdx = brokerId.indexOf('/')\n  if (slashIdx <= 0) return -1\n  const parsed = Number.parseInt(brokerId.slice(0, slashIdx), 10)\n  return Number.isNaN(parsed) ? -1 : parsed\n}\n\ninterface NalTypeInfo {\n  readonly isParamSet: boolean\n  readonly isKeyframe: boolean\n}\n\n/**\n * Detect the type of the first NAL in an Annex-B buffer.\n * Works for both H.264 (1-byte NAL header) and H.265 (2-byte NAL header).\n */\nfunction detectFirstNalType(annexB: Buffer, isHevc: boolean): NalTypeInfo {\n  for (let i = 0; i < annexB.length - 4; i++) {\n    if (annexB[i] === 0 && annexB[i + 1] === 0 && annexB[i + 2] === 0 && annexB[i + 3] === 1 && i + 4 < annexB.length) {\n      if (isHevc) {\n        const nalType = (annexB[i + 4]! & 0x7e) >> 1\n        const isParamSet = nalType === 32 || nalType === 33 || nalType === 34\n        const isKeyframe = isParamSet || (nalType >= 16 && nalType <= 21)\n        return { isParamSet, isKeyframe }\n      } else {\n        const nalType = annexB[i + 4]! & 0x1f\n        const isParamSet = nalType === 7 || nalType === 8\n        const isKeyframe = nalType === 5 || isParamSet\n        return { isParamSet, isKeyframe }\n      }\n    }\n  }\n  return { isParamSet: false, isKeyframe: false }\n}\n\n// ---------------------------------------------------------------------------\n// Push-based FrameSource factory\n// ---------------------------------------------------------------------------\n\nfunction createPushFrameSource(): {\n  source: FrameSource\n  pushFrame: (frame: MediaFrame) => void\n  close: () => void\n} {\n  const queue: MediaFrame[] = []\n  let resolve: ((value: IteratorResult<MediaFrame>) => void) | null = null\n  let done = false\n\n  const pushFrame = (mf: MediaFrame): void => {\n    if (done) return\n    if (resolve) {\n      const r = resolve\n      resolve = null\n      r({ value: mf, done: false })\n    } else {\n      queue.push(mf)\n      // Back-pressure: keep last 60 frames if queue overflows\n      if (queue.length > 120) queue.splice(0, queue.length - 60)\n    }\n  }\n\n  const close = (): void => {\n    done = true\n    if (resolve) {\n      const r = resolve\n      resolve = null\n      r({ value: undefined as never, done: true })\n    }\n  }\n\n  const source: FrameSource = (async function* () {\n    try {\n      while (true) {\n        const item = queue.shift()\n        if (item) { yield item; continue }\n        if (done) return\n        const result = await new Promise<IteratorResult<MediaFrame>>((r) => {\n          resolve = r\n        })\n        if (result.done) return\n        yield result.value\n      }\n    } finally {\n      done = true\n    }\n  })()\n\n  return { source, pushFrame, close }\n}\n\n// ---------------------------------------------------------------------------\n","import { makeSourceBrokerId } from '@camstack/types'\nimport type {\n  CameraStream, ProfileSlot, CamProfile,\n  RtspRestreamEntry, ProfileRtspEntry,\n  WebrtcClientHints, WebrtcStreamChoice, WebrtcStreamTarget,\n  EncodeProfile,\n  PickedCamStream, StreamCodec,\n  BrokerConsumerAttribution,\n} from '@camstack/types'\nimport type { StreamBrokerManager } from './stream-broker/stream-broker-manager.js'\nimport { withSourceResolution } from './encode-profile-resolver.js'\nimport { WEBRTC_FALLBACK_PROFILE } from './webrtc/webrtc-fallback-profile.js'\n\n// ── pickStream helpers ───────────────────────────────────────────────\n\n/** Canonicalise a stream codec string against the cap enum. */\nfunction canonicalCodec(raw: string | undefined): StreamCodec | null {\n  if (!raw) return null\n  const lower = raw.toLowerCase()\n  if (lower === 'hevc') return 'hevc'\n  if (lower === 'h265') return 'h265'\n  if (lower === 'h264') return 'h264'\n  if (lower === 'av1') return 'av1'\n  if (lower === 'mjpeg') return 'mjpeg'\n  if (lower === 'vp8') return 'vp8'\n  if (lower === 'vp9') return 'vp9'\n  return null\n}\n\n/** Two codecs match if they normalise to the same value OR are h265↔hevc aliases. */\nfunction codecMatches(streamCodec: string | undefined, accept: readonly StreamCodec[]): boolean {\n  const c = canonicalCodec(streamCodec)\n  if (!c) return false\n  if (accept.includes(c)) return true\n  // H.265 and HEVC are the same payload, naming varies by SDK / probe.\n  if ((c === 'h265' && accept.includes('hevc')) || (c === 'hevc' && accept.includes('h265'))) {\n    return true\n  }\n  return false\n}\n\nconst PROVIDER_PREFIX_DEFAULTS: readonly string[] = ['native:']\n\n/**\n * Device-scoped facade over the broker's cam-stream registry.\n * Implements `cameraStreamsCapability` (reads only — every mutation\n * lives on the system-scope `stream-broker` cap).\n */\nexport class CameraStreamsProvider {\n  constructor(private readonly manager: StreamBrokerManager) {}\n\n  async getCameraStreams(input: { deviceId: number }): Promise<readonly CameraStream[]> {\n    return this.manager.getCameraStreamsForDevice(input.deviceId)\n  }\n\n  async getBrokerStreams(input: { deviceId: number }): Promise<readonly ProfileSlot[]> {\n    return this.manager.getProfileSlotsForDevice(input.deviceId)\n  }\n\n  async getRtspEntries(\n    input: { deviceId: number; hostname?: string },\n  ): Promise<readonly RtspRestreamEntry[]> {\n    return this.manager.getRtspEntriesForDevice(input.deviceId, input.hostname)\n  }\n\n  async getProfileRtspEntries(\n    input: { deviceId: number; hostname?: string },\n  ): Promise<readonly ProfileRtspEntry[]> {\n    return this.manager.getProfileRtspEntriesForDevice(input.deviceId, input.hostname)\n  }\n\n  /**\n   * Pick the best source camStream for the caller's decode constraints.\n   * Returns `null` when nothing matches — the caller falls back to its\n   * existing path (derived-broker transcode, profile-slot pick, etc).\n   *\n   * Generalised from Alexa's `tryResolveDirectH264Target`: every broker\n   * consumer with codec restrictions (Echo decodes only H.264, HomeKit\n   * Secure Video prefers H.264 baseline, browsers without HEVC\n   * hardware…) needs the same \"is there a non-transcoded sibling\n   * stream I should dial?\" decision, so it lives on the cap.\n   */\n  async pickStream(input: {\n    deviceId: number\n    requirements: {\n      acceptCodecs?: readonly StreamCodec[]\n      minHeight?: number\n      minWidth?: number\n      excludeDerived?: boolean\n      requireSiblingCodec?: readonly StreamCodec[]\n    }\n    preferences?: {\n      preferredProviders?: readonly string[]\n      resolutionPreference?: 'highest' | 'lowest'\n    }\n  }): Promise<PickedCamStream | null> {\n    const streams = await this.manager.getCameraStreamsForDevice(input.deviceId)\n    if (streams.length === 0) {\n      this.manager.getLogger?.()?.info('camera-streams: pickStream — no streams for device', {\n        meta: { deviceId: input.deviceId, requirements: input.requirements },\n      })\n      return null\n    }\n\n    const requirements = input.requirements\n    const preferences = input.preferences\n    const excludeDerived = requirements.excludeDerived !== false\n    const accept = requirements.acceptCodecs ?? []\n    const minHeight = requirements.minHeight ?? 0\n    const minWidth = requirements.minWidth ?? 0\n\n    const candidates = streams.filter((s) => {\n      if (excludeDerived && s.camStreamId.startsWith('derived:')) return false\n      if (accept.length > 0 && !codecMatches(s.codec, accept)) return false\n      if ((s.resolution?.height ?? 0) < minHeight) return false\n      if ((s.resolution?.width ?? 0) < minWidth) return false\n      return true\n    })\n    if (candidates.length === 0) {\n      this.manager.getLogger?.()?.info('camera-streams: pickStream — no matching candidates', {\n        meta: {\n          deviceId: input.deviceId,\n          requirements: input.requirements,\n          streams: streams.map((s) => ({\n            camStreamId: s.camStreamId,\n            codec: s.codec,\n            resolution: s.resolution,\n          })),\n        },\n      })\n      return null\n    }\n\n    // Sibling-codec gate — only \"pick a bypass\" when the natural pick\n    // would have hit one of the rejected codecs. Skips the optimisation\n    // for devices that don't need it.\n    const requireSibling = requirements.requireSiblingCodec ?? []\n    if (requireSibling.length > 0) {\n      const hasSibling = streams.some((s) => {\n        if (s.camStreamId.startsWith('derived:')) return false\n        return codecMatches(s.codec, requireSibling)\n      })\n      if (!hasSibling) {\n        this.manager.getLogger?.()?.info('camera-streams: pickStream — no sibling codec, skipping bypass', {\n          meta: {\n            deviceId: input.deviceId,\n            requireSibling,\n            streams: streams.map((s) => ({\n              camStreamId: s.camStreamId,\n              codec: s.codec,\n            })),\n          },\n        })\n        return null\n      }\n    }\n\n    const preferredProviders = preferences?.preferredProviders ?? PROVIDER_PREFIX_DEFAULTS\n    const resolutionPreference = preferences?.resolutionPreference ?? 'highest'\n\n    // Provider rank: lower is better. Streams whose camStreamId starts\n    // with the earliest preferred prefix win. Unmatched streams sit at\n    // `preferredProviders.length` (last).\n    const providerRank = (camStreamId: string): number => {\n      for (let i = 0; i < preferredProviders.length; i += 1) {\n        if (camStreamId.startsWith(preferredProviders[i]!)) return i\n      }\n      return preferredProviders.length\n    }\n\n    candidates.sort((a, b) => {\n      const ra = providerRank(a.camStreamId)\n      const rb = providerRank(b.camStreamId)\n      if (ra !== rb) return ra - rb\n      const ha = a.resolution?.height ?? 0\n      const hb = b.resolution?.height ?? 0\n      return resolutionPreference === 'highest' ? hb - ha : ha - hb\n    })\n\n    const winner = candidates[0]!\n    const reasonParts: string[] = []\n    reasonParts.push(`codec=${winner.codec ?? 'unknown'}`)\n    if (winner.resolution) reasonParts.push(`res=${winner.resolution.width}x${winner.resolution.height}`)\n    reasonParts.push(`provider=${providerRank(winner.camStreamId)}`)\n    if (requireSibling.length > 0) reasonParts.push(`bypassed=${requireSibling.join('|')}`)\n\n    return {\n      camStreamId: winner.camStreamId,\n      codec: winner.codec,\n      resolution: winner.resolution,\n      reason: reasonParts.join(' '),\n    }\n  }\n}\n\n/**\n * Device-scoped WebRTC session facade.\n *\n * Implements `webrtcSessionCapability`. Resolves `{deviceId, profile}` to\n * a broker-integrated WebRTC session — no external webrtc addon needed.\n * The `BrokerWebrtcServer` subscribes directly to `IStreamBroker.onEncodedData()`\n * so frames flow Camera → Broker → RTP → WebRTC with zero re-encoding.\n */\nexport { BrokerWebrtcServer } from './webrtc/broker-webrtc-server.js'\nimport type { BrokerWebrtcServer } from './webrtc/broker-webrtc-server.js'\n\nexport class WebrtcSessionProvider {\n  constructor(\n    private readonly webrtcServer: BrokerWebrtcServer,\n    private readonly manager: StreamBrokerManager,\n  ) {}\n\n  private static readonly PROFILE_LABELS: Record<CamProfile, string> = {\n    high: 'High',\n    mid: 'Mid',\n    low: 'Low',\n  }\n\n  /**\n   * Per-device session debug flag passed to `BrokerWebrtcServer` (which\n   * forwards it into `AdaptiveSession.debug`). True when EITHER the\n   * \"Streaming debug logs\" toggle (`streamingDebug`) OR the dedicated\n   * \"Advanced WebRTC debug logging\" toggle (`webrtcDebug`) is on for the\n   * device. The latter is the operator switch that, when ON, turns on the\n   * full per-step WebRTC lifecycle logging in `session.ts` for every\n   * session created for this device. Read once at session creation so the\n   * next session for a device picks up a freshly-toggled flag.\n   */\n  private resolveSessionDebug(deviceId: number): boolean {\n    const streamingDebug = typeof this.manager.isStreamingDebug === 'function'\n      ? this.manager.isStreamingDebug(deviceId)\n      : false\n    const webrtcDebug = typeof this.manager.isWebrtcDebug === 'function'\n      ? this.manager.isWebrtcDebug(deviceId)\n      : false\n    return streamingDebug || webrtcDebug\n  }\n\n  /**\n   * Resolve a structured `WebrtcStreamTarget` to the brokerId the\n   * WebRTC server understands. `'adaptive'` yields the magic\n   * `${deviceId}/adaptive` sentinel that triggers `selectBestBroker`\n   * server-side; the other two kinds resolve to the canonical\n   * cam-stream-keyed brokerId.\n   */\n  private resolveTargetToStreamId(deviceId: number, target: WebrtcStreamTarget): string {\n    switch (target.kind) {\n      case 'adaptive':\n        return `${deviceId}/adaptive`\n      case 'profile': {\n        const slots = this.manager.getProfileSlotsForDevice(deviceId)\n        const slot = slots.find((s) => s.profile === target.profile)\n        if (!slot?.sourceCamStreamId) {\n          throw new Error(`webrtc-session: profile \"${target.profile}\" has no source assigned for device ${deviceId}`)\n        }\n        return makeSourceBrokerId(deviceId, slot.sourceCamStreamId)\n      }\n      case 'cam-stream':\n        return makeSourceBrokerId(deviceId, target.camStreamId)\n    }\n  }\n\n  /**\n   * Resolve a `WebrtcStreamTarget` to its concrete camStreamId, or\n   * `undefined` for adaptive (no single cam stream is pinned).\n   */\n  private resolveCamStreamId(deviceId: number, target: WebrtcStreamTarget): string | undefined {\n    switch (target.kind) {\n      case 'adaptive':\n        return undefined\n      case 'cam-stream':\n        return target.camStreamId\n      case 'profile': {\n        const slots = this.manager.getProfileSlotsForDevice(deviceId)\n        return slots.find((s) => s.profile === target.profile)?.sourceCamStreamId ?? undefined\n      }\n    }\n  }\n\n  /**\n   * Resolve the `EncodeProfile` to use for a session.\n   *\n   *   1. The hardcoded WebRTC fallback profile applies, smart-adapted to\n   *      the source's probed resolution so a 4K H.265 camera served to\n   *      a no-HW-HEVC browser stays 4K H.264 instead of dropping to\n   *      the baked 1280×720.\n   *   2. Adaptive targets have no concrete camStreamId and always fall\n   *      back to the raw fallback profile.\n   */\n  private resolveTranscodeProfile(deviceId: number, camStreamId: string | undefined): EncodeProfile {\n    if (camStreamId === undefined) return WEBRTC_FALLBACK_PROFILE\n    const probedSource = this.manager.getCameraStreamProbedParams(deviceId, camStreamId)\n    return withSourceResolution(WEBRTC_FALLBACK_PROFILE, probedSource)\n  }\n\n  async listStreams(input: { deviceId: number }): Promise<readonly WebrtcStreamChoice[]> {\n    const { deviceId } = input\n    const slots = this.manager.getProfileSlotsForDevice(deviceId)\n    const assignedSlots = slots.filter((s) => s.sourceCamStreamId !== null)\n    const camStreams = this.manager.getCameraStreamsForDevice(deviceId)\n    const camStreamById = new Map<string, CameraStream>()\n    for (const c of camStreams) camStreamById.set(c.camStreamId, c)\n\n    const choices: WebrtcStreamChoice[] = []\n\n    if (assignedSlots.length > 0) {\n      choices.push({\n        id: 'adaptive',\n        label: 'Adaptive',\n        target: { kind: 'adaptive' },\n        codec: null,\n        resolution: null,\n        status: null,\n        inputFps: null,\n        decodeFps: null,\n        bitrateKbps: null,\n      })\n    }\n\n    // Profile-tier choices: pin a session to high / mid / low. Source\n    // can change as the operator reassigns the slot — the session\n    // follows.\n    for (const slot of assignedSlots) {\n      const profile = slot.profile\n      const camStreamId = slot.sourceCamStreamId\n      if (!camStreamId) continue\n      const broker = await this.manager.getBroker({ brokerId: makeSourceBrokerId(deviceId, camStreamId) })\n      const stats = broker?.getStats()\n      const cam = camStreamById.get(camStreamId)\n      choices.push({\n        id: `profile:${profile}`,\n        label: WebrtcSessionProvider.PROFILE_LABELS[profile] ?? profile,\n        target: { kind: 'profile', profile },\n        codec: stats?.codec ?? cam?.codec ?? slot.codec ?? null,\n        resolution: cam?.resolution ?? slot.resolution ?? null,\n        status: stats?.status ?? null,\n        inputFps: stats?.inputFps ?? null,\n        decodeFps: stats?.decodeFps ?? null,\n        bitrateKbps: stats?.bitrateKbps ?? null,\n      })\n    }\n\n    // Cam-stream-direct choices: enumerate EVERY published camStream.\n    // Each published stream has a live broker (broker lifecycle is 1:1\n    // with `publishCameraStream` / `retractCameraStream`).\n    for (const cam of camStreams) {\n      const camStreamId = cam.camStreamId\n      const broker = await this.manager.getBroker({ brokerId: makeSourceBrokerId(deviceId, camStreamId) })\n      const stats = broker?.getStats()\n      choices.push({\n        id: `stream:${camStreamId}`,\n        label: cam.label ?? camStreamId,\n        target: { kind: 'cam-stream', camStreamId },\n        codec: stats?.codec ?? cam.codec ?? null,\n        resolution: cam.resolution ?? null,\n        status: stats?.status ?? null,\n        inputFps: stats?.inputFps ?? null,\n        decodeFps: stats?.decodeFps ?? null,\n        bitrateKbps: stats?.bitrateKbps ?? null,\n      })\n    }\n    return choices\n  }\n\n  async createSession(input: {\n    deviceId: number\n    target: WebrtcStreamTarget\n    hints?: WebrtcClientHints\n    /**\n     * SERVER-INJECTED relay-only flag (set by the hub when the viewer's\n     * IP is non-LAN). Forces TURN-relay-only ICE for this session so a\n     * CGNAT/4G client gets a clean relay↔relay media path. LAN viewers\n     * leave this absent and keep the direct host/srflx path. See the\n     * cap definition (`webrtc-session.cap.ts`) for the rationale.\n     */\n    relayOnly?: boolean\n    /**\n     * Subscriber attribution surfaced in the stream-broker widget's\n     * client list. Hub-router sets `remoteAddr` from the tRPC request\n     * context; callers (Alexa addon, browser hooks) provide `kind`,\n     * `label`, `userId`, and `sessionId` when known.\n     */\n    consumerAttribution?: BrokerConsumerAttribution\n  }): Promise<{ sessionId: string; sdpOffer: string }> {\n    const streamId = this.resolveTargetToStreamId(input.deviceId, input.target)\n    const streamingDebug = this.resolveSessionDebug(input.deviceId)\n    const camStreamId = this.resolveCamStreamId(input.deviceId, input.target)\n    const transcodeProfile = this.resolveTranscodeProfile(input.deviceId, camStreamId)\n    return this.webrtcServer.createSession(streamId, input.hints, {\n      streamingDebug,\n      relayOnly: input.relayOnly === true,\n      transcodeProfile,\n      ...(input.consumerAttribution !== undefined ? { consumerAttribution: input.consumerAttribution } : {}),\n    })\n  }\n\n  async handleOffer(input: {\n    deviceId: number\n    target?: WebrtcStreamTarget\n    sdpOffer: string\n    sessionId?: string\n    /**\n     * Relay-only flag. A cloud peer (Alexa's RTCSessionController) sets\n     * this `true` — it's only reachable via TURN, never our host/srflx.\n     * A browser doing client-offer passthrough leaves it absent/false so\n     * a direct LAN/Tailscale host pair carries full native quality. See\n     * the cap definition (`webrtc-session.cap.ts`) for the rationale.\n     */\n    relayOnly?: boolean\n    /**\n     * Non-trickle flag. A peer that doesn't support trickle ICE (Alexa's\n     * RTCSessionController) sets this `true` so the answer SDP carries the\n     * server's gathered candidates instead of trickling them separately. A\n     * browser leaves it absent/false and polls `getIceCandidates`. See the\n     * cap definition (`webrtc-session.cap.ts`) for the rationale.\n     */\n    nonTrickle?: boolean\n    /**\n     * Disable-TURN flag. Strips TURN servers from the per-session iceConfig\n     * so the answer carries ONLY host + srflx candidates. See the cap\n     * definition (`webrtc-session.cap.ts`) for the rationale.\n     */\n    disableTurn?: boolean\n    /**\n     * Disable-IPv6 flag. Suppresses IPv6 host candidates from werift\n     * gathering and from `additionalHostAddresses`. Set `true` by peers\n     * that document IPv4-only ICE (Alexa). See the cap definition.\n     */\n    disableIpv6?: boolean\n    /** Subscriber attribution. See `createSession` for the contract. */\n    consumerAttribution?: BrokerConsumerAttribution\n  }): Promise<{ sessionId: string; sdpAnswer: string }> {\n    // Default to adaptive: client-offer callers (Alexa, WHEP) don't\n    // know about our profile slot model; they just want \"give me a\n    // working video session for this device\". The same adaptive\n    // selection logic the UI uses applies cleanly.\n    const target: WebrtcStreamTarget = input.target ?? { kind: 'adaptive' }\n    const streamId = this.resolveTargetToStreamId(input.deviceId, target)\n    const streamingDebug = this.resolveSessionDebug(input.deviceId)\n    const camStreamId = this.resolveCamStreamId(input.deviceId, target)\n    const transcodeProfile = this.resolveTranscodeProfile(input.deviceId, camStreamId)\n    return this.webrtcServer.handleOffer(streamId, input.sdpOffer, undefined, {\n      streamingDebug,\n      sessionId: input.sessionId,\n      relayOnly: input.relayOnly === true,\n      nonTrickle: input.nonTrickle === true,\n      disableTurn: input.disableTurn === true,\n      disableIpv6: input.disableIpv6 === true,\n      transcodeProfile,\n      ...(input.consumerAttribution !== undefined ? { consumerAttribution: input.consumerAttribution } : {}),\n    })\n  }\n\n  async handleAnswer(input: {\n    deviceId: number\n    sessionId: string\n    sdpAnswer: string\n  }): Promise<void> {\n    await this.webrtcServer.handleAnswer(input.sessionId, input.sdpAnswer)\n  }\n\n  async addIceCandidate(input: {\n    deviceId: number\n    sessionId: string\n    candidate: string\n    sdpMid?: string | null\n    sdpMLineIndex?: number | null\n  }): Promise<void> {\n    await this.webrtcServer.addIceCandidate(input.sessionId, {\n      candidate: input.candidate,\n      sdpMid: input.sdpMid ?? null,\n      sdpMLineIndex: input.sdpMLineIndex ?? null,\n    })\n  }\n\n  async getIceCandidates(input: { deviceId: number; sessionId: string }): Promise<{\n    candidates: { candidate: string; sdpMid: string | null; sdpMLineIndex: number | null }[]\n    done: boolean\n  }> {\n    return this.webrtcServer.getIceCandidates(input.sessionId)\n  }\n\n  async closeSession(input: { deviceId: number; sessionId: string }): Promise<void> {\n    await this.webrtcServer.closeSession(input.sessionId)\n  }\n\n  async hasAdaptiveBitrate(_input: {\n    deviceId: number\n    profile: CamProfile\n  }): Promise<boolean> {\n    // Broker-direct mode: no adaptive bitrate switching (single source per profile).\n    // The broker serves the camera's native stream at its native bitrate.\n    return false\n  }\n\n  async getSessionState(input: { deviceId: number; sessionId: string }): Promise<{ pendingRenegotiation: { target: WebrtcStreamTarget; epoch: number } | null }> {\n    return this.webrtcServer.getSessionRenegotiationState(input.sessionId)\n  }\n}\n","import type {\n  ProviderRegistration,\n} from '@camstack/types'\nimport {\n  BaseAddon, asJsonObject, EventCategory, createEvent, streamBrokerCapability,\n  cameraStreamsCapability, webrtcSessionCapability, errMsg,\n  addonWidgetsSourceCapability,\n  type IAddonWidgetsSourceProvider,\n} from '@camstack/types'\nimport {\n  StreamBrokerManager, type ProfileMapPersister,\n  type DeviceOverride, DeviceOverridesMapSchema, type DeviceOverridesMap,\n  RtspTokensSchema, RtspEnabledSchema, ProfileMapSchema,\n  type RtspTokensBlob, type RtspEnabledBlob, type ProfileMapBlob,\n} from './stream-broker/stream-broker-manager'\nimport { CameraStreamsProvider, WebrtcSessionProvider } from './device-scoped-providers'\nimport { BrokerWebrtcServer } from './webrtc/broker-webrtc-server'\nimport type { FfmpegHwAccelSetting } from './stream-broker/ffmpeg-invocation'\nimport { hwAccelSettingToPrefer, pickDecodeHwAccel } from './stream-broker/ffmpeg-invocation'\nimport type { CamProfile } from '@camstack/types'\n\nconst FFMPEG_HW_ACCEL_VALUES: readonly FfmpegHwAccelSetting[] = [\n  'auto', 'none', 'videotoolbox', 'vaapi', 'qsv', 'cuda',\n]\nfunction isFfmpegHwAccel(v: unknown): v is FfmpegHwAccelSetting {\n  return typeof v === 'string' && (FFMPEG_HW_ACCEL_VALUES as readonly string[]).includes(v)\n}\n\ninterface StreamBrokerConfig {\n  readonly defaultPreBufferSec: number\n  readonly rtspPort: number\n  readonly maxDecodeFps: number\n  readonly initialReconnectDelayMs: number\n  readonly maxReconnectDelayMs: number\n  /**\n   * How often (seconds) the broker re-pulls every camera's `stream-catalog`\n   * to reconcile its cam-stream registry. The backstop for any missed\n   * device / stream-params event; primary signal is those events, so a\n   * generous cadence keeps the cost negligible.\n   */\n  readonly catalogReconcileIntervalSec: number\n}\n\ninterface PersistedAssignment {\n  readonly map: Partial<Record<CamProfile, string>>\n  readonly auto: boolean\n}\n\nconst PROFILE_KEYS: readonly CamProfile[] = ['high', 'mid', 'low']\n\n/**\n * Cadence for the per-broker metrics-snapshot bus event. ~1 Hz keeps\n * UI dashboards (StreamBrokerPanel, broker stats overlays) live without\n * needing to poll `getBrokerStats` every 3s — admin-ui subscribes to\n * the snapshot category and updates state from the event payload.\n */\nconst BROKER_METRICS_SNAPSHOT_INTERVAL_MS = 1_000\n\n/**\n * Build a dedup-key for a snapshot by serialising the payload with\n * monotonic counter fields stripped. Without this, fields like\n * `uptimeMs` / `totalBytes` / `packetCount` on broker stats — which\n * tick up every second regardless of state — make every JSON-equal\n * check fail and the defer never fires. Stripping happens in the\n * key only; the full payload still rides the bus when an emit\n * fires.\n */\nfunction stableSnapshotKey<T>(payload: T, dropKeys: readonly string[]): string {\n  const dropSet = new Set(dropKeys)\n  const strip = (v: unknown): unknown => {\n    if (Array.isArray(v)) return v.map(strip)\n    if (v && typeof v === 'object') {\n      const out: Record<string, unknown> = {}\n      for (const [k, val] of Object.entries(v as Record<string, unknown>)) {\n        if (dropSet.has(k)) continue\n        out[k] = strip(val)\n      }\n      return out\n    }\n    return v\n  }\n  return JSON.stringify(strip(payload))\n}\n/**\n * Force a broker-stats emit at least every 30s even when the\n * payload hasn't changed — gives consumers a heartbeat without\n * the per-tick spam an unconditional emit produces. Picks up\n * idle brokers (no encoded subscribers, status frozen at\n * `streaming` with stale fps/bitrate).\n */\nconst BROKER_METRICS_HEARTBEAT_MS = 30_000\n\nexport class StreamBrokerAddon extends BaseAddon<StreamBrokerConfig> {\n  private brokerManager: StreamBrokerManager | null = null\n  private metricsSnapshotTimer: ReturnType<typeof setInterval> | null = null\n  private catalogReconcileTimer: ReturnType<typeof setInterval> | null = null\n  /** Kept so onShutdown can close every live PeerConnection — releasing\n   *  the ICE UDP sockets. Without this, a graceful runner respawn (deploy)\n   *  leaks those sockets and wedges the next runner's werift ICE agent\n   *  (DTLS never connects until a full hub restart). */\n  private webrtcServer: BrokerWebrtcServer | null = null\n  /**\n   * Snapshot-equality cache for the broker-metrics emit. Each\n   * broker emits a stats snapshot every BROKER_METRICS_SNAPSHOT_-\n   * INTERVAL_MS; for an idle broker (status=streaming, fps=0,\n   * subscribers=0, bitrate frozen) the per-tick payload is\n   * bytewise-identical and floods the bus to no end. We skip the\n   * emit when the JSON-serialised payload matches the last one\n   * and reset the heartbeat clock so a quiet broker still emits\n   * every BROKER_METRICS_HEARTBEAT_MS.\n   *\n   * Keyed by broker.deviceId (the brokerId — `<deviceId>/<profile>`\n   * string). Cleared when the broker manager is destroyed.\n   */\n  private readonly lastEmittedBrokerSnapshot = new Map<string, { json: string; emittedAt: number }>()\n\n  constructor() {\n    super({\n      defaultPreBufferSec: 10,\n      rtspPort: 8554,\n      maxDecodeFps: 5,\n      initialReconnectDelayMs: 1000,\n      maxReconnectDelayMs: 30000,\n      catalogReconcileIntervalSec: 30,\n    })\n  }\n\n  /**\n   * Read the cluster `ffmpeg` config section (binaryPath / hwAccel /\n   * threadCount) and push it into the broker. Defensive: any missing or\n   * malformed field falls back to the safe default (PATH `ffmpeg`, `auto`\n   * hw-accel, `0` threads), and a read failure leaves the defaults in place.\n   */\n  private async loadFfmpegConfig(): Promise<void> {\n    if (!this.brokerManager) return\n    try {\n      const section = (await this.ctx.settings?.getSection('ffmpeg')) ?? {}\n      const binaryPathRaw = section['binaryPath']\n      const binaryPath =\n        typeof binaryPathRaw === 'string' && binaryPathRaw.trim().length > 0\n          ? binaryPathRaw\n          : 'ffmpeg'\n      const hwAccel = isFfmpegHwAccel(section['hwAccel']) ? section['hwAccel'] : 'auto'\n      const threadCountRaw = section['threadCount']\n      const threadCount =\n        typeof threadCountRaw === 'number' && Number.isFinite(threadCountRaw) && threadCountRaw >= 0\n          ? Math.floor(threadCountRaw)\n          : 0\n      this.brokerManager.setFfmpegConfig({ binaryPath, hwAccel, threadCount })\n      this.ctx.logger.info('Loaded ffmpeg config', { meta: { binaryPath, hwAccel, threadCount } })\n    } catch (err) {\n      this.ctx.logger.warn('Failed to read ffmpeg config section — using defaults', {\n        meta: { error: errMsg(err) },\n      })\n    }\n  }\n\n  /**\n   * Resolve the decode `-hwaccel` backend for the WebRTC transcode feed: the\n   * configured `ffmpeg.hwAccel` preference through `platformProbe.resolveHwAccel`,\n   * intersected with what the configured ffmpeg can actually decode\n   * (`getHardwareDecodeAccels`). `null` ⇒ software decode. Mirrors the transcode\n   * egress's gate; the feed itself falls back to software on a per-stream miss.\n   */\n  private async resolveTranscodeDecodeHwAccel(): Promise<string | null> {\n    try {\n      const section = (await this.ctx.settings?.getSection('ffmpeg')) ?? {}\n      const hwAccel = isFfmpegHwAccel(section['hwAccel']) ? section['hwAccel'] : 'auto'\n      const prefer = hwAccelSettingToPrefer(hwAccel)\n      const res = await this.ctx.api.platformProbe.resolveHwAccel.query({ prefer })\n      const supported = await this.ctx.api.platformProbe.getHardwareDecodeAccels.query()\n      return pickDecodeHwAccel(res.preferred, supported.methods)\n    } catch (err) {\n      this.ctx.logger.warn('resolveTranscodeDecodeHwAccel failed — software decode', {\n        meta: { error: errMsg(err) },\n      })\n      return null\n    }\n  }\n\n  protected async onInitialize(): Promise<ProviderRegistration[]> {\n    this.brokerManager = new StreamBrokerManager(undefined, this.ctx.logger)\n    this.brokerManager.setDefaultPreBufferSec(this.config.defaultPreBufferSec)\n    this.brokerManager.setEventBus(this.ctx.eventBus)\n    // Wire the tRPC api proxy so the broker can reach `decoder` (and\n    // any future cross-cap consumer) via `localProviderLink` /\n    // `brokerTransportLink` regardless of where the decoder addon\n    // physically runs. Replaces the previous direct `CapabilitiesAccess`\n    // lookup which only worked on the hub.\n    this.brokerManager.setApiAccess(this.ctx.api)\n    if (this.capabilities) {\n      this.brokerManager.setCapabilitiesAccess(this.capabilities)\n    }\n    // Load the system `ffmpeg` config (binaryPath / hwAccel / threadCount) so\n    // the transcode egress spawns the configured binary with the operator's\n    // hw-accel preference. Defaults (PATH ffmpeg / auto / 0) on any miss.\n    await this.loadFfmpegConfig()\n\n    // Decoder placement resolver removed: the `decoder` cap is now a\n    // singleton (phase 2g+). The capability registry resolves the one\n    // active provider per node; no per-device decoder-node routing.\n    const rtspProvider = this.brokerManager.getRtspRestreamProvider()\n\n    // Load persisted RTSP state from addon settings store\n    try {\n      const addonStore = (await this.ctx.settings?.readAddonStore()) ?? {}\n      const tokenData = asJsonObject(addonStore['rtspTokens'])\n      if (tokenData) {\n        const tokens = new Map<string, string>()\n        for (const [k, v] of Object.entries(tokenData)) {\n          if (typeof v === 'string') tokens.set(k, v)\n        }\n        this.brokerManager.loadPersistedTokens(tokens)\n        this.ctx.logger.info('Loaded persisted RTSP tokens', { meta: { tokenCount: tokens.size } })\n      }\n      const enabledData = asJsonObject(addonStore['rtspEnabled'])\n      if (enabledData) {\n        const states = new Map<string, boolean>()\n        for (const [k, v] of Object.entries(enabledData)) {\n          states.set(k, Boolean(v))\n        }\n        rtspProvider.loadPersistedEnabled(states)\n        this.ctx.logger.info('Loaded persisted RTSP enabled states', { meta: { stateCount: states.size } })\n      }\n    } catch { /* first boot — no persisted state yet */ }\n\n    const rtspTokensState = this.state<RtspTokensBlob>('rtspTokens', RtspTokensSchema, {})\n    rtspProvider.setTokenPersister(async (tokens) => {\n      // MERGE, never replace. `tokens` holds ONLY the brokers currently\n      // registered with the listen server; a broker that was later\n      // `unregisterRestreamer`-ed disappears from this map. A naive replace\n      // would erase its persisted token, turning an external NVR/recording\n      // URL into a dead 404 on the next idle/reboot. The contract is\n      // \"tokens are stable forever per (deviceId, camStreamId) unless\n      // explicitly regenerated\" — `update` reads the durable set and merges.\n      try {\n        await rtspTokensState.update(prev => {\n          const merged: Record<string, string> = { ...prev }\n          for (const [k, v] of tokens) merged[k] = v\n          return merged\n        })\n        this.ctx.logger.info('Persisted RTSP tokens', { meta: { current: tokens.size } })\n      } catch (err: unknown) {\n        this.ctx.logger.warn('Failed to persist RTSP tokens', { meta: { error: errMsg(err) } })\n      }\n    })\n\n    const rtspEnabledState = this.state<RtspEnabledBlob>('rtspEnabled', RtspEnabledSchema, {})\n    rtspProvider.setEnabledPersister((states) => {\n      const obj: Record<string, boolean> = {}\n      for (const [k, v] of states) obj[k] = v\n      rtspEnabledState.set(obj).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist RTSP enabled states', { meta: { error: errMsg(err) } })\n      })\n    })\n\n    // Device overrides (per-profile pre-buffer, debug flags, derived\n    // streams) — whole-object durable handle so NO field is ever dropped.\n    // The previous hand-rolled persister silently omitted `derivedStreams`,\n    // which made derived streams vanish on every restart.\n    const deviceOverridesState = this.state<DeviceOverridesMap>('deviceOverrides', DeviceOverridesMapSchema, {})\n    try {\n      const obj = await deviceOverridesState.get()\n      const overrides = new Map<number, DeviceOverride>()\n      for (const [rawKey, ov] of Object.entries(obj)) {\n        const id = Number(rawKey)\n        if (Number.isInteger(id)) overrides.set(id, ov)\n      }\n      this.brokerManager.loadPersistedDeviceOverrides(overrides)\n      this.ctx.logger.info('Loaded persisted device overrides', { meta: { overrideCount: overrides.size } })\n    } catch (err) {\n      this.ctx.logger.warn('Failed to load device overrides', { meta: { error: errMsg(err) } })\n    }\n\n    this.brokerManager.setDeviceOverridePersister((overrides) => {\n      const obj: DeviceOverridesMap = {}\n      for (const [deviceId, ov] of overrides) obj[`${deviceId}`] = ov\n      deviceOverridesState.set(obj).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist device overrides', { meta: { error: errMsg(err) } })\n      })\n    })\n\n    // Profile-map persistence: load on boot, persist on every mutation —\n    // through a durable handle (whole-object Zod round-trip).\n    const profileMapState = this.state<ProfileMapBlob>('profileMap', ProfileMapSchema, {})\n    try {\n      const obj = await profileMapState.get()\n      const loaded = new Map<number, PersistedAssignment>()\n      for (const [rawKey, val] of Object.entries(obj)) {\n        const deviceId = Number(rawKey)\n        if (!Number.isInteger(deviceId)) continue\n        const map: Partial<Record<CamProfile, string>> = {}\n        for (const profile of PROFILE_KEYS) {\n          const v = val.map[profile]\n          if (typeof v === 'string' && v.length > 0) map[profile] = v\n        }\n        loaded.set(deviceId, { map, auto: val.auto })\n      }\n      this.brokerManager.loadPersistedProfileMap(loaded)\n      this.ctx.logger.info('Loaded persisted profile map', { meta: { deviceCount: loaded.size } })\n    } catch (err) {\n      this.ctx.logger.warn('Failed to load profile map', { meta: { error: errMsg(err) } })\n    }\n\n    const profileMapPersister: ProfileMapPersister = (assignments) => {\n      const obj: ProfileMapBlob = {}\n      for (const [deviceId, assignment] of assignments) {\n        obj[`${deviceId}`] = { map: { ...assignment.map }, auto: assignment.auto }\n      }\n      profileMapState.set(obj).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist profile map', { meta: { error: errMsg(err) } })\n      })\n    }\n    this.brokerManager.setProfileMapPersister(profileMapPersister)\n\n    // Start RTSP restream server\n    await this.brokerManager.startRtspServer(this.config.rtspPort)\n\n    // Per-agent hwaccel override changed — force-rotate any active shared\n    // decoder session whose owning node matches.\n    this.subscribe(\n      { category: EventCategory.PipelineAgentHwaccelChanged },\n      (event) => {\n        const { agentNodeId, reason } = event.data\n        if (typeof agentNodeId !== 'string' || agentNodeId.length === 0) return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = typeof reason === 'string' ? reason : 'hwaccel-changed'\n        void manager.rotateDecodersOnNode(agentNodeId, reasonLabel).then((rotated) => {\n          if (rotated > 0) {\n            this.ctx.logger.info('decoders rotated for hwaccel change', {\n              tags: { nodeId: agentNodeId },\n              meta: { rotated, reason: reasonLabel },\n            })\n          }\n        }).catch((err: unknown) => {\n          this.ctx.logger.warn('hwaccel-driven decoder rotation failed', {\n            tags: { nodeId: agentNodeId },\n            meta: { error: errMsg(err) },\n          })\n        })\n      },\n    )\n\n    // Decoder down → rotate active shared sessions to a healthy provider.\n    this.watchCapability(['decoder'], {\n      onDown: (nodeId, capName) => {\n        if (capName !== 'decoder') return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = `decoder-down:${nodeId}`\n        void manager.rotateDecodersOnNode(nodeId, reasonLabel).then((rotated) => {\n          if (rotated > 0) {\n            this.ctx.logger.info('decoders rotated after provider went down', {\n              tags: { nodeId },\n              meta: { rotated, reason: reasonLabel },\n            })\n          }\n        }).catch((err: unknown) => {\n          this.ctx.logger.warn('decoder-down rotation failed', {\n            tags: { nodeId },\n            meta: { error: errMsg(err) },\n          })\n        })\n      },\n    })\n\n    // ── Catalog reconcile (PULL) ──────────────────────────────────────\n    // The broker is the authority for its cam-stream registry: it PULLS each\n    // camera's `stream-catalog` and reconciles, rather than relying on a push.\n    // Trigger set: a start-up fetch, a configurable poll (the loss-is-a-bug\n    // backstop for any missed event), and low-latency device / stream-params\n    // events. The poll cadence is generous so its cost stays negligible; the\n    // events make it feel instant.\n    const reconcileAll = (reason: string): void => {\n      void this.brokerManager?.reconcileAllCatalogs().catch((err: unknown) => {\n        this.ctx.logger.warn('catalog reconcile failed', { meta: { reason, error: errMsg(err) } })\n      })\n    }\n    const reconcileDevice = (deviceId: number, reason: string): void => {\n      void this.brokerManager?.reconcileDeviceCatalog(deviceId).catch((err: unknown) => {\n        this.ctx.logger.warn('device catalog reconcile failed', {\n          tags: { deviceId }, meta: { reason, error: errMsg(err) },\n        })\n      })\n    }\n\n    await (this.brokerManager?.reconcileAllCatalogsAtStartup().catch((err: unknown) => {\n      this.ctx.logger.warn('catalog startup reconcile failed', { meta: { error: errMsg(err) } })\n    }) ?? Promise.resolve())\n    const reconcileMs = Math.max(5, this.config.catalogReconcileIntervalSec) * 1000\n    this.catalogReconcileTimer = setInterval(() => reconcileAll('poll'), reconcileMs)\n\n    this.subscribe({ category: EventCategory.DeviceRegistered }, (event) => {\n      const deviceId = event.data.deviceId\n      if (typeof deviceId === 'number') reconcileDevice(deviceId, 'device-registered')\n    })\n    this.subscribe({ category: EventCategory.DeviceUnregistered }, (event) => {\n      const deviceId = event.data.deviceId\n      if (typeof deviceId === 'number') {\n        void this.brokerManager?.retractDevice(deviceId).catch((err: unknown) => {\n          this.ctx.logger.warn('device retract failed', { tags: { deviceId }, meta: { error: errMsg(err) } })\n        })\n      }\n    })\n    this.subscribe({ category: EventCategory.StreamParamsChanged }, (event) => {\n      const deviceId = event.data.deviceId\n      if (typeof deviceId === 'number') reconcileDevice(deviceId, 'stream-params-changed')\n    })\n\n    const cameraStreamsProvider = new CameraStreamsProvider(this.brokerManager)\n\n    // Broker-integrated WebRTC server — subscribes directly to each\n    // StreamBroker's onEncodedData() callback, zero ffmpeg re-encoding.\n    //\n    // `getIceServers` is the single per-session entry-point into the\n    // `turn-provider` collection cap. `getTurnServers` is an array-output\n    // method on a collection cap, so calling it without an `addonId`\n    // hits the auto-mount's generic aggregate — the union of ICE servers\n    // from every ENABLED `turn-provider` (cloudflare-turn, future coturn\n    // / Twilio, …). There's NO per-device round-trip; the providers\n    // cache at their own layer (cloudflare-turn refreshes ICE\n    // credentials in the background ~12 h before the 24 h TTL), so the\n    // per-session call costs a single cap dispatch. Without TURN\n    // configured the call returns `[]` and WebRTC falls back to\n    // host/srflx candidates.\n    const turnApi = this.ctx.api as unknown as {\n      turnProvider?: {\n        getTurnServers: { query: (input?: object) => Promise<readonly { urls: string | string[]; username?: string; credential?: string }[]> }\n      }\n    }\n    const webrtcServer = new BrokerWebrtcServer({\n      logger: this.ctx.logger.child('webrtc'),\n      getIceServers: async () => {\n        if (!turnApi.turnProvider) return []\n        try {\n          const servers = await turnApi.turnProvider.getTurnServers.query()\n          // The cap returns readonly entries; normalise to mutable shape\n          // the werift session expects + drop entries with empty url lists.\n          const out: { urls: string | string[]; username?: string; credential?: string }[] = []\n          for (const s of servers) {\n            if (!s.urls || (Array.isArray(s.urls) && s.urls.length === 0)) continue\n            out.push({\n              urls: Array.isArray(s.urls) ? [...s.urls] : s.urls,\n              ...(s.username ? { username: s.username } : {}),\n              ...(s.credential ? { credential: s.credential } : {}),\n            })\n          }\n          this.ctx.logger.info('broker getIceServers resolved', {\n            meta: { count: out.length, urls: out.map((s) => (Array.isArray(s.urls) ? s.urls[0] : s.urls)) },\n          })\n          return out\n        } catch (err) {\n          // Non-fatal — log + return empty so the session still gets\n          // built (LAN-only / public-IP scenarios don't need TURN).\n          this.ctx.logger.warn('turnProvider.getTurnServers failed — session will start without TURN', {\n            meta: { error: err instanceof Error ? err.message : String(err) },\n          })\n          return []\n        }\n      },\n      // Feed the AdaptiveController the device's latest client-reported\n      // packet-loss % over async tRPC. The broker runs in its own\n      // addon-runner process, so it reads the hub-resident\n      // NetworkQualityTracker via `ctx.api.networkQuality.getDeviceStats`\n      // (a safe, idempotent query) and synthesizes the monotonic loss sample\n      // locally (see BrokerWebrtcServer.startAdaptive). This lets the\n      // benchmark \"Network Simulation\" presets drive the downgrade ladder\n      // without inducing real RTCP loss. Only polled inside `startAdaptive`,\n      // i.e. for `/adaptive` sessions; the RTCP Receiver-Report path\n      // (`onReceiverReport`) drives the ladder independently.\n      clientLossPct: async (deviceId) => {\n        try {\n          const stats = await this.ctx.api.networkQuality.getDeviceStats.query({ deviceId })\n          return stats?.client?.packetLossPercent ?? null\n        } catch {\n          return null\n        }\n      },\n      // Decode hw-accel for the in-process transcode feed. Resolved once\n      // (cached by the server): the configured `ffmpeg.hwAccel` preference\n      // through the platform probe, gated by what the configured ffmpeg can\n      // actually decode (`getHardwareDecodeAccels`). The feed falls back to\n      // software decode if hw decode produces no output (per-stream fragility).\n      resolveTranscodeDecodeHwAccel: () => this.resolveTranscodeDecodeHwAccel(),\n    })\n    // Wire the server into the broker manager so it registers/unregisters\n    // brokers as camera profiles come and go.\n    this.brokerManager.setWebrtcServer(webrtcServer)\n    // Retain for onShutdown teardown (closes PeerConnections → frees ICE sockets).\n    this.webrtcServer = webrtcServer\n\n    const webrtcSessionProvider = new WebrtcSessionProvider(webrtcServer, this.brokerManager)\n\n    // Per-broker metrics snapshot emission (~1 Hz). UI dashboards\n    // subscribe to `stream-broker.metrics-snapshot` and drive their\n    // overlays from event payloads instead of polling the cap.\n    this.metricsSnapshotTimer = setInterval(\n      () => this.emitBrokerMetricsSnapshot(),\n      BROKER_METRICS_SNAPSHOT_INTERVAL_MS,\n    )\n\n    // Widget bundle contribution — declares the per-camera\n    // stream-broker panel as an addon-shipped widget. Admin-ui's\n    // <WidgetRegistryProvider> registers the Module Federation remote\n    // at /api/addon-widgets/stream-broker/remoteEntry.js?v=<mtime>,\n    // then loads the exposed `./widgets` module; the form-builder /\n    // dashboard / device-tab dispatcher mounts each widget via\n    // <WidgetSlot widgetId=\"stream-broker/stream-broker-panel\" deviceId=…/>.\n    const widgetsProvider: IAddonWidgetsSourceProvider = {\n      listWidgets: async () => [\n        {\n          tab: 'device-tab',\n          label: 'Stream Brokers',\n          kind: 'remote' as const,\n          remote: {\n            remoteName: 'addon_stream_broker_widgets',\n            exposedModule: './widgets',\n            componentKey: 'stream-broker-panel',\n          },\n          stableId: 'stream-broker-panel',\n          description: 'Per-camera stream broker panel with adaptive controls.',\n          icon: 'radio',\n          bundle: 'remoteEntry.js',\n          hosts: ['device-tab', 'dashboard'] as const,\n          requires: { deviceContext: true, integrationContext: false },\n          defaultSize: 'lg' as const,\n          allowedSizes: ['md', 'lg', 'xl'] as const,\n          defaultColumns: 8,\n          defaultRows: 2,\n        },\n        {\n          tab: 'device-tab',\n          label: 'FFmpeg Parameters',\n          kind: 'remote' as const,\n          remote: {\n            remoteName: 'addon_stream_broker_widgets',\n            exposedModule: './widgets',\n            componentKey: 'ffmpeg-params',\n          },\n          stableId: 'ffmpeg-params',\n          description: 'Per-camera FFmpeg encoding parameter editor.',\n          icon: 'settings',\n          bundle: 'remoteEntry.js',\n          hosts: ['device-tab'] as const,\n          requires: { deviceContext: true, integrationContext: false },\n          defaultSize: 'md' as const,\n          allowedSizes: ['md', 'lg'] as const,\n          defaultColumns: 6,\n          defaultRows: 2,\n        },\n      ],\n    }\n\n    this.ctx.logger.info('Stream broker manager initialized')\n    // `webrtc-session` is a singleton cap. The broker registers its builtin\n    // (werift) provider unconditionally; `addon-webrtc-native` (libwebrtc) can\n    // register a second provider for the same cap, and the operator selects the\n    // active one via the framework's singleton provider preference\n    // (capabilities.setPreference → `capabilities.singleton.webrtc-session`),\n    // i.e. the system Capabilities UI — no broker-specific setting.\n    const registrations: ProviderRegistration[] = [\n      { capability: streamBrokerCapability, provider: this.brokerManager },\n      {\n        capability: cameraStreamsCapability,\n        provider: cameraStreamsProvider,\n      },\n      {\n        capability: webrtcSessionCapability,\n        provider: webrtcSessionProvider,\n      },\n      { capability: addonWidgetsSourceCapability, provider: widgetsProvider },\n    ]\n    return registrations\n  }\n\n  protected async onShutdown(): Promise<void> {\n    if (this.metricsSnapshotTimer) {\n      clearInterval(this.metricsSnapshotTimer)\n      this.metricsSnapshotTimer = null\n    }\n    if (this.catalogReconcileTimer) {\n      clearInterval(this.catalogReconcileTimer)\n      this.catalogReconcileTimer = null\n    }\n    // Close the WebRTC server FIRST: it iterates live sessions and closes\n    // each PeerConnection, which is the only thing that releases the werift\n    // ICE UDP sockets. destroyAll() handles brokers + RTSP but NOT these —\n    // leaving them open leaks the sockets past process exit and wedges the\n    // next respawned runner's ICE agent (DTLS never connects until a clean\n    // hub restart). Wrapped so a teardown error can't block broker cleanup.\n    try {\n      await this.webrtcServer?.stop()\n    } catch (err) {\n      this.ctx?.logger?.warn?.('WebRTC server stop failed during shutdown', { meta: { error: errMsg(err) } })\n    }\n    this.webrtcServer = null\n    await this.brokerManager?.destroyAll()\n    this.brokerManager = null\n  }\n\n  /**\n   * Emit one `stream-broker.metrics-snapshot` event per active broker.\n   * BrokerStats already aggregates everything UI consumers need\n   * (status, fps, bitrate, codec, subscriber count). Skipped when no\n   * brokers are registered so quiet dev runs don't emit noise.\n   */\n  private emitBrokerMetricsSnapshot(): void {\n    const manager = this.brokerManager\n    const eventBus = this.ctx.eventBus\n    if (!manager || !eventBus) return\n    const brokers = manager.getBrokerInstances()\n    if (brokers.length === 0) return\n    const rawNodeId = this.ctx.kernel.localNodeId ?? this.ctx.id\n    const nodeId = rawNodeId.includes('/') ? rawNodeId.split('/')[0]! : rawNodeId\n    const timestamp = Date.now()\n    const seenBrokerIds = new Set<string>()\n    for (const broker of brokers) {\n      const stats = broker.getStats()\n      // brokerId encoded as `<deviceId>/<profile>` (StreamBroker.deviceId\n      // field is overloaded to mean brokerId — see broker constructor).\n      const brokerId = broker.deviceId\n      seenBrokerIds.add(brokerId)\n      const slash = brokerId.indexOf('/')\n      const deviceId = slash > 0 ? Number(brokerId.slice(0, slash)) : Number(brokerId)\n      const profile = slash > 0 ? brokerId.slice(slash + 1) : ''\n      if (!Number.isFinite(deviceId)) continue\n      // Snapshot-equality skip — same pattern as pipeline-runner's\n      // emitMetricsSnapshot. Idle broker → bytewise-identical payload\n      // → no emit. Quiet brokers still emit every\n      // BROKER_METRICS_HEARTBEAT_MS so the UI's \"broker reachable\"\n      // chip doesn't go stale.\n      //\n      // Strip monotonic counters before equality — `uptimeMs` ticks\n      // every second on every broker, `totalBytes` / `packetCount`\n      // grow continuously while streaming, so the raw JSON of stats\n      // always differs and the defer never fires. We only want\n      // observable-state changes (status, fps, subscribers, codec,\n      // bitrate). The full payload still goes on the bus when an\n      // emit fires; the strip only affects the dedup key.\n      const json = stableSnapshotKey(stats, ['uptimeMs', 'totalBytes', 'packetCount'])\n      const prev = this.lastEmittedBrokerSnapshot.get(brokerId)\n      const heartbeatDue = !prev || timestamp - prev.emittedAt >= BROKER_METRICS_HEARTBEAT_MS\n      if (prev && prev.json === json && !heartbeatDue) continue\n      this.lastEmittedBrokerSnapshot.set(brokerId, { json, emittedAt: timestamp })\n      eventBus.emit(createEvent(\n        EventCategory.StreamBrokerMetricsSnapshot,\n        { type: 'device', id: deviceId, nodeId },\n        { brokerId, deviceId, profile, nodeId, stats, timestamp },\n      ))\n    }\n    // Drop cache entries for brokers that no longer exist (camera\n    // detached, stream profile unassigned). Cheap because brokers\n    // is the canonical truth and seenBrokerIds is a Set.\n    if (this.lastEmittedBrokerSnapshot.size > seenBrokerIds.size) {\n      for (const cachedId of this.lastEmittedBrokerSnapshot.keys()) {\n        if (!seenBrokerIds.has(cachedId)) this.lastEmittedBrokerSnapshot.delete(cachedId)\n      }\n    }\n  }\n\n  protected globalSettingsSchema() {\n    return this.schema({\n      sections: [\n        {\n          id: 'stream-broker-settings',\n          title: 'Stream Broker',\n          columns: 3,\n          fields: [\n            {\n              type: 'number',\n              key: 'defaultPreBufferSec',\n              label: 'Pre-buffer Duration',\n              description: 'Seconds of video to keep buffered before a detection event',\n              min: 0,\n              max: 30,\n              step: 1,\n              default: 10,\n              unit: 's',\n            },\n            {\n              type: 'number',\n              key: 'rtspPort',\n              label: 'RTSP Restream Port',\n              description: 'Port for the RTSP restream server. Requires restart.',\n              min: 1024,\n              max: 65535,\n              step: 1,\n              default: 8554,\n            },\n            {\n              type: 'number',\n              key: 'maxDecodeFps',\n              label: 'Max Decode FPS',\n              description: 'Maximum frames per second for decoded stream subscribers',\n              min: 1,\n              max: 30,\n              step: 1,\n              default: 5,\n            },\n            {\n              type: 'number',\n              key: 'catalogReconcileIntervalSec',\n              label: 'Catalog Reconcile Interval',\n              description: 'How often the broker re-pulls every camera’s stream catalog (reconcile backstop)',\n              min: 5,\n              max: 300,\n              step: 5,\n              default: 30,\n              unit: 's',\n            },\n          ],\n        },\n        {\n          id: 'stream-broker-reconnect',\n          title: 'Reconnection',\n          description: 'Controls how quickly the broker retries a failed RTSP connection (exponential backoff).',\n          columns: 2,\n          fields: [\n            {\n              type: 'number',\n              key: 'initialReconnectDelayMs',\n              label: 'Initial Reconnect Delay',\n              description: 'First retry delay after a stream disconnect',\n              min: 100,\n              max: 10000,\n              step: 100,\n              default: 1000,\n              unit: 'ms',\n            },\n            {\n              type: 'number',\n              key: 'maxReconnectDelayMs',\n              label: 'Max Reconnect Delay',\n              description: 'Upper bound for exponential backoff between retries',\n              min: 1000,\n              max: 120000,\n              step: 1000,\n              default: 30000,\n              unit: 'ms',\n            },\n          ],\n        },\n      ],\n    })\n  }\n}\n\nexport default StreamBrokerAddon\n","export class FrameDropper {\n  private intervalMs: number\n  private lastPassedAt = -Infinity\n\n  constructor(maxFps: number) {\n    this.intervalMs = maxFps > 0 ? 1000 / maxFps : 0\n  }\n\n  shouldKeep(): boolean {\n    if (this.intervalMs === 0) return true\n\n    const now = Date.now()\n    if (now - this.lastPassedAt >= this.intervalMs) {\n      this.lastPassedAt = now\n      return true\n    }\n    return false\n  }\n\n  setMaxFps(maxFps: number): void {\n    this.intervalMs = maxFps > 0 ? 1000 / maxFps : 0\n  }\n}\n","import { spawn, type ChildProcess } from 'node:child_process'\nimport type { IDecoderSession, DecoderSessionConfig, DecoderStats, EncodedPacket, DecodedFrame, Unsubscribe, IScopedLogger } from '@camstack/types'\nimport { FrameDropper } from './frame-dropper'\nimport { logBannerArgs } from './ffmpeg-args-common.js'\n\n/** Minimal no-op logger for default parameter */\nconst noopLogger: IScopedLogger = {\n  debug() {},\n  info() {},\n  warn() {},\n  error() {},\n  child() { return noopLogger },\n  withTags() { return noopLogger },\n}\n\nconst SOI = Buffer.from([0xff, 0xd8])\nconst EOI = Buffer.from([0xff, 0xd9])\n\nfunction codecToInputFormat(codec: string): string {\n  switch (codec) {\n    case 'h265':\n    case 'hevc':\n      return 'hevc'\n    case 'mjpeg':\n      return 'mjpeg'\n    case 'h264':\n    default:\n      return 'h264'\n  }\n}\n\nfunction buildFfmpegArgs(config: DecoderSessionConfig): string[] {\n  const inputFormat = codecToInputFormat(config.codec)\n  const args = [\n    ...logBannerArgs('error'),\n    // Low-latency flags: skip stream probing, disable buffering, flush immediately\n    '-fflags', '+nobuffer+flush_packets',\n    '-flags', 'low_delay',\n    '-probesize', '32',\n    '-analyzeduration', '0',\n    '-f', inputFormat, '-i', 'pipe:0',\n  ]\n\n  if (config.scale > 1) {\n    args.push('-vf', `scale=iw/${config.scale}:ih/${config.scale}`)\n  }\n\n  args.push('-f', 'image2pipe', '-vcodec', 'mjpeg', '-q:v', '3', '-threads', '1', 'pipe:1')\n  return args\n}\n\nexport class FfmpegDecoderSession implements IDecoderSession {\n  private config: DecoderSessionConfig\n  private frameDropper: FrameDropper\n  private process: ChildProcess | null = null\n  private frameCallbacks = new Set<(frame: DecodedFrame) => void>()\n  private outputBuffer = Buffer.alloc(0)\n  private destroyed = false\n  private readonly logger: IScopedLogger\n\n  // Cached dimensions — won't change between frames from the same FFmpeg session\n  private cachedWidth = 0\n  private cachedHeight = 0\n\n  // Stats tracking\n  private inputPackets = 0\n  private outputFrames = 0\n  private droppedFrames = 0\n  private totalDecodeTimeMs = 0\n  private decodeCount = 0\n  private startTime = Date.now()\n\n  constructor(config: DecoderSessionConfig, logger: IScopedLogger = noopLogger) {\n    this.config = { ...config }\n    this.logger = logger\n    this.frameDropper = new FrameDropper(config.maxFps)\n    this.spawnFfmpeg()\n  }\n\n  private spawnFfmpeg(): void {\n    if (this.destroyed) return\n\n    this.killFfmpeg()\n    this.outputBuffer = Buffer.alloc(0)\n    this.cachedWidth = 0\n    this.cachedHeight = 0\n\n    const args = buildFfmpegArgs(this.config)\n    this.process = spawn('ffmpeg', args)\n\n    // Swallow EPIPE errors on stdin during shutdown\n    this.process.stdin?.on('error', () => {})\n\n    this.process.stdout?.on('data', (chunk: Buffer) => {\n      this.handleOutputData(chunk)\n    })\n\n    this.process.stderr?.on('data', (data: Buffer) => {\n      const line = data.toString().trim()\n      if (line) this.logger.debug('ffmpeg stderr', { meta: { line } })\n    })\n\n    this.process.on('error', (err: Error) => {\n      this.logger.error('FFmpeg decoder spawn error', { meta: { error: err.message } })\n    })\n\n    this.process.on('close', (_code, _signal) => {\n      if (!this.destroyed) {\n        this.process = null\n      }\n    })\n  }\n\n  private killFfmpeg(): void {\n    if (this.process) {\n      try {\n        this.process.kill('SIGKILL')\n      } catch {\n        // already dead\n      }\n      this.process = null\n    }\n  }\n\n  private handleOutputData(chunk: Buffer): void {\n    this.outputBuffer = Buffer.concat([this.outputBuffer, chunk])\n\n    // Extract complete JPEG frames from the buffer\n    let searchFrom = 0\n    while (true) {\n      const soiIndex = this.outputBuffer.indexOf(SOI, searchFrom)\n      if (soiIndex === -1) break\n\n      const eoiIndex = this.outputBuffer.indexOf(EOI, soiIndex + 2)\n      if (eoiIndex === -1) break\n\n      const frameEnd = eoiIndex + 2\n      const jpegData = this.outputBuffer.subarray(soiIndex, frameEnd)\n\n      // Advance past the consumed frame\n      searchFrom = frameEnd\n\n      this.emitFrame(Buffer.from(jpegData))\n    }\n\n    // Keep only unprocessed tail\n    if (searchFrom > 0) {\n      this.outputBuffer = Buffer.from(this.outputBuffer.subarray(searchFrom))\n    }\n  }\n\n  private emitFrame(data: Buffer): void {\n    const decodeStart = Date.now()\n\n    if (!this.frameDropper.shouldKeep()) {\n      this.droppedFrames++\n      return\n    }\n\n    const decodeTime = Date.now() - decodeStart\n    this.totalDecodeTimeMs += decodeTime\n    this.decodeCount++\n    this.outputFrames++\n\n    // Only parse dimensions on first frame or after FFmpeg restart (cached=0)\n    if (this.cachedWidth === 0) {\n      const dims = parseJpegDimensions(data)\n      this.cachedWidth = dims.width\n      this.cachedHeight = dims.height\n    }\n\n    const frame: DecodedFrame = {\n      data,\n      width: this.cachedWidth,\n      height: this.cachedHeight,\n      format: 'jpeg',\n      timestamp: Date.now(),\n    }\n\n    for (const cb of this.frameCallbacks) {\n      cb(frame)\n    }\n  }\n\n  pushPacket(packet: EncodedPacket): void {\n    if (this.destroyed || !this.process?.stdin) return\n    this.inputPackets++\n    try {\n      this.process.stdin.write(packet.data)\n    } catch {\n      // stdin may be closed if ffmpeg crashed\n    }\n  }\n\n  onFrame(callback: (frame: DecodedFrame) => void): Unsubscribe {\n    this.frameCallbacks.add(callback)\n    return () => {\n      this.frameCallbacks.delete(callback)\n    }\n  }\n\n  updateConfig(update: Partial<DecoderSessionConfig>): void {\n    const needsRestart =\n      (update.scale !== undefined && update.scale !== this.config.scale) ||\n      (update.outputFormat !== undefined && update.outputFormat !== this.config.outputFormat) ||\n      (update.codec !== undefined && update.codec !== this.config.codec)\n\n    this.config = { ...this.config, ...update }\n\n    if (update.maxFps !== undefined) {\n      this.frameDropper.setMaxFps(update.maxFps)\n    }\n\n    if (needsRestart) {\n      this.spawnFfmpeg()\n    }\n  }\n\n  async destroy(): Promise<void> {\n    if (this.destroyed) return\n    this.destroyed = true\n    this.killFfmpeg()\n    this.frameCallbacks.clear()\n  }\n\n  getStats(): DecoderStats {\n    const uptimeSec = Math.max((Date.now() - this.startTime) / 1000, 1)\n    return {\n      inputFps: this.inputPackets / uptimeSec,\n      outputFps: this.outputFrames / uptimeSec,\n      avgDecodeTimeMs: this.decodeCount > 0 ? this.totalDecodeTimeMs / this.decodeCount : 0,\n      droppedFrames: this.droppedFrames,\n    }\n  }\n}\n\n/**\n * Parse JPEG SOF0/SOF2 marker to extract image dimensions.\n * Scans for 0xFF 0xC0 (baseline) or 0xFF 0xC2 (progressive) markers.\n * Returns {width: 0, height: 0} if not found.\n */\nfunction parseJpegDimensions(data: Buffer): { width: number; height: number } {\n  for (let i = 0; i < data.length - 8; i++) {\n    if (data[i] === 0xFF && (data[i + 1] === 0xC0 || data[i + 1] === 0xC2)) {\n      const height = (data[i + 5]! << 8) | data[i + 6]!\n      const width = (data[i + 7]! << 8) | data[i + 8]!\n      return { width, height }\n    }\n  }\n  return { width: 0, height: 0 }\n}\n","import type { IDecoderProvider, DecoderSessionConfig, IDecoderSession } from '@camstack/types'\nimport { FfmpegDecoderSession } from './ffmpeg-decoder-session'\n\nconst SUPPORTED_CODECS = new Set(['h264', 'h265', 'hevc', 'mjpeg'])\n\nexport class FfmpegDecoderProvider implements IDecoderProvider {\n  readonly id = 'ffmpeg'\n  readonly name = 'FFmpeg Decoder'\n  /** Software decoder — used as fallback when hardware decoders are unavailable. */\n  readonly priority = 50\n\n  async supportsCodec(input: { codec: string }): Promise<boolean> {\n    return SUPPORTED_CODECS.has(input.codec)\n  }\n\n  async createSession(config: DecoderSessionConfig): Promise<IDecoderSession> {\n    return new FfmpegDecoderSession(config)\n  }\n}\n","/**\n * Generic length-prefix codec.\n *\n * Wire format: [4 bytes length (BE uint32)] [payload bytes]\n *\n * The length field is the size of the payload only (not including the\n * 4-byte prefix itself). Maximum payload size: 2^32 - 1 bytes. Single\n * frames larger than this will throw at encode time.\n *\n * Decode is stateful — callers feed raw bytes via `push()` and receive\n * 0+ complete payloads. Bytes straddling frame boundaries are buffered\n * internally until the next `push()` completes the frame.\n *\n * This codec is transport-agnostic. It's used on top of any stream-\n * oriented transport (TCP, UDS, in-process). It does NOT work for\n * datagram transports (UDP) — those need a different framing strategy\n * because the transport itself provides message boundaries.\n */\nimport type { FrameCodec, FrameDecoder } from '../types.js'\n\nconst LENGTH_PREFIX_BYTES = 4\n/** Hard cap — 2^31-1 to stay safely within uint32 BE range and V8 Buffer limits. */\nconst MAX_PAYLOAD_BYTES = 0x7fffffff\n\nexport interface LengthPrefixCodecOptions<TMessage> {\n  /** Encode a message to its payload bytes (without any length prefix). */\n  readonly serialize: (msg: TMessage) => Uint8Array\n  /** Decode a single complete payload back to a message. */\n  readonly deserialize: (bytes: Uint8Array) => TMessage\n  /** Codec identifier for diagnostics/metrics. */\n  readonly kind?: string\n}\n\nexport class LengthPrefixCodec<TMessage> implements FrameCodec<TMessage> {\n  readonly kind: string\n  private readonly serialize: (msg: TMessage) => Uint8Array\n  private readonly deserialize: (bytes: Uint8Array) => TMessage\n\n  constructor(options: LengthPrefixCodecOptions<TMessage>) {\n    this.serialize = options.serialize\n    this.deserialize = options.deserialize\n    this.kind = options.kind ?? 'length-prefix'\n  }\n\n  encode(msg: TMessage): Uint8Array {\n    const payload = this.serialize(msg)\n    if (payload.length > MAX_PAYLOAD_BYTES) {\n      throw new Error(`LengthPrefixCodec.encode: payload too large (${payload.length} bytes, max ${MAX_PAYLOAD_BYTES})`)\n    }\n    const buf = Buffer.allocUnsafe(LENGTH_PREFIX_BYTES + payload.length)\n    buf.writeUInt32BE(payload.length, 0)\n    buf.set(payload, LENGTH_PREFIX_BYTES)\n    return buf\n  }\n\n  createDecoder(): FrameDecoder<TMessage> {\n    return new LengthPrefixDecoder<TMessage>(this.deserialize)\n  }\n}\n\nclass LengthPrefixDecoder<TMessage> implements FrameDecoder<TMessage> {\n  /** Accumulates bytes until at least one full frame is available. */\n  private buffer = Buffer.alloc(0)\n  private readonly deserialize: (bytes: Uint8Array) => TMessage\n\n  constructor(deserialize: (bytes: Uint8Array) => TMessage) {\n    this.deserialize = deserialize\n  }\n\n  push(chunk: Uint8Array): readonly TMessage[] {\n    // Concat the new chunk onto the pending buffer. Allocation-light for the\n    // common case where chunk already contains one or more complete frames.\n    this.buffer = this.buffer.length === 0\n      ? Buffer.from(chunk)\n      : Buffer.concat([this.buffer, Buffer.from(chunk)])\n\n    const out: TMessage[] = []\n    let offset = 0\n\n    while (true) {\n      // Need at least the length prefix.\n      if (this.buffer.length - offset < LENGTH_PREFIX_BYTES) break\n\n      const payloadLen = this.buffer.readUInt32BE(offset)\n      if (payloadLen > MAX_PAYLOAD_BYTES) {\n        // Corrupted stream — reset to avoid cascading failures.\n        this.buffer = Buffer.alloc(0)\n        throw new Error(`LengthPrefixDecoder: declared payload length ${payloadLen} exceeds max ${MAX_PAYLOAD_BYTES}`)\n      }\n\n      const totalFrameBytes = LENGTH_PREFIX_BYTES + payloadLen\n      if (this.buffer.length - offset < totalFrameBytes) {\n        // Frame is incomplete — wait for more bytes.\n        break\n      }\n\n      const payloadStart = offset + LENGTH_PREFIX_BYTES\n      const payloadEnd = payloadStart + payloadLen\n      const payload = this.buffer.subarray(payloadStart, payloadEnd)\n      out.push(this.deserialize(payload))\n      offset = payloadEnd\n    }\n\n    // Trim the consumed prefix to free memory once per call (vs. per-frame).\n    if (offset > 0) {\n      this.buffer = this.buffer.subarray(offset)\n    }\n\n    return out\n  }\n\n  reset(): void {\n    this.buffer = Buffer.alloc(0)\n  }\n}\n","/**\n * Codec specialized for `DecodedFrame` values.\n *\n * Wraps a fixed-size binary header + raw payload inside a\n * `LengthPrefixCodec` so the transport layer doesn't need to know the\n * shape. Using a bespoke binary layout (instead of JSON via superjson)\n * keeps per-frame overhead low — critical on the hot path where a broker\n * fans out a frame to many consumers at 30+ fps.\n *\n * Wire format (after the length prefix added by LengthPrefixCodec):\n *\n *   Offset  Size  Field\n *   ──────  ────  ─────────────────────────────────────────────────\n *   0       8     timestamp (BE bigint64, ms epoch)\n *   8       4     width  (BE uint32)\n *   12      4     height (BE uint32)\n *   16      1     format enum (0=jpeg, 1=rgb, 2=yuv420, 3=gray, 4=bgr)\n *   17      3     reserved (zero-padded for 4-byte alignment)\n *   20      N     frame data\n *\n * Fixed header: 20 bytes. All subsequent bytes are the raw frame payload.\n * The outer `LengthPrefixCodec` adds the 4-byte length prefix, so the\n * total on-wire size per frame is `4 + 20 + data.length` = 24 + data.\n *\n * Enum values are append-only: existing values must never be reassigned\n * — a worker built before a new format is added would still send the\n * old code over the wire.\n */\nimport type { DecodedFrame } from '@camstack/types'\nimport { LengthPrefixCodec } from './length-prefix-codec.js'\n\nconst HEADER_BYTES = 20\nconst FORMAT_MAP = {\n  jpeg: 0,\n  rgb: 1,\n  yuv420: 2,\n  gray: 3,\n  bgr: 4,\n} as const satisfies Record<DecodedFrame['format'], number>\n\ntype FormatNumber = typeof FORMAT_MAP[keyof typeof FORMAT_MAP]\n\nconst FORMAT_REVERSE: Readonly<Record<FormatNumber, DecodedFrame['format']>> = {\n  0: 'jpeg',\n  1: 'rgb',\n  2: 'yuv420',\n  3: 'gray',\n  4: 'bgr',\n}\n\nfunction isKnownFormatNumber(n: number): n is FormatNumber {\n  return n === 0 || n === 1 || n === 2 || n === 3 || n === 4\n}\n\nexport function encodeDecodedFrame(frame: DecodedFrame): Uint8Array {\n  const payload = Buffer.allocUnsafe(HEADER_BYTES + frame.data.length)\n  payload.writeBigInt64BE(BigInt(frame.timestamp), 0)\n  payload.writeUInt32BE(frame.width, 8)\n  payload.writeUInt32BE(frame.height, 12)\n  payload.writeUInt8(FORMAT_MAP[frame.format], 16)\n  // Zero-fill the 3 reserved bytes (17..19)\n  payload.writeUInt8(0, 17)\n  payload.writeUInt8(0, 18)\n  payload.writeUInt8(0, 19)\n  payload.set(frame.data, HEADER_BYTES)\n  return payload\n}\n\nexport function decodeDecodedFrame(bytes: Uint8Array): DecodedFrame {\n  if (bytes.length < HEADER_BYTES) {\n    throw new Error(`decodeDecodedFrame: payload too short (${bytes.length} < ${HEADER_BYTES})`)\n  }\n  const buf = Buffer.from(bytes.buffer, bytes.byteOffset, bytes.byteLength)\n  const timestampBig = buf.readBigInt64BE(0)\n  // Safe because timestamps in ms fit in Number.MAX_SAFE_INTEGER until year 287396.\n  const timestamp = Number(timestampBig)\n  const width = buf.readUInt32BE(8)\n  const height = buf.readUInt32BE(12)\n  const formatNum = buf.readUInt8(16)\n  if (!isKnownFormatNumber(formatNum)) {\n    throw new Error(`decodeDecodedFrame: unknown format enum ${formatNum}`)\n  }\n  const format = FORMAT_REVERSE[formatNum]\n  const data = Buffer.from(buf.subarray(HEADER_BYTES))\n  return { data, width, height, format, timestamp }\n}\n\n/**\n * Pre-built codec instance for `DecodedFrame`.\n * Composes `LengthPrefixCodec` with the DecodedFrame serializers above.\n */\nexport const decodedFrameCodec = new LengthPrefixCodec<DecodedFrame>({\n  kind: 'decoded-frame',\n  serialize: encodeDecodedFrame,\n  deserialize: decodeDecodedFrame,\n})\n","/**\n * TCP frame transport.\n *\n * Wire:\n *   - URL scheme: `tcp://host:port/`\n *   - Producer calls `listen({host, port})` → server bound to a TCP port\n *   - Consumer calls `connect(\"tcp://host:port/\")` → opens a socket\n *   - Bidirectional byte stream via `net.Socket`\n *\n * The transport is framing-agnostic. The codec stacked on top (usually\n * `LengthPrefixCodec`) handles message boundaries.\n *\n * Backpressure: `net.Socket.write()` returns false when the internal\n * buffer is full. We expose `bufferedBytes` (from `socket.writableLength`)\n * so the FrameServer's drop policy can react. We do NOT wait for 'drain'\n * inside `send()` — the caller (FrameServer) decides whether to drop.\n *\n * Max outbound buffer is set via `socket.setDefaultEncoding`-style config;\n * Node's default high-water mark is 16 KB which is too small for video\n * frames. We bump the socket's writable high-water mark at connection time.\n */\nimport net from 'node:net'\nimport { randomUUID } from 'node:crypto'\nimport type {\n  FrameTransport,\n  TransportEndpoint,\n  TransportConnection,\n  TransportListenOptions,\n  TransportConnectOptions,\n} from '../types.js'\nimport type { Unsubscribe } from '@camstack/types'\n\nconst URL_SCHEME = 'tcp://'\n\nexport class TcpTransport implements FrameTransport {\n  readonly kind = 'tcp' as const\n\n  async listen(options?: TransportListenOptions): Promise<TransportEndpoint> {\n    const host = options?.host ?? '127.0.0.1'\n    const port = options?.port ?? 0\n    return new Promise<TransportEndpoint>((resolve, reject) => {\n      const connectionHandlers = new Set<(conn: TransportConnection) => void>()\n      const connections = new Set<TcpConnection>()\n\n      const server = net.createServer((socket) => {\n        socket.setNoDelay(true)\n        const conn = new TcpConnection(socket, () => {\n          connections.delete(conn)\n        })\n        connections.add(conn)\n        for (const handler of connectionHandlers) handler(conn)\n      })\n\n      server.once('error', (err) => reject(err))\n\n      server.listen(port, host, () => {\n        server.removeAllListeners('error')\n        const address = server.address()\n        if (address === null || typeof address === 'string') {\n          server.close()\n          reject(new Error('TcpTransport: server.address() returned unexpected value'))\n          return\n        }\n        // Use the actual bound address/port (host may be 0.0.0.0, port may be 0 → OS assigned).\n        const boundHost = address.address === '::' ? '0.0.0.0' : address.address\n        const url = `${URL_SCHEME}${boundHost}:${address.port}/`\n        const namespace = options?.namespace\n        const finalUrl = namespace !== undefined ? `${url}${namespace}` : url\n\n        const endpoint: TransportEndpoint = {\n          kind: 'tcp',\n          url: finalUrl,\n          get activeConnections() { return connections.size },\n          onConnection(handler) {\n            connectionHandlers.add(handler)\n            return () => connectionHandlers.delete(handler)\n          },\n          async close() {\n            for (const conn of [...connections]) {\n              conn.close('endpoint-closed')\n            }\n            await new Promise<void>((res) => {\n              server.close(() => res())\n            })\n          },\n        }\n        resolve(endpoint)\n      })\n    })\n  }\n\n  async connect(url: string, options?: TransportConnectOptions): Promise<TransportConnection> {\n    const parsed = parseTcpUrl(url)\n    if (!parsed) {\n      throw new Error(`TcpTransport: invalid URL \"${url}\"`)\n    }\n\n    const timeoutMs = options?.timeoutMs ?? 10_000\n\n    return new Promise<TransportConnection>((resolve, reject) => {\n      const socket = new net.Socket()\n      socket.setNoDelay(true)\n\n      let settled = false\n      const timer = setTimeout(() => {\n        if (settled) return\n        settled = true\n        socket.destroy()\n        reject(new Error(`TcpTransport: connect timeout after ${timeoutMs}ms (${url})`))\n      }, timeoutMs)\n\n      socket.once('error', (err) => {\n        if (settled) return\n        settled = true\n        clearTimeout(timer)\n        reject(err)\n      })\n\n      socket.once('connect', () => {\n        if (settled) return\n        settled = true\n        clearTimeout(timer)\n        socket.removeAllListeners('error')\n        resolve(new TcpConnection(socket, () => { /* client-side dispose noop */ }))\n      })\n\n      socket.connect(parsed.port, parsed.host)\n    })\n  }\n}\n\nfunction parseTcpUrl(url: string): { host: string; port: number } | null {\n  if (!url.startsWith(URL_SCHEME)) return null\n  const rest = url.slice(URL_SCHEME.length)\n  // Strip trailing path/namespace — only host:port matters for routing.\n  const authority = rest.split('/')[0] ?? ''\n  const lastColon = authority.lastIndexOf(':')\n  if (lastColon === -1) return null\n  const host = authority.slice(0, lastColon)\n  const port = Number(authority.slice(lastColon + 1))\n  if (!host || !Number.isInteger(port) || port <= 0 || port > 65535) return null\n  return { host, port }\n}\n\nclass TcpConnection implements TransportConnection {\n  readonly id = randomUUID()\n  readonly kind = 'tcp' as const\n  readonly remoteAddress: string\n\n  private readonly dataHandlers = new Set<(chunk: Uint8Array) => void>()\n  private readonly closeHandlers = new Set<(reason?: string) => void>()\n  private closed = false\n\n  constructor(\n    private readonly socket: net.Socket,\n    private readonly onDispose: () => void,\n  ) {\n    this.remoteAddress = socket.remoteAddress !== undefined && socket.remotePort !== undefined\n      ? `${socket.remoteAddress}:${socket.remotePort}`\n      : 'unknown'\n\n    socket.on('data', (chunk: Buffer) => {\n      if (this.closed) return\n      for (const handler of this.dataHandlers) handler(chunk)\n    })\n\n    const emitClose = (reason?: string) => {\n      if (this.closed) return\n      this.closed = true\n      for (const handler of this.closeHandlers) handler(reason)\n      this.onDispose()\n    }\n    socket.on('close', () => emitClose('socket-closed'))\n    socket.on('error', (err) => emitClose(err.message))\n    socket.on('end', () => emitClose('peer-ended'))\n  }\n\n  send(chunk: Uint8Array): void {\n    if (this.closed) return\n    // We ignore the boolean returned by write() — the FrameServer's drop\n    // policy uses `bufferedBytes` to decide when to back off.\n    this.socket.write(chunk)\n  }\n\n  onData(handler: (chunk: Uint8Array) => void): Unsubscribe {\n    this.dataHandlers.add(handler)\n    return () => this.dataHandlers.delete(handler)\n  }\n\n  onClose(handler: (reason?: string) => void): Unsubscribe {\n    this.closeHandlers.add(handler)\n    return () => this.closeHandlers.delete(handler)\n  }\n\n  close(reason?: string): void {\n    if (this.closed) return\n    this.closed = true\n    for (const handler of this.closeHandlers) handler(reason)\n    this.socket.destroy()\n    this.onDispose()\n  }\n\n  get bufferedBytes(): number {\n    return this.socket.writableLength\n  }\n}\n","/**\n * Unix Domain Socket frame transport.\n *\n * Same-host IPC with lower overhead than TCP loopback (no TCP/IP stack\n * traversal, no three-way handshake). Best choice when producer and\n * consumer are on the same machine — roughly 2-3× the throughput of TCP\n * loopback for large binary payloads like decoded frames.\n *\n * Wire:\n *   - URL scheme: `unix:///absolute/path/to/socket.sock`\n *   - Producer calls `listen({socketPath})` → creates the socket file\n *   - Consumer calls `connect(\"unix:///.../socket.sock\")` → opens AF_UNIX\n *   - Bidirectional stream via `net.Socket` (Node treats AF_UNIX as a\n *     regular socket behind the same API)\n *\n * The socket file is removed on `close()` and at server startup if stale.\n * Windows note: Node supports AF_UNIX on Windows 10+ but named pipes may\n * be preferable. This transport uses `net.createServer` which handles\n * AF_UNIX natively on all platforms Node targets.\n */\nimport net from 'node:net'\nimport * as fs from 'node:fs'\nimport * as path from 'node:path'\nimport * as os from 'node:os'\nimport { randomUUID } from 'node:crypto'\nimport type {\n  FrameTransport,\n  TransportEndpoint,\n  TransportConnection,\n  TransportListenOptions,\n  TransportConnectOptions,\n} from '../types.js'\nimport type { Unsubscribe } from '@camstack/types'\n\nconst URL_SCHEME = 'unix://'\n\nexport class UdsTransport implements FrameTransport {\n  readonly kind = 'uds' as const\n\n  async listen(options?: TransportListenOptions): Promise<TransportEndpoint> {\n    const socketPath = options?.socketPath ?? path.join(os.tmpdir(), `camstack-frame-${randomUUID()}.sock`)\n\n    // Remove stale socket file if it exists (common after crash).\n    try {\n      fs.unlinkSync(socketPath)\n    } catch (err) {\n      const code = (err as NodeJS.ErrnoException).code\n      if (code !== 'ENOENT') throw err\n    }\n\n    return new Promise<TransportEndpoint>((resolve, reject) => {\n      const connectionHandlers = new Set<(conn: TransportConnection) => void>()\n      const connections = new Set<UdsConnection>()\n\n      const server = net.createServer((socket) => {\n        const conn = new UdsConnection(socket, () => {\n          connections.delete(conn)\n        })\n        connections.add(conn)\n        for (const handler of connectionHandlers) handler(conn)\n      })\n\n      server.once('error', (err) => reject(err))\n\n      server.listen(socketPath, () => {\n        server.removeAllListeners('error')\n        const url = `${URL_SCHEME}${socketPath}`\n\n        const endpoint: TransportEndpoint = {\n          kind: 'uds',\n          url,\n          get activeConnections() { return connections.size },\n          onConnection(handler) {\n            connectionHandlers.add(handler)\n            return () => connectionHandlers.delete(handler)\n          },\n          async close() {\n            for (const conn of [...connections]) {\n              conn.close('endpoint-closed')\n            }\n            await new Promise<void>((res) => {\n              server.close(() => res())\n            })\n            try {\n              fs.unlinkSync(socketPath)\n            } catch {\n              // File may already be gone — ignore.\n            }\n          },\n        }\n        resolve(endpoint)\n      })\n    })\n  }\n\n  async connect(url: string, options?: TransportConnectOptions): Promise<TransportConnection> {\n    const socketPath = parseUdsUrl(url)\n    if (socketPath === null) {\n      throw new Error(`UdsTransport: invalid URL \"${url}\"`)\n    }\n\n    const timeoutMs = options?.timeoutMs ?? 10_000\n\n    return new Promise<TransportConnection>((resolve, reject) => {\n      const socket = new net.Socket()\n\n      let settled = false\n      const timer = setTimeout(() => {\n        if (settled) return\n        settled = true\n        socket.destroy()\n        reject(new Error(`UdsTransport: connect timeout after ${timeoutMs}ms (${url})`))\n      }, timeoutMs)\n\n      socket.once('error', (err) => {\n        if (settled) return\n        settled = true\n        clearTimeout(timer)\n        reject(err)\n      })\n\n      socket.once('connect', () => {\n        if (settled) return\n        settled = true\n        clearTimeout(timer)\n        socket.removeAllListeners('error')\n        resolve(new UdsConnection(socket, () => { /* client-side dispose noop */ }))\n      })\n\n      socket.connect(socketPath)\n    })\n  }\n}\n\nfunction parseUdsUrl(url: string): string | null {\n  if (!url.startsWith(URL_SCHEME)) return null\n  const rest = url.slice(URL_SCHEME.length)\n  // Expected form: `unix:///abs/path.sock` → rest starts with `/`\n  if (!rest.startsWith('/')) return null\n  return rest\n}\n\nclass UdsConnection implements TransportConnection {\n  readonly id = randomUUID()\n  readonly kind = 'uds' as const\n  readonly remoteAddress: string\n\n  private readonly dataHandlers = new Set<(chunk: Uint8Array) => void>()\n  private readonly closeHandlers = new Set<(reason?: string) => void>()\n  private closed = false\n\n  constructor(\n    private readonly socket: net.Socket,\n    private readonly onDispose: () => void,\n  ) {\n    this.remoteAddress = 'unix-peer'\n\n    socket.on('data', (chunk: Buffer) => {\n      if (this.closed) return\n      for (const handler of this.dataHandlers) handler(chunk)\n    })\n\n    const emitClose = (reason?: string) => {\n      if (this.closed) return\n      this.closed = true\n      for (const handler of this.closeHandlers) handler(reason)\n      this.onDispose()\n    }\n    socket.on('close', () => emitClose('socket-closed'))\n    socket.on('error', (err) => emitClose(err.message))\n    socket.on('end', () => emitClose('peer-ended'))\n  }\n\n  send(chunk: Uint8Array): void {\n    if (this.closed) return\n    this.socket.write(chunk)\n  }\n\n  onData(handler: (chunk: Uint8Array) => void): Unsubscribe {\n    this.dataHandlers.add(handler)\n    return () => this.dataHandlers.delete(handler)\n  }\n\n  onClose(handler: (reason?: string) => void): Unsubscribe {\n    this.closeHandlers.add(handler)\n    return () => this.closeHandlers.delete(handler)\n  }\n\n  close(reason?: string): void {\n    if (this.closed) return\n    this.closed = true\n    for (const handler of this.closeHandlers) handler(reason)\n    this.socket.destroy()\n    this.onDispose()\n  }\n\n  get bufferedBytes(): number {\n    return this.socket.writableLength\n  }\n}\n","/**\n * In-process transport — connects producer and consumer via EventEmitter\n * pairs, no real I/O. Used for:\n *\n *   - Unit tests of FrameServer/FrameClient without TCP/UDS setup\n *   - Same-process fast path (broker and consumer in the same Node process,\n *     e.g. hub-local addons consuming their own decoder)\n *\n * The URL scheme `inprocess://<namespace>` registers the endpoint in a\n * module-local registry; connects look it up by namespace. Each endpoint\n * is isolated by namespace so multiple in-process transports can coexist.\n *\n * No auth token enforcement at the transport level (the FrameServer still\n * runs its `authenticate` callback). In-process callers are assumed trusted.\n */\nimport { EventEmitter } from 'node:events'\nimport { randomUUID } from 'node:crypto'\nimport type {\n  FrameTransport,\n  TransportEndpoint,\n  TransportConnection,\n  TransportListenOptions,\n  TransportConnectOptions,\n} from '../types.js'\nimport type { Unsubscribe } from '@camstack/types'\n\nconst URL_SCHEME = 'inprocess://'\n\n/** Module-level registry: namespace → endpoint. Allows `connect()` to locate `listen()`. */\nconst registry = new Map<string, InProcessEndpoint>()\n\nexport class InProcessTransport implements FrameTransport {\n  readonly kind = 'inprocess' as const\n\n  async listen(options?: TransportListenOptions): Promise<TransportEndpoint> {\n    const namespace = options?.namespace ?? `ns-${randomUUID()}`\n    if (registry.has(namespace)) {\n      throw new Error(`InProcessTransport: namespace \"${namespace}\" already in use`)\n    }\n    const endpoint = new InProcessEndpoint(namespace)\n    registry.set(namespace, endpoint)\n    return endpoint\n  }\n\n  async connect(url: string, _options?: TransportConnectOptions): Promise<TransportConnection> {\n    if (!url.startsWith(URL_SCHEME)) {\n      throw new Error(`InProcessTransport: unsupported URL \"${url}\"`)\n    }\n    const namespace = url.slice(URL_SCHEME.length)\n    const endpoint = registry.get(namespace)\n    if (!endpoint) {\n      throw new Error(`InProcessTransport: no endpoint for namespace \"${namespace}\"`)\n    }\n    return endpoint.acceptNewConsumer()\n  }\n}\n\nclass InProcessEndpoint implements TransportEndpoint {\n  readonly kind = 'inprocess' as const\n  readonly url: string\n  private readonly connectionHandlers = new Set<(conn: TransportConnection) => void>()\n  private readonly connections = new Set<InProcessConnection>()\n  private closed = false\n\n  constructor(private readonly namespace: string) {\n    this.url = `${URL_SCHEME}${namespace}`\n  }\n\n  get activeConnections(): number {\n    return this.connections.size\n  }\n\n  onConnection(handler: (conn: TransportConnection) => void): Unsubscribe {\n    this.connectionHandlers.add(handler)\n    return () => this.connectionHandlers.delete(handler)\n  }\n\n  async close(): Promise<void> {\n    if (this.closed) return\n    this.closed = true\n    for (const conn of [...this.connections]) {\n      conn.close('endpoint-closed')\n    }\n    registry.delete(this.namespace)\n  }\n\n  /** Internal — called by the transport's `connect()` to wire a new pair. */\n  acceptNewConsumer(): InProcessConnection {\n    if (this.closed) {\n      throw new Error(`InProcessEndpoint: \"${this.namespace}\" is closed`)\n    }\n    // Two linked connections — one for the server side (what handlers see),\n    // one for the client side (what `connect()` returns). Each writes into\n    // the peer's incoming emitter.\n    const serverEmitter = new EventEmitter()\n    const clientEmitter = new EventEmitter()\n    const serverConn = new InProcessConnection(clientEmitter, serverEmitter, () => {\n      this.connections.delete(serverConn)\n    })\n    const clientConn = new InProcessConnection(serverEmitter, clientEmitter, () => {\n      // Closing the client side also tears down the server side.\n      serverConn.close('peer-closed')\n    })\n    this.connections.add(serverConn)\n    for (const handler of this.connectionHandlers) {\n      // Fire synchronously so the server can register `onData` before the\n      // client starts writing. Order of registration is the handler's\n      // responsibility, but this matches real TCP semantics well enough.\n      handler(serverConn)\n    }\n    return clientConn\n  }\n}\n\nclass InProcessConnection implements TransportConnection {\n  readonly id = randomUUID()\n  readonly kind = 'inprocess' as const\n  readonly remoteAddress = 'inprocess'\n\n  private readonly dataHandlers = new Set<(chunk: Uint8Array) => void>()\n  private readonly closeHandlers = new Set<(reason?: string) => void>()\n  private closed = false\n\n  constructor(\n    /** The peer's emitter — we write to this one via `.emit('data', ...)`. */\n    private readonly outbound: EventEmitter,\n    /** Our own emitter — we listen to this one for incoming bytes. */\n    private readonly inbound: EventEmitter,\n    private readonly onDispose: () => void,\n  ) {\n    this.inbound.on('data', (chunk: Uint8Array) => {\n      if (this.closed) return\n      for (const handler of this.dataHandlers) handler(chunk)\n    })\n    this.inbound.on('close', (reason?: string) => {\n      if (this.closed) return\n      this.closed = true\n      for (const handler of this.closeHandlers) handler(reason)\n      this.onDispose()\n    })\n  }\n\n  send(chunk: Uint8Array): void {\n    if (this.closed) return\n    this.outbound.emit('data', chunk)\n  }\n\n  onData(handler: (chunk: Uint8Array) => void): Unsubscribe {\n    this.dataHandlers.add(handler)\n    return () => this.dataHandlers.delete(handler)\n  }\n\n  onClose(handler: (reason?: string) => void): Unsubscribe {\n    this.closeHandlers.add(handler)\n    return () => this.closeHandlers.delete(handler)\n  }\n\n  close(reason?: string): void {\n    if (this.closed) return\n    // Notify both sides. The peer listens to its own inbound, which is our\n    // outbound, so emitting 'close' there tears down the peer too.\n    this.outbound.emit('close', reason)\n    this.inbound.emit('close', reason)\n  }\n\n  get bufferedBytes(): number {\n    // No real buffer — delivery is synchronous via EventEmitter.\n    return 0\n  }\n}\n","/**\n * FrameServer — producer-side high-level API.\n *\n * Responsibilities (everything a consumer of the frame-transport layer\n * wants \"just to work\"):\n *\n *   1. Open a transport endpoint.\n *   2. On each incoming consumer connection:\n *        a. Wait for a handshake message carrying the auth token.\n *        b. Run the configured `authenticate` callback — reject on failure.\n *        c. Register the connection as an active consumer.\n *   3. On `broadcast(msg)` / `sendTo(id, msg)`:\n *        a. Encode the message via the codec.\n *        b. For each target consumer, check backpressure.\n *        c. If over threshold, apply drop policy (default drop-newest).\n *        d. Otherwise `connection.send(bytes)`.\n *   4. Expose metrics (framesSent, framesDropped, activeConsumers).\n *\n * Handshake protocol (first bytes sent by the client):\n *   [4 bytes length BE] [N bytes JSON: { \"authToken\": \"...\" }]\n *\n * The handshake reuses the length-prefix framing. The payload is a small\n * JSON blob (not the frame codec) so the server can read it without\n * needing the full codec chain. Subsequent bytes from the client are\n * currently unused but reserved for future control messages (pause,\n * resume, rate changes).\n */\nimport type {\n  FrameServer,\n  FrameServerOptions,\n  FrameServerStats,\n  TransportConnection,\n  TransportEndpoint,\n} from './types.js'\nimport type { Unsubscribe } from '@camstack/types'\nimport { errMsg } from '@camstack/types'\n\n/** Pending state while waiting for the handshake. */\ninterface PendingConsumer {\n  readonly conn: TransportConnection\n  readonly unsubscribeData: Unsubscribe\n  readonly unsubscribeClose: Unsubscribe\n  /** Bytes received before the handshake completes. Capped to avoid memory abuse. */\n  buffer: Buffer\n  readonly timer: ReturnType<typeof setTimeout>\n}\n\n/** An authenticated consumer ready to receive frames. */\ninterface ActiveConsumer {\n  readonly conn: TransportConnection\n  readonly unsubscribeClose: Unsubscribe\n}\n\nconst HANDSHAKE_MAX_BYTES = 4096\nconst HANDSHAKE_TIMEOUT_MS = 5_000\nconst DEFAULT_BACKPRESSURE_THRESHOLD = 4 * 1024 * 1024\n\nexport async function createFrameServer<TMessage>(\n  options: FrameServerOptions<TMessage>,\n): Promise<FrameServer<TMessage>> {\n  const endpoint = await options.transport.listen(options.listen)\n  const server = new FrameServerImpl<TMessage>(endpoint, options)\n  return server\n}\n\nclass FrameServerImpl<TMessage> implements FrameServer<TMessage> {\n  private readonly pending = new Map<string, PendingConsumer>()\n  private readonly active = new Map<string, ActiveConsumer>()\n  private readonly connectHandlers = new Set<(id: string) => void>()\n  private readonly disconnectHandlers = new Set<(id: string, reason?: string) => void>()\n  private readonly startedAt = Date.now()\n  private framesSent = 0\n  private framesDropped = 0\n  private bytesSent = 0\n  private stopped = false\n\n  private readonly backpressureThreshold: number\n\n  constructor(\n    private readonly endpoint: TransportEndpoint,\n    private readonly options: FrameServerOptions<TMessage>,\n  ) {\n    this.backpressureThreshold = options.backpressureThresholdBytes ?? DEFAULT_BACKPRESSURE_THRESHOLD\n    endpoint.onConnection((conn) => this.handleIncomingConnection(conn))\n  }\n\n  get url(): string {\n    return this.endpoint.url\n  }\n\n  get activeConsumers(): number {\n    return this.active.size\n  }\n\n  broadcast(msg: TMessage): void {\n    if (this.stopped || this.active.size === 0) return\n    const bytes = this.options.codec.encode(msg)\n    for (const consumer of this.active.values()) {\n      this.sendTo_unsafe(consumer, bytes)\n    }\n  }\n\n  sendTo(connectionId: string, msg: TMessage): void {\n    if (this.stopped) return\n    const consumer = this.active.get(connectionId)\n    if (!consumer) return\n    const bytes = this.options.codec.encode(msg)\n    this.sendTo_unsafe(consumer, bytes)\n  }\n\n  onConsumerConnect(handler: (id: string) => void): Unsubscribe {\n    this.connectHandlers.add(handler)\n    return () => this.connectHandlers.delete(handler)\n  }\n\n  onConsumerDisconnect(handler: (id: string, reason?: string) => void): Unsubscribe {\n    this.disconnectHandlers.add(handler)\n    return () => this.disconnectHandlers.delete(handler)\n  }\n\n  getStats(): FrameServerStats {\n    return {\n      activeConsumers: this.active.size,\n      framesSent: this.framesSent,\n      framesDropped: this.framesDropped,\n      bytesSent: this.bytesSent,\n      uptimeMs: Date.now() - this.startedAt,\n    }\n  }\n\n  async stop(): Promise<void> {\n    if (this.stopped) return\n    this.stopped = true\n    for (const pending of this.pending.values()) {\n      clearTimeout(pending.timer)\n      pending.unsubscribeData()\n      pending.unsubscribeClose()\n      pending.conn.close('server-stopped')\n    }\n    this.pending.clear()\n    for (const consumer of this.active.values()) {\n      consumer.unsubscribeClose()\n      consumer.conn.close('server-stopped')\n    }\n    this.active.clear()\n    await this.endpoint.close()\n  }\n\n  // ── Internal ──────────────────────────────────────────────────────────────\n\n  private handleIncomingConnection(conn: TransportConnection): void {\n    const timer = setTimeout(() => {\n      const p = this.pending.get(conn.id)\n      if (!p) return\n      this.pending.delete(conn.id)\n      p.unsubscribeData()\n      p.unsubscribeClose()\n      conn.close('handshake-timeout')\n    }, HANDSHAKE_TIMEOUT_MS)\n\n    const unsubscribeData = conn.onData((chunk) => {\n      const pending = this.pending.get(conn.id)\n      if (!pending) return\n      pending.buffer = pending.buffer.length === 0\n        ? Buffer.from(chunk)\n        : Buffer.concat([pending.buffer, Buffer.from(chunk)])\n      if (pending.buffer.length > HANDSHAKE_MAX_BYTES) {\n        this.rejectPending(conn.id, 'handshake-too-large')\n        return\n      }\n      this.tryCompleteHandshake(conn.id)\n    })\n\n    const unsubscribeClose = conn.onClose((reason) => {\n      const p = this.pending.get(conn.id)\n      if (p) {\n        clearTimeout(p.timer)\n        p.unsubscribeData()\n        this.pending.delete(conn.id)\n      }\n      const a = this.active.get(conn.id)\n      if (a) {\n        this.active.delete(conn.id)\n        for (const h of this.disconnectHandlers) h(conn.id, reason)\n      }\n    })\n\n    this.pending.set(conn.id, {\n      conn,\n      buffer: Buffer.alloc(0),\n      timer,\n      unsubscribeData,\n      unsubscribeClose,\n    })\n  }\n\n  private tryCompleteHandshake(connectionId: string): void {\n    const pending = this.pending.get(connectionId)\n    if (!pending) return\n    if (pending.buffer.length < 4) return\n    const len = pending.buffer.readUInt32BE(0)\n    if (pending.buffer.length < 4 + len) return\n\n    const handshakeBytes = pending.buffer.subarray(4, 4 + len)\n    let token: string | null = null\n    try {\n      const json: unknown = JSON.parse(handshakeBytes.toString('utf-8'))\n      if (json !== null && typeof json === 'object' && 'authToken' in json) {\n        const t = (json as { authToken: unknown }).authToken\n        if (typeof t === 'string') token = t\n      }\n    } catch {\n      this.rejectPending(connectionId, 'handshake-invalid-json')\n      return\n    }\n\n    // Run the authenticate callback (if any). Async-safe.\n    const runAuth = async () => {\n      if (this.options.authenticate) {\n        await this.options.authenticate(token, pending.conn)\n      }\n    }\n\n    runAuth().then(\n      () => {\n        // Promote to active.\n        const p = this.pending.get(connectionId)\n        if (!p) return  // already gone (e.g. closed mid-handshake)\n        clearTimeout(p.timer)\n        p.unsubscribeData()\n        this.pending.delete(connectionId)\n\n        this.active.set(connectionId, {\n          conn: p.conn,\n          unsubscribeClose: p.unsubscribeClose,\n        })\n\n        for (const h of this.connectHandlers) h(connectionId)\n      },\n      (err: unknown) => {\n        const msg = errMsg(err)\n        this.rejectPending(connectionId, `auth-failed:${msg}`)\n      },\n    )\n  }\n\n  private rejectPending(connectionId: string, reason: string): void {\n    const p = this.pending.get(connectionId)\n    if (!p) return\n    clearTimeout(p.timer)\n    p.unsubscribeData()\n    p.unsubscribeClose()\n    this.pending.delete(connectionId)\n    p.conn.close(reason)\n  }\n\n  private sendTo_unsafe(consumer: ActiveConsumer, bytes: Uint8Array): void {\n    const policy = this.options.dropPolicy ?? 'drop-newest'\n\n    if (consumer.conn.bufferedBytes >= this.backpressureThreshold) {\n      if (policy === 'drop-newest') {\n        this.framesDropped++\n        return\n      }\n      if (policy === 'drop-oldest') {\n        // Best-effort for stream transports: we can't pop already-sent bytes\n        // out of the socket buffer. Fall back to drop-newest semantics.\n        this.framesDropped++\n        return\n      }\n      // policy === 'block' → send anyway, let Node's internal buffer grow.\n    }\n\n    consumer.conn.send(bytes)\n    this.framesSent++\n    this.bytesSent += bytes.length\n  }\n}\n","/**\n * FrameClient — consumer-side high-level API.\n *\n * Responsibilities:\n *   1. Connect the transport to the given URL.\n *   2. Send the handshake (length-prefixed JSON with authToken).\n *   3. Stream incoming bytes through the codec's incremental decoder.\n *   4. Fan out decoded messages to both:\n *        - callback handlers (via `onMessage(fn)`)\n *        - AsyncIterable (`for await (const msg of client.messages)`)\n *   5. Cleanly disconnect on `disconnect()` or remote close.\n *\n * The AsyncIterable is implemented with a small unbounded queue. If no\n * `for await` is iterating, messages are buffered; if the buffer grows\n * indefinitely this is a consumer-side memory leak. Callers who care\n * about backpressure should prefer `onMessage()` + explicit pacing, or\n * iterate promptly.\n */\nimport type {\n  FrameClient,\n  FrameClientOptions,\n  FrameClientStats,\n  FrameDecoder,\n  TransportConnection,\n} from './types.js'\nimport type { Unsubscribe } from '@camstack/types'\n\nexport function createFrameClient<TMessage>(\n  options: FrameClientOptions<TMessage>,\n): FrameClient<TMessage> {\n  return new FrameClientImpl<TMessage>(options)\n}\n\ninterface IterableState<TMessage> {\n  /** Messages waiting for a `for await` puller. */\n  buffer: TMessage[]\n  /** Pending puller's resolve function — called when a new message arrives. */\n  waiter: ((value: IteratorResult<TMessage>) => void) | null\n  done: boolean\n}\n\nclass FrameClientImpl<TMessage> implements FrameClient<TMessage> {\n  private connection: TransportConnection | null = null\n  private decoder: FrameDecoder<TMessage> | null = null\n  private readonly messageHandlers = new Set<(msg: TMessage) => void>()\n  private readonly disconnectHandlers = new Set<(reason?: string) => void>()\n  private readonly iterState: IterableState<TMessage> = {\n    buffer: [],\n    waiter: null,\n    done: false,\n  }\n\n  // Stats\n  private framesReceived = 0\n  private bytesReceived = 0\n  private connectedAtMs: number | null = null\n\n  constructor(private readonly options: FrameClientOptions<TMessage>) {}\n\n  async connect(url: string): Promise<void> {\n    if (this.connection !== null) {\n      throw new Error('FrameClient: already connected')\n    }\n\n    const conn = await this.options.transport.connect(url, this.options.connect)\n    this.connection = conn\n    this.decoder = this.options.codec.createDecoder()\n    this.connectedAtMs = Date.now()\n\n    conn.onData((chunk) => this.handleIncomingChunk(chunk))\n    conn.onClose((reason) => this.handleDisconnect(reason))\n\n    // Send the handshake. Format matches FrameServerImpl.tryCompleteHandshake:\n    // [4 bytes BE length][JSON payload]\n    const payload = Buffer.from(JSON.stringify({\n      authToken: this.options.connect?.authToken ?? null,\n    }), 'utf-8')\n    const header = Buffer.allocUnsafe(4)\n    header.writeUInt32BE(payload.length, 0)\n    conn.send(Buffer.concat([header, payload]))\n  }\n\n  get messages(): AsyncIterable<TMessage> {\n    const state = this.iterState\n    return {\n      [Symbol.asyncIterator](): AsyncIterator<TMessage> {\n        return {\n          next(): Promise<IteratorResult<TMessage>> {\n            if (state.buffer.length > 0) {\n              const value = state.buffer.shift() as TMessage\n              return Promise.resolve({ value, done: false })\n            }\n            if (state.done) {\n              return Promise.resolve({ value: undefined, done: true })\n            }\n            return new Promise((resolve) => {\n              state.waiter = resolve\n            })\n          },\n          return(): Promise<IteratorResult<TMessage>> {\n            state.done = true\n            state.buffer = []\n            if (state.waiter) {\n              state.waiter({ value: undefined, done: true })\n              state.waiter = null\n            }\n            return Promise.resolve({ value: undefined, done: true })\n          },\n        }\n      },\n    }\n  }\n\n  onMessage(handler: (msg: TMessage) => void): Unsubscribe {\n    this.messageHandlers.add(handler)\n    return () => this.messageHandlers.delete(handler)\n  }\n\n  onDisconnect(handler: (reason?: string) => void): Unsubscribe {\n    this.disconnectHandlers.add(handler)\n    return () => this.disconnectHandlers.delete(handler)\n  }\n\n  getStats(): FrameClientStats {\n    return {\n      framesReceived: this.framesReceived,\n      bytesReceived: this.bytesReceived,\n      connected: this.connection !== null,\n      connectedAtMs: this.connectedAtMs,\n    }\n  }\n\n  async disconnect(): Promise<void> {\n    const conn = this.connection\n    if (conn === null) return\n    conn.close('client-disconnect')\n    // handleDisconnect will be invoked by the onClose handler.\n  }\n\n  // ── Internal ──────────────────────────────────────────────────────────────\n\n  private handleIncomingChunk(chunk: Uint8Array): void {\n    if (this.decoder === null) return\n    this.bytesReceived += chunk.length\n    let messages: readonly TMessage[]\n    try {\n      messages = this.decoder.push(chunk)\n    } catch (err) {\n      // Framing error — tear down the connection. The decoder's internal\n      // buffer has been reset already by its own error handling.\n      const reason = err instanceof Error ? `decoder-error:${err.message}` : 'decoder-error'\n      this.handleDisconnect(reason)\n      if (this.connection) this.connection.close(reason)\n      return\n    }\n\n    for (const msg of messages) {\n      this.framesReceived++\n\n      // Fan out to callbacks.\n      for (const handler of this.messageHandlers) handler(msg)\n\n      // Push to AsyncIterable buffer or resolve the pending waiter.\n      if (this.iterState.done) continue\n      if (this.iterState.waiter) {\n        const w = this.iterState.waiter\n        this.iterState.waiter = null\n        w({ value: msg, done: false })\n      } else {\n        this.iterState.buffer.push(msg)\n      }\n    }\n  }\n\n  private handleDisconnect(reason?: string): void {\n    if (this.connection === null) return\n    this.connection = null\n    this.decoder?.reset()\n    this.decoder = null\n\n    for (const handler of this.disconnectHandlers) handler(reason)\n\n    // Drain AsyncIterable state.\n    this.iterState.done = true\n    if (this.iterState.waiter) {\n      const w = this.iterState.waiter\n      this.iterState.waiter = null\n      w({ value: undefined, done: true })\n    }\n  }\n}\n","/**\n * HMAC-based auth tokens for FrameServer subscriptions.\n *\n * Flow:\n *   1. Server issues `createAuthToken(sessionId, expiryMs)` and returns the\n *      token to the consumer via tRPC.\n *   2. Consumer sends the token in the first handshake message after\n *      connecting the transport.\n *   3. Server validates via `verifyAuthToken(token, expectedSessionId)`\n *      before accepting frame broadcasts.\n *\n * Tokens are self-contained: `${sessionId}.${expiryEpochMs}.${signature}`.\n * The signature is HMAC-SHA256 of the payload using the server's secret.\n * Base64url encoding — URL-safe, no padding.\n *\n * The secret is per-FrameServer instance and is generated at construction\n * time via `crypto.randomBytes(32)`. It never leaves the server process.\n * Rotating the secret invalidates all existing tokens.\n */\nimport * as crypto from 'node:crypto'\n\n/** Issue a signed token for a single subscription. */\nexport interface AuthSigner {\n  sign(sessionId: string, expiryEpochMs: number): string\n}\n\n/** Verify a token against expected sessionId and current wall-clock. */\nexport interface AuthVerifier {\n  verify(token: string, expectedSessionId: string): AuthVerifyResult\n}\n\nexport type AuthVerifyResult =\n  | { readonly ok: true; readonly sessionId: string; readonly expiresAt: number }\n  | { readonly ok: false; readonly reason: 'malformed' | 'bad-signature' | 'expired' | 'wrong-session' }\n\n/** Create a matched signer + verifier pair sharing the same secret. */\nexport function createAuthPair(): { signer: AuthSigner; verifier: AuthVerifier } {\n  const secret = crypto.randomBytes(32)\n  return {\n    signer: {\n      sign: (sessionId, expiryEpochMs) => signToken(secret, sessionId, expiryEpochMs),\n    },\n    verifier: {\n      verify: (token, expectedSessionId) => verifyToken(secret, token, expectedSessionId),\n    },\n  }\n}\n\n// ── Internal helpers ────────────────────────────────────────────────────────\n\nconst SEPARATOR = '.'\n\nfunction signToken(secret: Buffer, sessionId: string, expiryEpochMs: number): string {\n  if (sessionId.includes(SEPARATOR)) {\n    throw new Error(`sessionId must not contain \"${SEPARATOR}\"`)\n  }\n  const payload = `${sessionId}${SEPARATOR}${expiryEpochMs}`\n  const sig = crypto.createHmac('sha256', secret).update(payload).digest()\n  return `${payload}${SEPARATOR}${toBase64Url(sig)}`\n}\n\nfunction verifyToken(secret: Buffer, token: string, expectedSessionId: string): AuthVerifyResult {\n  const parts = token.split(SEPARATOR)\n  if (parts.length !== 3) {\n    return { ok: false, reason: 'malformed' }\n  }\n  const [sessionId, expiryStr, signatureB64] = parts as [string, string, string]\n\n  const expiryEpochMs = 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