@camstack/addon-pipeline 0.1.10 → 0.1.11

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

@@ -1 +1 @@
-{"version":3,"file":"index.js","sources":["../../src/stream-broker/stream-broker/decoder-session-proxy.ts","../../src/stream-broker/rtsp/audio/aac-config.ts","../../src/stream-broker/stream-broker/audio-codec-session.ts","../../src/stream-broker/stream-broker/frame-dropper.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/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/stream-broker-manager.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/session.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/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 } 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  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  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 * 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// Match AudioRtpDecoder's emit cadence so YAMNet/Apple SoundAnalysis\n// see the same window size regardless of source codec. A YAMNet hop is\n// ~0.48s and Apple SA classifies on rolling ~0.96s windows; 0.5s of\n// target-rate samples gives both classifiers a stable input. Codec\n// frames are typically 64–128ms so ~4–8 frames coalesce into one chunk.\nconst TARGET_CHUNK_MS = 500\nconst F32_BYTES = 4\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  // Accumulator for the ~0.5s emit window. The codec cap returns a PCM\n  // frame per decoded codec packet (~21–128ms depending on codec), which\n  // is too short for the audio classifiers downstream. We coalesce\n  // until we've buffered TARGET_CHUNK_MS worth of samples, then emit\n  // one chunk with the earliest packet's timestamp.\n  private pendingPcm: Buffer[] = []\n  private pendingBytes = 0\n  private pendingSampleRate = 0\n  private pendingChannels = 0\n  private pendingFirstTs = 0\n  private readonly chunkBytesTarget: number\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    this.chunkBytesTarget =\n      Math.max(1, Math.round(cfg.targetSampleRate * (TARGET_CHUNK_MS / 1000))) *\n      Math.max(1, cfg.targetChannels) *\n      F32_BYTES\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    // Emit whatever's in the accumulator so downstream audio\n    // analyzers don't miss the last partial window. Skipped if the\n    // accumulator is empty.\n    this.flushPending()\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   * Stage a freshly-pulled PCM frame into the accumulator. When we've\n   * buffered enough samples for the configured ~0.5s window we\n   * concatenate, copy out, and emit a single DecodedAudioChunk so\n   * downstream audio classifiers see a stable input length regardless\n   * of source codec frame size (G.711 packets ≈ 20ms, AAC ≈ 21–43ms,\n   * Opus ≈ 20ms).\n   *\n   * Format changes (sampleRate/channels) flush the previous window\n   * first so a chunk never spans two formats.\n   */\n  private appendPcm(pcm: { data: Uint8Array; sampleRate: number; channels: number; pts: number }): void {\n    const incoming = pcm.data.byteLength\n    if (incoming === 0) return\n\n    // Format change → flush the previous window before swapping params.\n    if (\n      this.pendingBytes > 0 &&\n      (pcm.sampleRate !== this.pendingSampleRate ||\n        pcm.channels !== this.pendingChannels)\n    ) {\n      this.flushPending()\n    }\n\n    if (this.pendingBytes === 0) {\n      this.pendingSampleRate = pcm.sampleRate\n      this.pendingChannels = pcm.channels\n      // pcm.pts is a codec-relative PTS (often starting at 0), not a\n      // wall-clock timestamp. Downstream consumers (audio-metrics\n      // snapshot, UI \"X seconds ago\" chip) interpret the chunk\n      // timestamp as ms-since-epoch — emitting `pts` produced\n      // bogus values like \"1777832862s ago\" because Date.now() - 0\n      // ≈ Date.now(). AudioRtpDecoder always uses Date.now() too,\n      // so the codec path now matches.\n      this.pendingFirstTs = Date.now()\n    }\n    // Buffer.from(uint8array) deep-copies — safe to keep across the\n    // RPC tick that owns the source buffer.\n    this.pendingPcm.push(Buffer.from(pcm.data))\n    this.pendingBytes += incoming\n\n    if (this.pendingBytes >= this.chunkBytesTarget) {\n      this.flushPending()\n    }\n  }\n\n  private flushPending(): void {\n    if (this.pendingBytes === 0) return\n    const merged = this.pendingPcm.length === 1\n      ? this.pendingPcm[0]!\n      : Buffer.concat(this.pendingPcm, this.pendingBytes)\n    const sampleRate = this.pendingSampleRate\n    const channels = this.pendingChannels\n    const timestamp = this.pendingFirstTs\n    this.pendingPcm = []\n    this.pendingBytes = 0\n    this.pendingSampleRate = 0\n    this.pendingChannels = 0\n    this.pendingFirstTs = 0\n    this.emit({\n      data: merged,\n      sampleRate,\n      channels,\n      timestamp,\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    // Drain whatever survived from the dead session — the new session\n    // may negotiate a different sampleRate/channels, and emitting the\n    // tail keeps timeline continuity for the classifier.\n    this.flushPending()\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","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 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","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 { 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  /** 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  }\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.sdp = buildSdp({ codec, parameterSets: all.slice(0, needed), streamName: this.streamName })\n        this.mutedSdp = buildSdp({ codec, parameterSets: all.slice(0, needed), streamName: `${this.streamName} (muted)` })\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` — same wire format the scrypted Reolink\n * plugin consumes through its `x-scrypted/x-rfc4571` MediaObject.\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 * Ported from `koush/scrypted/plugins/prebuffer-mixin/src/flv.ts` (ISC).\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 * Ported from `koush/scrypted/plugins/prebuffer-mixin/src/rtmp-client.ts`\n * (ISC). Adaptations:\n *   - `console` replaced with project `IScopedLogger`.\n *   - Inlined `readLength` helper (no dependency on @scrypted/common).\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 (replaces @scrypted/common/src/read-stream) ──\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: 'AAAAAWdCwB/aAUAW7ARAAAADAEAAAAMAg8YMqAAAAAFozg/IAAABZYiEOiYoAAjgycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJycnJyddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddddcdgARwAaAfgvaKICMwoLAOgADxEIjArGFj0HzJsgAA2AHATLIAAMgBQE3ACIwbKGUWeUI9Xg4ABsAOAmWHAANgBwEyywADYAAgMGRwADYAAgMGQ4ABkAKAm4QAAYAHAq4HAAMgBQE3A4ABgAcCrn4/ABwjgGMCXnxbX/yxvWnTbBwke+QAqCJtK/C/ANHYwBiaibAhFAwNAYBETMKEp/DdMAAVluoAAgKyf8BWwZFmQC+raGBG1IAwEgPQdQ7AEAYHCVS5MuqyLBIm21NwPwEdMjMJ8h2KQPTQfEi5poyZEw/1NvqAA8ZCKyiNLHIZ6XyAADIAQDbgYAro/YCAZstAw/vNEAqAAIBiyrw4ABkAIBtzDgAGQAgG3B/4BjvB/2xbteXFxm4BcTAgkL5YdfLwCIgyByUFDRzz1eDr5YdfLLAANgACAwZHAANgACAwZyAACgAgCrmHAAKACAKufvZY7AA7AACAQAQPJEWx8kFDGRyZ/eABHf9E/EeMqpJFVe/gAeJVOoQrZdIPEev4AH8J5YVwwpkr3+37/+/wGgClGk1MvAN9VJ6BIUWVMODsABAPgACACEiJKqGeGhQqwRGuD0AAYoXquoGIBbe/gWsgEpqcCgPRV6f7/tuZ7kEjdqDyQOaD0b0opPh4ZntwudgChUAgBWE/I8AAYkAGAVEv9aOCaoOfkLoEDDg+wVRCyEzMH/47AEQAayAAR6Q5aqJwZJPKiFgcIGQwKLx3gGA/YUFyppIB6j9xlWJQlzF8AIR0tGJhUX5SyHgEeZDUxcwVTWB4Mhp2eFARY1zoCOhp+Q6EC81gYGownlbiDaOEev9m6a2TT4oI8tUGheSeAZ2wBSQx3By0gcR7ATMD52cJQBSw7AAaQAAgBDwr4jSJP+D78Smc2YeABjq29EmwAKbIkxv0IjvgWGqYq9G5MlTBwwAdKU7iOhZJzRwZE7A1p+RoRfc5GbQmylVm76/b2bbxhNYoe9uCgBB4d8ABgUF3NjrYxnooRAfUhy/+tDhwTrnkAGr/QLuwDeijTe1eucm2iJMaolgqbh9iNSAISYVWzNzwopGPTybBCbFBQ5JPK2BCZFhQ5L3nDDenskjAMgMJBh//HYAgAQQsOA94lm1yVgZeY2cNwH3IA0lEGnefYEY7RC8pKgDWP11gofhtuOSAcqzslqMti6KBV/dO4/AuNc3VWMubLGAjpk2J8hUdv5MUFiom+G2LF4fZLU1smnooE+7gSv6RoQfZJixhcL6DAD/O0XaOYuqDsAB/BKpBxAu+mztAehC/2HABDIIcjMWcJIi4QAAbADgJlkEBgACAfc/0jmOxh4W454cgMAAQD7gAGC3JTNpByAwABAPuEAAQcdcHYAPAEAAU4A9ku2WFlBraIzQdgA6I89A0GF0ZefeAugfNPfhxkXTG9/dO4/hgbwFvygbK0JCH34AWUpSYgW0uCBv790yayNGLqn9Ejz4GgwuTLz7x2AAjIzIBDuwAEARuvvAUjhDgrkrdwOAAbADgJlgAUcGcPaePJPARkACYAAQELkrdyVu5hwBMAAICFzDgCYAAQELg78DMBGicZPcWI2EZwBSHIMAlVeYmQkdwgAA2AHATLADl6pChRdKfTGUD/7/DpHcw6R3MAZTHJlf1vCVmBBRTNFBLwOwBEQAceXMmdPjz4SymUiET4AHnSHvKNSiQkY3/f/TCXVheWRrXYj1rC8w3iShymRpDMpRNUvSzBiRfohpwSKP/3/jsABGAlUbH44AOAAEAGS+83AYQAO54I4aWUITZHVIAAJIAAICtwgAA2AHA+XzhwACSAACArchwACSAACArcnMdgAIBgzwPWjWI94CnTFwGENkADOLBHDXyDiTwEyAIgA24QAAbADgJlhwADYAcBMvAJNAAboxQABAQR6vQ4EQAbchwIgA25kARjLmHAjGXB//jT00//HYAEADAxRXdwAGJAUEjIBwADYAcBMsAIzZMAG7GCpDRHYcAA2AAIDBmnC4OvluAEe9Xg///jsABABjUSAAps2wkVCYFveHWDFwAHDXw0UMV8rsCT4AFFoMWey5MIAS3YsiA5KKNFb1/Px+m2u4BULtQ1oBSzAAesDedhpYQLHd/wYFlrxiiU4/Lv4EMVnQQSnBA4ICQpuLjScwigQKPgw4/AEBA4O0KspzCvHCBLT3kFUS2uMVoPFADMCCoTybAQE+KMf08myAAIA2lAxdzwOHCgk2rtAgHWGP6psYVYZgFqpgTbim6ablNNACVP2HAE1//7DILXP1RMBAvb47AEAPgAOAkinIIqsBw96Y6QfAGyvTkaaD0p4BK7I4GsEmnwFX907gAERtMxa8oQAAbADgJly2yZGR1XiwADYAAgMGQwVgFaoRzZPAK24AH5Amzi6sAyq6t8IzwucBsAogwf4/ABwIdwM+020dDpYaemIWkz+ADqCKwVJAxd31eLAANAACAyZ8gAAmAIA25hwACYAgDbmOwAE2wP8AS2nHNDAAObANAOAAbADgJlhwADYAcBMsAiIGwKWoo8Ahj1eLAANgACAwZDgAGwA4CZYcAA2AHATLHAANgACAwZHAANgACAwZIAAMADAVcw4ABgAYCrn47AB8L8hh1IACAD9V94OR6UXAAmMiGikKOI5fJEwxsQAAVAIC7hAABsAOAmWAEfYyuOlBxbSQdkAEBgCLmHAAKgEBdyHAAKgEBdzDgEBgCLmHAIDAEXB34GYCNE4ye4sRsIzgCkOQYBKq8xMhI7hAABsAOAmWAHL1SFCi6U+mMoH/3+HSO5h0juYAymOTK/reErMCCimaKCXgdgCIgA48uZM6fHnwllMpEInwAPOkPeUalEhIxv+/+mEurC8sjWuxHrWF5hvElDlMjSGZSiapelmDEi/RDTgkUf/v/HYACMBKo2PxwAcAAIAMl95uAwgAdzwRw0soQmyOqQAASQAAQFbhAABsAOB8vnDgAEkAAEBW5DgAEkAAEBW5OY7AAQIIChUpoTlx1ji2YnYJRI+ADsJx/w5hpafAAexE5beQAmBJX9ufszMGXuFS9sADGUszZCyvgAAJ2DFKkkA3dmAcwNxxNVWfNEYcaRsKPq4SfBgPj5ZLCAxkGA4dgCADAcfAmKkDuKaaR4AklDUBE5ITBqmbSAJTmxAZacChOftXsBINVifsKfe1LxuLMRXACuOjrfPCiiMenjtABeADAKkn5NhGyLCB6WecpEREETMNylBh//HYAwABNwABAgc/AwcEJsScUwJQO0gDSCDTvPdQIx0QvqlagC2O1pAOhNuENAMV5umu9dgy+7jt+gVnG1pGaVYJEn3+Bo1Y6EBfQuggbutXrbBgAGaF/5AYuTQKP4DTmIN2a2xI1Ge1FT8vrwA3hxx5dddddddddddddddddddddddddddddddddddddddddddddddcdgAuFDBzgAIhvaQDe4hevgA/IE6B1lHFifV7LAANAACAyZBwADQAAgMmYC5EMroYwcZ/OisMgAA0AHARLDgAGgA4HS/H4ADAABAGFAtUJHLEWUdvNCv7yNwBd27nkBUgNf+wG3ooaEQJvQIA+VMNErfFAADdtAAEBad6AOARXzBRgv4PEAAGwAwHywKfu51mgkU0QMXsOgCBF1TAAbeAcZ1jrsyAATAV91KgFcSLlpGTE+Uzlbh3AOd2yALQLNDv8AACBBAY44BxsyCmxoH5bDAFGx6cIhkuLP4AO9wBI5UcC2HHHxIAAmAAIBdz/7jSVKSCQABMAAQDbgZQAAmAAIBtz8dgAITADeHcdCk6dPaGtxFKngAPERBEDsU8Wk+ZN5AABsAOAmWHAANgBwEy9EQZA5KSRo576vfLAANgACAwZHAANgACAwZx+AAmgB4NdF2UXRVlF/4APkDIFx5rjAMT74fBkKUAg8knzfHYAsAAnASQDgrEP88/wqPjMGT46q84Bj1VIBteOgAFgBpkDg8I4ADEmARnv4FAACwAlQBmeEMAAxIkEZ4HFWEwykHKfzADEVCgT0ywx+ABEcjOsiMy8MP/AB2MIotEMLbu0E2ycNHRPh2AIAAKAsQAFxD0G+RADWZCgXdUQoAAQBAWLD88CIHfdABBEIL3z3CJAAzQAFz0lAAEAQFmATnvDgBjkqkAqV7hG4AbQAK1XCCi9gBDJSe1PL8dgCACc4AKCNGDqDq8PpgAGGSgy84YYJgAC6eAATPaMeSgAFU1AAme0YBW3eAAYZCTt72jA7DwEpQBA7JYog6goBtZ4pQIuveIwAAgCh5kOz3/iHAAEAcPEB2e8dAY5UIBDXV/hDAEgAHCRqvzo0D/X3Qregu6QHsaeOZSD4xtsvHE9p3lG2DAnhEFms+WCSAm//caB809+HGRdMb3907gMDeAt+UDZWhIQ+/fwAPKUpMQLaXBA39+6ZNZGjF1QB7GnjmUg+MbbLxxPad5Rth2AIAMCiC7DbbRcwWMtGTBj//9Ejz4GgwuTLz7+NA+ae/DjIumM7+53H+/jszH/CovoqIf+AQ4P4FGlz4c2WUwT5y+GSw1Z8QFvmhA7t+D8AAhFEqsuWinhG/8AB50h7yjUokJGt/39MJdWF5ZHtdiPWsLzDeJKEJWYE4gmijPj/KYl2cylE1S9LMGJF+iGnBIo//WGrPiAt80IHdvwY7AAQDHc9z3ue5z3PYz//4AHnSHvKNSiQka3/f47AAQRQAIkCcZEKGK6ljQvYf8ADAUgANEU6ADgX/9RoADfuUAAQEE+r3x+JDTBtvbb409NPHYACMwB4AK9oBdsCVhzc4EWBwADYAcBMsACjgzh7Tx5J4CIAjDcRHv54MlAAJgNEgnPDgAGwABAYM1LBnRUAECinq9/yYAAmAlwH56IRKxWPnh24AJyAAYCsMsoOogIirks1hgU5Cr3wgIjLghxgAkRWe/8gIjLgjBgAkzWeMwE0sMI58ICnJVa5/64/AHaaMyaRkZF//4APhodyOYIf6fAVf3TnH9Zl1oKYszA7Mx/wqL6KiH/gEOD+BRpc+HNllME+cvhksNWfEBb5oQO7fg/AAIRRKrLlop4Rv/AAedIe8o1KJCRrf9/TCXVheWR7XYj1rC8w3iShCVmBOIJooz4/ymJdnMpRNUvSzBiRfohpwSKP/1hqz4gLfNCB3b8GOwAEAx3Pc97nuc9z2M//+AB50h7yjUokJGt/3+OwBAABaA4CW4gtCt2QwABpACi659/XpOAAIB4BbkMzwUHgjBWoMqvECDHuP0AA0AFugOpcQIMe5JAABANALUB2ehGABkcAUfTCH/nlQABgALEgdDiBBj3MarwxC2QkQRAgx7g7ABwAGAgwQ9RokTFiwdI3bt1//4HARjbmABopxlo0RkAAGgAwCJY/AAQnc7AAEA8ZqqrRNkaNF///8DgAGgAoHXAAHiIRCB2KJGoPmTagABsDAk2Mtsy6666666666666666666666666666666666666666666666647AEAARnue5735/4AD5kMjgjG4gkeY5xGYiMCLXqHBJreHH4AiKAGGmzZzbQ++E8pmJJI8ZWus7fsE4kVky0YQPj/3LRGmJ8pFdgSnI4nsPwo/boxLs5lKNqm/+BzkLiB7Q8IHfv3x34yAAEBEEAWmqVIDzAtRmIKDVoiDj2PBB9HST4PBobWFQcZNsh0A2FPI5Dyphw8gCs5QNkwEhD78ABdJgihBE7/VGvX/28YAC3lwAKBkj4EEOioQXPM19lcZEMroIzEkpYodgAIBDPB32vtnj58ErgnUyN/wAfmEUMQkkYO++rxAABoAOAiWJAANAoabGHeYsAA0AAIDJkHAANAACAyZIAANABwESw4ABoAOAiW+OwAH2AAIBgAOGlyFtDJwM1ArO//7wnA5BxSlvKr36kmgw4rk8lzgFpV7IQQBDK3H/7dNcAAYBqmAPAAcRhMPjhrL3q94EE0MjDoQcU0gHBkAAGABAKucaDq1Ah4WSvE4BsOAAYAEAq4QAAbAEA24OwBBYAA44RfMRbgqUp0QT8QRAGwK8h5i52D0WCSKOdkIoPGHf4YAFBbEFncqDiQPaAC53nNEfg7gOdf+ocRgUog0UtV6zKBOWRQMAXjvDiAHIAMAuKfkuR2Iomg82HVcHYACABgUkLAwr846nzA1209AxcAC2QVL76YBuHk3wLAK2CPLXBKxYTOwsge/jx3xbzMAfXqABsPEbZQWDUiDX34FBy1A1TAAlP/3nPbManL1FStbsQIgpozHeawvIwLtTegHKP2BO6RkFSPy8C3WqQk2IZBuYl/7lJEJDMLjkKAKEAQeARwBJQABAADgvfTxvRgUqZCsLhrSTBa9TOcymHivOCApywW4MJcyzF7+/4gi3qCTtQPDx7hYG+uYC3uSYAntFpkaMXFYGDBLvjmycYqhU//6mgF4i/gJ4U+5gS6vPTjR7BHAUAIGBtjo8jM6KKkQM6Azf/caeIngsTXD7Eyn840000+0WsjmL9MB2nwfIjPANSNRiohW2lQBzPbrfIAltEGnu+sGoQuxYUNyTwNQQuRQUNyzxlaMLzxnTMLTwIYACAAMOaCQPepKNsFYGfuTnRuAAZNRZu+HWKlf4CYZqgaE7BoB/ncF8imPonfFvM9j6rg8Imzg9J8YEPvwGjlNg2TG+AJTN3xbZnsfVbXoAklFGnOfmtFaKXtIUAYt+uwPiVE3Vj8mChuUiJITZCIzbjSVKSAx3giBhisDTe9pMi+UikMKJ9DRk55AABcAUBNz8gAAuAKAm5hwAC4AoCbngAzEGMRtHiSCB/gIbRIAE79CACgRcdgAIGY5zHMcwAAgZAAP3rQHBb4YE8Igs1nsAhkhR/+kAAQDAKuYeHAAgGAVchwAIBgFXB34ATAAMAwJTJs2AF4hoHFgpSxmFnAWvKBrsJFP/bACylKTECz7ggd+/ZotZGA06/9//9WNPHMph8Y+2VjiJErVSbsMwAngmHvEXY2JIXf++QAAaADgIlgAYOCMHqJOJOJHYACIzIgGM7d573Mf/4YEj82/IAANABwESwAMFgjB6qziTix+AAgAMCCSrDKzJQqUKtEipD/AAIRRKrLlop4Rv/QlZgTiCaKM+MGmvtNDvwACRAABASFSYMjBW1YPyZ4m6YS6sKjCPaAnEetYXmG8SUOUxLs5qwmqADZjkAd6gACApXO/+g1Z8QFFdIAF99sQAATUAAQFLjAABAAAQDbgATRMCKEERbVgqNMuQAAaADgdLx2AAgEDvA5aFIj7UgUSoVf3/wAMJYMwaps8k4qAE8wAJu5AABART6vEAAGgA4CJYkAA0UcFjGuYfhUtH/uZOZ+cyczj8ACIDgDDFZmgAemBQTMv//AByNAAyoQUAVz1eLAANgACAwZAhNGgAaoQbgwR0IAANgBgEyx+AICC44ABBItz5kMgXdYtANAJ0VSAwOQ4G9/tZxpe4glKCZpAT0AnuQVxMiB9/wfwALDsw+rI7iaDxR94YAFhS0bGP2zLBh99tXrbP1vmwKcAwuRCngFrdrMVMG+5WoApntb5AF7RBpzf5h+AAgDDAR0Tb1G1mN0KurvL+AB4qo+qI/iIDRe4NQIC/FiOWngMiMAuTFCuV7zIB+GLueMwBnCHxPAPHDHDZceTh/N+4D/t5QABAMsrwMFBdQpXrEFSjm/U8MMMcHYAogABAXoAChhRWMYgS1ESUlAajL0DFo2CbGttWQmAP/4HAgABwAkLIgP5Q+4oQMbfv/0fwBYpUBhMzAefRgE6hibjSUAAIAWmAECBGW5/jsABDZncVjO72tYxv/+ABVIwysMqklrc4gAA0AHARLHYAEH1zdzf/gBMRiKyHMpRkqWH47AARBqhDjKTaFvcon/4AD5EY1QhM4voRkAAGgA4CJYAQtDIw6J8ppDx2AAiMyIBjO3ee9zH/+GBI/NvyAADQAcBEsADBYIweqs4k4sfgAIADAgkqwysyUKlCrRIqQ/wACEUSqy5aKeEb/0JWYE4gmijPjBpr7TQ78AAkQAAQEhUmDIwVtWD8meJumEurCowj2gJxHrWF5hvElDlMS7OasJqgA2Y5AHeoAAgKVzv/oNWfEBRXSABffbEAAE1AAEBS4wAAQAAEA24AE0TAihBEW1YKjTLkAAGgA4HS8fgAIGHBQoVzpjA7v1bsT4PxEgrgFIFpyZIAPcgPZk4xdCozhgB6De+qQEPg0EOzwYwJxxFVGfMGQf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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  DecodedFrame,\n  DecodedAudioChunk,\n  DecodeOptions,\n  BrokerStatus,\n  BrokerStats,\n  Unsubscribe,\n  IRestreamer,\n  IScopedLogger,\n  FrameFormat,\n} from '@camstack/types'\n\nimport { DecoderSessionProxy, type DecoderCapApi } from './decoder-session-proxy.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 { FrameDropper } from './frame-dropper'\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'\nimport type { Sharp } from 'sharp'\n\ntype SharpFn = (input: Buffer | Uint8Array, opts?: Record<string, unknown>) => Sharp\n\nlet cachedSharp: SharpFn | null = null\nasync function getSharp(): Promise<SharpFn> {\n  if (cachedSharp) return cachedSharp\n  const mod = await import('sharp')\n  cachedSharp = mod.default as unknown as SharpFn\n  return cachedSharp\n}\n\n/**\n * In-place R↔B swap on a packed RGB24 buffer. Used by the broker's\n * per-frame conversion cache to satisfy `bgr` subscribers when the\n * decoder's canonical output is `rgb`. Returns a new Buffer so the\n * source frame stays untouched (other subscribers may still need it).\n */\nfunction swapRedBlue(rgb: Buffer): Buffer {\n  const out = Buffer.allocUnsafe(rgb.length)\n  for (let i = 0; i + 2 < rgb.length; i += 3) {\n    out[i] = rgb[i + 2]!\n    out[i + 1] = rgb[i + 1]!\n    out[i + 2] = rgb[i]!\n  }\n  return out\n}\n\nconst DEFAULT_MAX_FPS = 5\nconst DEFAULT_SCALE = 1\n\nconst INITIAL_RECONNECT_DELAY_MS = 1_000\nconst MAX_RECONNECT_DELAY_MS = 30_000\nconst DECODER_TEARDOWN_GRACE_MS = 2_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\ninterface DecodedSubscriber {\n  readonly callback: (frame: DecodedFrame) => void\n  readonly frameDropper: FrameDropper\n  /** Caller-supplied identity — used by `listClients` for diagnostics. */\n  readonly tag: string\n  readonly maxFps: number\n  /** Requested output format. Gray consumers don't need JPEG encoding. */\n  readonly format: FrameFormat\n  readonly subscribedAt: number\n  framesDelivered: number\n  framesDropped: number\n}\n\ninterface AudioSubscriber {\n  readonly callback: (chunk: DecodedAudioChunk) => void\n  readonly tag: string\n  readonly subscribedAt: number\n  chunksDelivered: number\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 — mirrors Scrypted's\n   * `ensureRfcServer`-on-each-`getVideoStream` pattern. `null` for\n   * tests and stand-alone callers; falls back to the source passed\n   * 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  private decoderProxy: DecoderSessionProxy | null = null\n  /** Current output format of the shared decoder session. */\n  private decoderOutputFormat: FrameFormat | null = null\n  /**\n   * Moleculer nodeID of the decoder provider currently hosting\n   * `decoderProxy`. Set from the `{ nodeId }` returned by\n   * `decoderApi.createSession`; used by the broker-manager's\n   * hwaccel-change subscription to decide which brokers need a forced\n   * decoder rotation. Null when no shared decoder session exists.\n   */\n  private decoderNodeId: string | 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  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  private manualStop = false\n  private stopping = false\n  private decoderTeardownTimer: ReturnType<typeof setTimeout> | undefined\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 Set<(packet: EncodedPacket) => void>()\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 Set<(rtpData: Buffer) => void>()\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  private readonly decodedSubscribers = new Map<symbol, DecodedSubscriber>()\n  private readonly audioSubscribers = new Map<symbol, AudioSubscriber>()\n  /** Stored options from first onDecodedFrame call when codec was not yet known */\n  private pendingDecodeOptions: DecodeOptions | undefined\n  /** Pull mode decoder needs to wait for the first keyframe before opening */\n  private pendingPullModeOpen: (() => void) | null = null\n  private firstKeyframeReceived = false\n  /** Retry handle for \"no decoder provider yet\" — addons register in\n   *  arbitrary order, the broker must wait rather than fail. Cancelled\n   *  on decoder attach or broker destroy. */\n  private decoderResolveRetryTimer: ReturnType<typeof setTimeout> | undefined\n  /** Attempt counter for the retry loop above — feeds bounded backoff. */\n  private decoderResolveAttempts = 0\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  /** Debug counters for decoder troubleshooting */\n  private decoderPushCount = 0\n  private decoderFrameCount = 0\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  /** Cached decoder stats — updated by the polling loop, read synchronously by getStats(). */\n  private cachedDecoderStats: import('@camstack/types').DecoderStats | undefined\n\n  /** Tracking flags set synchronously by the RTP depacketizer callback. */\n  private _lastNalKeyframe = false\n  private _lastNalParamSet = false\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\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 — same Scrypted prebuffer-\n    //     mixin contract. Holding the loopback open 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\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.decoderTeardownTimer) {\n      clearTimeout(this.decoderTeardownTimer)\n      this.decoderTeardownTimer = 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.decoderProxy) {\n      await this.destroyDecoder()\n    }\n\n    await this.pipeServer.stop()\n    this.rtspRestreamer.destroy()\n    this.preBuffer.clear()\n\n    this.pendingParamNals = []\n    this.pendingPullModeOpen = null\n    this.firstKeyframeReceived = false\n    this.encodedCallbacks.clear()\n    this.decodedSubscribers.clear()\n    this.audioSubscribers.clear()\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    if (this.decodedSubscribers.size > 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.audioSubscribers.size > 0) return true\n    if (this.rtspRestreamer.getSessionCount() > 0) return true\n    if (this.pipeServer.getClientCount() > 0) return true\n    if (this._preBufferEnabled && this.preBuffer.getDuration() > 0) 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  /**\n   * Pick the decoder's native output format given the current subscriber\n   * mix. Phase 3 collapsed three concerns into this one decision:\n   *\n   *   1. Avoid wasted CPU when subscribers are homogeneous — `gray`-\n   *      only stays `gray` (motion-only camera), `jpeg`-only stays\n   *      `jpeg` (no detection consumer joined yet, broker doesn't have\n   *      to encode either).\n   *   2. Pick the cheapest *canonical* mode for mixed subscribers —\n   *      `rgb` is the universal source for the broker-side conversion\n   *      cache: gray is one luminance pass, jpeg is one sharp encode,\n   *      bgr is a channel swap. Producing JPEG decoder-side and then\n   *      *decoding back* to feed a raw consumer would cost more than\n   *      both consumers combined.\n   *   3. Stay stable when there are no subscribers — defaults to\n   *      `jpeg` so the legacy \"no subscriber yet\" code path keeps\n   *      behaving as it did pre-refactor.\n   */\n  private resolveDecoderFormat(): FrameFormat {\n    if (this.decodedSubscribers.size === 0) return 'jpeg'\n    let allGray = true\n    let allJpeg = true\n    for (const sub of this.decodedSubscribers.values()) {\n      if (sub.format !== 'gray') allGray = false\n      if (sub.format !== 'jpeg') allJpeg = false\n    }\n    if (allGray) return 'gray'\n    if (allJpeg) return 'jpeg'\n    return 'rgb'\n  }\n\n  /**\n   * Check if the decoder's output format needs upgrading because a new\n   * subscriber requires a richer format (e.g. gray → jpeg when detection\n   * joins after motion). Calls updateConfig on the decoder proxy to\n   * reinitialize the scaler without recreating the session.\n   */\n  private maybeUpgradeDecoderFormat(): void {\n    if (!this.decoderProxy || !this.decoderOutputFormat) return\n    const needed = this.resolveDecoderFormat()\n    if (needed === this.decoderOutputFormat) return\n    this.logger?.info('upgrading decoder output format', {\n      meta: { from: this.decoderOutputFormat, to: needed },\n    })\n    this.decoderOutputFormat = needed\n    this.decoderProxy.updateConfig({ outputFormat: needed }).catch((err: unknown) => {\n      this.logger?.warn('decoder format upgrade failed', { meta: { error: errMsg(err) } })\n    })\n  }\n\n  /**\n   * Fan a single decoded frame out to every active subscriber. Each\n   * subscriber may have requested a different `format` than the one the\n   * decoder is currently producing — the per-frame `convertedCache` map\n   * memoises any format the decoder didn't already supply, so multiple\n   * subscribers asking for the same target share the conversion.\n   *\n   * Conversion paths covered today:\n   *   - rgb → jpeg (sharp encode)\n   *   - rgb → bgr  (R/B channel swap, in-process)\n   *   - rgb → gray (BT.601 luminance)\n   *   - rgb → yuv420 (sharp toFormat)\n   *   - gray → jpeg (sharp encode 1ch)\n   *   - jpeg → rgb / bgr / gray / yuv420 — not supported in Phase 3.\n   *     Decoder picks `jpeg` as canonical only when ALL subscribers\n   *     want jpeg, so this branch should never trigger. Such a request\n   *     is logged and the subscriber receives the JPEG passthrough as a\n   *     graceful fallback (caller is expected to re-decode).\n   */\n  private fanoutDecodedFrame(frame: DecodedFrame): void {\n    const convertedCache = new Map<FrameFormat, DecodedFrame | Promise<DecodedFrame>>()\n    convertedCache.set(frame.format, frame)\n    for (const subscriber of this.decodedSubscribers.values()) {\n      if (!subscriber.frameDropper.shouldKeep()) {\n        subscriber.framesDropped++\n        continue\n      }\n      this.deliverConvertedFrame(subscriber, frame, convertedCache)\n    }\n  }\n\n  private deliverConvertedFrame(\n    subscriber: DecodedSubscriber,\n    frame: DecodedFrame,\n    cache: Map<FrameFormat, DecodedFrame | Promise<DecodedFrame>>,\n  ): void {\n    if (subscriber.format === frame.format) {\n      subscriber.callback(frame)\n      subscriber.framesDelivered++\n      return\n    }\n    const cached = cache.get(subscriber.format)\n    if (cached && !(cached instanceof Promise)) {\n      subscriber.callback(cached)\n      subscriber.framesDelivered++\n      return\n    }\n    if (cached instanceof Promise) {\n      cached.then((converted) => {\n        subscriber.callback(converted)\n        subscriber.framesDelivered++\n      }).catch((err: unknown) => {\n        subscriber.framesDropped++\n        this.logger?.warn('frame conversion failed', {\n          meta: { tag: subscriber.tag, target: subscriber.format, error: errMsg(err) },\n        })\n      })\n      return\n    }\n    const conversion = this.convertFrame(frame, subscriber.format)\n    cache.set(subscriber.format, conversion)\n    conversion.then((converted) => {\n      cache.set(subscriber.format, converted)\n      subscriber.callback(converted)\n      subscriber.framesDelivered++\n    }).catch((err: unknown) => {\n      subscriber.framesDropped++\n      this.logger?.warn('frame conversion failed', {\n        meta: { tag: subscriber.tag, target: subscriber.format, error: errMsg(err) },\n      })\n    })\n  }\n\n  private async convertFrame(source: DecodedFrame, target: FrameFormat): Promise<DecodedFrame> {\n    if (source.format === target) return source\n    const sharp = await getSharp()\n    if (source.format === 'rgb' || source.format === 'gray') {\n      const channels = source.format === 'gray' ? 1 : 3\n      const raw = { width: source.width, height: source.height, channels: channels as 1 | 3 }\n      const pipeline = sharp(source.data, { raw })\n      if (target === 'jpeg') {\n        const data = await pipeline.jpeg({ quality: 80, mozjpeg: false }).toBuffer()\n        return { ...source, data, format: 'jpeg' }\n      }\n      if (target === 'bgr') {\n        // Channel swap stays in-process — sharp's extractChannel + join\n        // would require an extra encode/decode, so handle it directly.\n        if (source.format !== 'rgb') {\n          throw new Error(`bgr conversion requires rgb source, got ${source.format}`)\n        }\n        const data = swapRedBlue(source.data)\n        return { ...source, data, format: 'bgr' }\n      }\n      if (target === 'gray') {\n        const data = await pipeline.toColorspace('b-w').raw().toBuffer()\n        return { ...source, data, format: 'gray' }\n      }\n      if (target === 'rgb') {\n        // gray → rgb: replicate the single luminance channel into three\n        // identical planes so consumers expecting RGB24 don't crash.\n        const data = await pipeline.toColorspace('srgb').raw().toBuffer()\n        return { ...source, data, format: 'rgb' }\n      }\n      if (target === 'yuv420') {\n        const data = await pipeline.toColorspace('yuv').raw().toBuffer()\n        return { ...source, data, format: 'yuv420' }\n      }\n    }\n    if (source.format === 'jpeg') {\n      // Decoder only picks `jpeg` when every subscriber wants jpeg,\n      // so a non-jpeg request here is a misconfiguration. Log and pass\n      // the JPEG through so the caller can decide what to do.\n      this.logger?.warn('jpeg-source frame requested as non-jpeg — passthrough', {\n        meta: { target },\n      })\n      return source\n    }\n    throw new Error(`unsupported conversion ${source.format} → ${target}`)\n  }\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\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 (clip creation, late-join)\n    if (packet.type === 'video') {\n      this.preBuffer.push(packet)\n    }\n\n    for (const cb of this.encodedCallbacks) {\n      cb(packet)\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 decoder or opening pull mode.\n      // H264 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        if (this.pendingPullModeOpen) {\n          this.pendingPullModeOpen()\n          this.pendingPullModeOpen = null\n        }\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 decoder after the first keyframe — the decoder needs SPS/PPS to start.\n    // Without this, streams that start mid-GOP silently produce 0 decoded frames.\n    if (packet.type === 'video' && this.firstKeyframeReceived) {\n      if (this.decoderProxy) {\n        this.decoderPushCount++\n        if (this.decoderPushCount === 1 || this.decoderPushCount % 500 === 0) {\n          this.logger?.info('decoder push', {\n            meta: {\n              pushCount: this.decoderPushCount,\n              size: packet.data.length,\n              keyframe: packet.keyframe,\n              decodedCount: this.decoderFrameCount,\n            },\n          })\n        }\n        this.decoderProxy.pushPacket(packet).catch((err: unknown) => {\n          this.logger?.warn('decoder push error', { meta: { error: errMsg(err) } })\n        })\n      }\n    }\n  }\n\n  onEncodedData(callback: (packet: EncodedPacket) => void): Unsubscribe {\n    this.encodedCallbacks.add(callback)\n    if (this.streamingDebug) {\n      this.logger?.info('encoded subscriber added', { meta: { total: this.encodedCallbacks.size } })\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', { meta: { total: this.encodedCallbacks.size } })\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) 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(callback: (rtpData: Buffer) => void): Unsubscribe {\n    this.rtpVideoCallbacks.add(callback)\n    if (this.streamingDebug) {\n      this.logger?.info('rtp video subscriber added', {\n        meta: { total: this.rtpVideoCallbacks.size },\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 },\n        })\n      }\n      this.checkDemand()\n    }\n  }\n\n  getSdpParameterSets(): ReadonlyArray<Buffer> | null {\n    return this.sdpParameterSets\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. Fans\n   * out to every `onVideoRtp` subscriber. Mirrors the dispatch the\n   * native RTSP reader performs in its `onVideoRtp` callback. No-ops\n   * when no subscribers are attached so providers can fire-and-forget.\n   */\n  pushVideoRtp(rtpData: Buffer): void {\n    if (this.stopping) return\n    if (this.rtpVideoCallbacks.size === 0) return\n    for (const cb of this.rtpVideoCallbacks) {\n      try {\n        cb(rtpData)\n      } catch (err) {\n        this.logger?.warn('rtp video subscriber threw', {\n          meta: { error: errMsg(err) },\n        })\n      }\n    }\n  }\n\n  /** Toggle the per-device streaming debug flag. */\n  setStreamingDebug(enabled: boolean): void {\n    this.streamingDebug = enabled\n  }\n\n  onDecodedFrame(callback: (frame: DecodedFrame) => void, options?: DecodeOptions): Unsubscribe {\n    const maxFps = options?.maxFps ?? DEFAULT_MAX_FPS\n    const subscriberId = Symbol('decoded-subscriber')\n    const frameDropper = new FrameDropper(maxFps)\n    const tag = options?.tag\n    if (!tag) {\n      this.logger?.warn(`onDecodedFrame called without tag — listClients will show 'unknown' (Phase 10: every caller must pass options.tag)`)\n    }\n\n    const subscriber: DecodedSubscriber = {\n      callback,\n      frameDropper,\n      tag: tag ?? 'unknown',\n      maxFps,\n      format: options?.format ?? 'jpeg',\n      subscribedAt: Date.now(),\n      framesDelivered: 0,\n      framesDropped: 0,\n    }\n    this.decodedSubscribers.set(subscriberId, subscriber)\n    this.logger?.info('decoded subscriber added', {\n      meta: { tag: subscriber.tag, total: this.decodedSubscribers.size, maxFps, format: subscriber.format },\n    })\n    this.checkDemand()\n\n    // Cancel pending teardown if a new subscriber joins while grace period is active\n    if (this.decoderTeardownTimer) {\n      clearTimeout(this.decoderTeardownTimer)\n      this.decoderTeardownTimer = undefined\n    }\n\n    // Create shared decoder on first subscriber (or reuse existing).\n    // If codec is not yet known (probe in progress), defer creation until codec is detected.\n    if (!this.decoderProxy && this.source) {\n      const codec = this.detectedCodec ?? this.source.videoCodec\n      if (codec) {\n        this.createSharedDecoderSession(options)\n      } else {\n        this.pendingDecodeOptions = options\n        this.logger?.info('Decoder creation deferred — waiting for codec probe')\n      }\n    } else if (this.decoderProxy) {\n      // Decoder already running — check if new subscriber requires a format upgrade\n      this.maybeUpgradeDecoderFormat()\n    }\n\n    return () => {\n      this.decodedSubscribers.delete(subscriberId)\n      this.logger?.info('decoded subscriber removed', { meta: { total: this.decodedSubscribers.size } })\n      this.checkDemand()\n\n      if (this.decodedSubscribers.size === 0 && this.decoderProxy) {\n        this.logger?.info('last decoded subscriber left — scheduling decoder teardown', {\n          meta: { graceMs: DECODER_TEARDOWN_GRACE_MS },\n        })\n        this.decoderTeardownTimer = setTimeout(() => {\n          this.decoderTeardownTimer = undefined\n          if (this.decodedSubscribers.size === 0 && this.decoderProxy) {\n            this.logger?.info('decoder teardown — no subscribers reconnected')\n            this.destroyDecoder()\n          }\n        }, DECODER_TEARDOWN_GRACE_MS)\n      }\n    }\n  }\n\n  onDecodedAudioChunk(\n    callback: (chunk: DecodedAudioChunk) => void,\n    options?: { readonly tag?: string },\n  ): Unsubscribe {\n    const tag = options?.tag\n    if (!tag) {\n      this.logger?.warn(`onDecodedAudioChunk called without tag — listClients will show 'unknown' (Phase 10)`)\n    }\n    const subscriberId = Symbol('audio-subscriber')\n    const subscriber: AudioSubscriber = {\n      callback,\n      tag: tag ?? 'unknown',\n      subscribedAt: Date.now(),\n      chunksDelivered: 0,\n    }\n    this.audioSubscribers.set(subscriberId, subscriber)\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\n    return () => {\n      this.audioSubscribers.delete(subscriberId)\n      this.checkDemand()\n    }\n  }\n\n  getStats(): BrokerStats {\n    const decoderStats = this._suspended ? undefined : this.cachedDecoderStats\n\n    return {\n      status: this._suspended ? 'idle' : this._status,\n      inputFps: decoderStats?.inputFps ?? this.encodedInputFps,\n      decodeFps: decoderStats?.outputFps ?? 0,\n      encodedSubscribers: this.encodedCallbacks.size,\n      decodedSubscribers: this.decodedSubscribers.size,\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.decoderNodeId,\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 pipeClients: number\n    readonly encodedSubscribers: number\n  } {\n    return {\n      rtsp: this.rtspRestreamer.listSessionInfos(),\n      decoded: [...this.decodedSubscribers.values()].map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        maxFps: s.maxFps,\n        framesDelivered: s.framesDelivered,\n        framesDropped: s.framesDropped,\n      })),\n      audio: [...this.audioSubscribers.values()].map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        chunksDelivered: s.chunksDelivered,\n      })),\n      pipeClients: this.pipeServer.getClientCount(),\n      encodedSubscribers: this.encodedCallbacks.size,\n    }\n  }\n\n  /**\n   * Force-disconnect a single consumer. Matches by channel:\n   *   - `rtsp`: by `sessionId` (calls `RtspRestreamer.killSession`).\n   *   - `decoded` / `audio`: by `tag` — drops every subscriber whose tag\n   *     matches, then re-evaluates demand so the decoder can wind down\n   *     if nothing else is holding it.\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    const subscribers = channel === 'decoded' ? this.decodedSubscribers : this.audioSubscribers\n    const victims: symbol[] = []\n    for (const [key, sub] of subscribers) {\n      if (sub.tag === handle) victims.push(key)\n    }\n    if (victims.length === 0) return false\n    for (const key of victims) subscribers.delete(key)\n    this.logger?.info('subscribers force-disconnected', {\n      meta: { channel, handle, killed: victims.length, remaining: subscribers.size },\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          // Create deferred decoder session if subscribers were waiting\n          if (!this.decoderProxy && this.decodedSubscribers.size > 0 && this.pendingDecodeOptions !== undefined) {\n            this.logger?.info('native: codec detected — creating deferred decoder', {\n              meta: { codec: track.codec },\n            })\n            this.createSharedDecoderSession(this.pendingDecodeOptions)\n            this.pendingDecodeOptions = undefined\n          }\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\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          // 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 of this.rtpVideoCallbacks) {\n              try {\n                cb(rtpData)\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              for (const sub of this.audioSubscribers.values()) {\n                sub.callback(chunk)\n                sub.chunksDelivered++\n              }\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          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 of this.encodedCallbacks) {\n                cb(packet)\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  // ── 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  private destroyDecoder(): Promise<void> {\n    if (this.decoderResolveRetryTimer) {\n      clearTimeout(this.decoderResolveRetryTimer)\n      this.decoderResolveRetryTimer = undefined\n    }\n    this.decoderResolveAttempts = 0\n    const proxy = this.decoderProxy\n    this.decoderProxy = null\n    this.decoderNodeId = null\n    this.cachedDecoderStats = undefined\n    return proxy?.destroy() ?? Promise.resolve()\n  }\n\n  /**\n   * Moleculer nodeID of the decoder provider currently owning this broker's\n   * shared session. `null` when no session is active. Used by the\n   * broker-manager's `rotateDecodersOnNode` method to drop only the\n   * sessions affected by a per-agent event (e.g. hwaccel override change).\n   */\n  getDecoderNodeId(): string | null {\n    return this.decoderNodeId\n  }\n\n  /**\n   * Force the current shared decoder session to tear down. The next\n   * subscriber will trigger a fresh `createSharedDecoderSession` which\n   * re-pulls the latest per-agent preferences (hwaccel backend, …). Safe\n   * to call when there's no active session — resolves to a no-op.\n   */\n  async rotateDecoderSession(reason: string): Promise<void> {\n    if (!this.decoderProxy) return\n    this.logger?.info('rotating decoder session', { meta: { reason, decoderNodeId: this.decoderNodeId } })\n    await this.destroyDecoder()\n    // If subscribers are still attached, rebuild immediately so streaming\n    // resumes without waiting for a new subscribe event.\n    if (this.decodedSubscribers.size > 0 && this.source) {\n      const codec = this.detectedCodec ?? this.source.videoCodec\n      if (codec) {\n        this.createSharedDecoderSession(this.pendingDecodeOptions).catch((err: unknown) => {\n          this.logger?.warn('decoder session rebuild after rotation failed', { meta: { error: errMsg(err) } })\n        })\n      }\n    }\n  }\n\n  /**\n   * Poll the decoder resolver until a provider for `codec` shows up, then\n   * build the session. Addons register asynchronously and out-of-order:\n   * the broker must NOT fail when the decoder addon hasn't yet finished\n   * spawning its isolated process — it must wait.\n   *\n   * Bounded backoff: 500ms → 1s → 2s → 4s → 8s (capped), with a hard\n   * ceiling of ~60s total wait. If the decoder still isn't registered\n   * after that, log once at error level and stop retrying — at that\n   * point something is genuinely wrong with the addon config.\n   *\n   * The first failure is logged at debug so a healthy boot (decoder\n   * arriving within the first few hundred ms) doesn't pollute the log\n   * with noisy errors.\n   */\n  private scheduleDecoderResolveRetry(codec: string): void {\n    if (this.decoderResolveRetryTimer) return\n    const attempt = this.decoderResolveAttempts++\n    const backoffMs = Math.min(500 * Math.pow(2, attempt), 8000)\n    const cumulativeMs = (Math.pow(2, this.decoderResolveAttempts) - 1) * 500\n    if (cumulativeMs > 60000) {\n      this.logger?.error(\n        'No decoder provider — giving up. Check that addon-decoder-nodeav or addon-decoder-ffmpeg is installed and enabled.',\n        { meta: { codec, attempts: this.decoderResolveAttempts } },\n      )\n      this.decoderResolveAttempts = 0\n      return\n    }\n    if (attempt === 0) {\n      this.logger?.debug(\n        'Decoder not yet registered — will retry (addon boot race)',\n        { meta: { codec } },\n      )\n    }\n    this.decoderResolveRetryTimer = setTimeout(() => {\n      this.decoderResolveRetryTimer = undefined\n      // Only retry if we still have subscribers and no session yet — a\n      // teardown between schedule and fire is valid and should no-op.\n      if (this.decoderProxy) return\n      if (this.decodedSubscribers.size === 0) {\n        this.decoderResolveAttempts = 0\n        this.pendingDecodeOptions = undefined\n        return\n      }\n      this.createSharedDecoderSession(this.pendingDecodeOptions)\n    }, backoffMs)\n  }\n\n\n  private destroyNativeClient(): void {\n    if (this.nativeClient) {\n      this.nativeClient.destroy()\n      this.nativeClient = null\n    }\n    this.resetRtpPipeline()\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  }\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.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  private async createSharedDecoderSession(options?: DecodeOptions): Promise<void> {\n    const codec = this.detectedCodec ?? this.source?.videoCodec ?? 'h264'\n\n    if (!this.decoderApi) {\n      this.logger?.warn('no decoder API available — decoded frames will not be produced')\n      return\n    }\n\n    // Check if the decoder supports this codec before creating a session.\n    const supported = await this.decoderApi.supportsCodec({ codec })\n    if (!supported) {\n      // Decoder addon may not be registered yet (boot race).\n      this.pendingDecodeOptions = options\n      this.scheduleDecoderResolveRetry(codec)\n      return\n    }\n\n    // A provider is now available — clear any pending retry so we don't\n    // double-create the session on the next tick.\n    if (this.decoderResolveRetryTimer) {\n      clearTimeout(this.decoderResolveRetryTimer)\n      this.decoderResolveRetryTimer = undefined\n    }\n    this.decoderResolveAttempts = 0\n\n    // Shared decoder runs at unlimited FPS — each subscriber throttles locally.\n    // Use resolveDecoderFormat() to start with the cheapest format that\n    // satisfies all current subscribers (e.g. gray when only motion is active).\n    const resolvedFormat = this.resolveDecoderFormat()\n    // Stream-broker `deviceId` is actually the brokerId (`<numeric>/<profile>`).\n    // Parse out the numeric for the cap config so decoder logs land with\n    // a clean integer `deviceId` tag — the full brokerId goes through `tag`.\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    const config = {\n      codec,\n      maxFps: 0,\n      outputFormat: resolvedFormat,\n      scale: options?.scale ?? DEFAULT_SCALE,\n      ...(Number.isFinite(numericDeviceId) ? { deviceId: numericDeviceId } : {}),\n      tag: `broker:${this.deviceId}`,\n    }\n    this.decoderOutputFormat = resolvedFormat\n\n    const { sessionId, nodeId } = await this.decoderApi.createSession(config)\n    const proxy = new DecoderSessionProxy(this.decoderApi, sessionId)\n    this.decoderProxy = proxy\n    this.decoderNodeId = nodeId\n\n    // Fan-out decoded frames to all subscribers with per-subscriber\n    // throttling. In Phase 3 the decoder produces one *canonical*\n    // output (gray / rgb / jpeg per `resolveDecoderFormat`) and the\n    // broker derives any other format on demand via a per-frame\n    // conversion cache, so that an `rgb`-source frame fanned out to\n    // five `jpeg` subscribers performs exactly one sharp encode and\n    // pushes the same buffer to each. Subscribers that match the\n    // canonical format short-circuit the cache entirely.\n    const onFrame = (frame: DecodedFrame) => {\n      this.decoderFrameCount++\n      if (this.decoderFrameCount === 1) {\n        this.logger?.info('first decoded frame', {\n          meta: { width: frame.width, height: frame.height, format: frame.format },\n        })\n      }\n      // Update cached stats periodically (every 30 frames)\n      if (this.decoderFrameCount % 30 === 0) {\n        proxy.getStats().then((stats) => {\n          this.cachedDecoderStats = stats\n        }).catch(() => { /* stats refresh failure is non-fatal */ })\n      }\n      this.fanoutDecodedFrame(frame)\n    }\n\n    // Start the polling loop in the background — it runs until stopPolling or destroy\n    proxy.startPolling(onFrame).catch((err: unknown) => {\n      this.logger?.warn('decoder polling error', { meta: { error: errMsg(err) } })\n    })\n\n    // Pull mode: decoder reads from the broker's local pipe server (no duplicate RTSP connection)\n    // Must wait for the first keyframe so the pipe contains SPS/PPS for the decoder.\n    const info = await this.decoderApi.getInfo()\n    if (info.isPullMode) {\n      const localUrl = this.pipeServer.getUrl()\n      const isHevc = codec === 'h265' || codec === 'hevc'\n      const inputUrl = `${localUrl}?format=${isHevc ? 'hevc' : 'h264'}`\n\n      const doOpen = () => {\n        this.logger?.info('Pull mode decoder: opening local pipe', {\n          meta: { localUrl, codec },\n        })\n        proxy.openStream(inputUrl).catch((err: unknown) => {\n          this.logger?.error('Pull mode decoder failed', { meta: { error: errMsg(err) } })\n        })\n      }\n\n      if (this.firstKeyframeReceived) {\n        doOpen()\n      } else {\n        this.logger?.info('Pull mode decoder: waiting for first keyframe before opening')\n        this.pendingPullModeOpen = doOpen\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 `audioSubscribers` unchanged.\n      for (const sub of this.audioSubscribers.values()) {\n        sub.callback(chunk)\n        sub.chunksDelivered++\n      }\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 of this.encodedCallbacks) {\n            cb(packet)\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 * 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\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(brokerId: string, restreamer: RtspRestreamer, existingToken?: string): 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    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  }\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\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      const sdp = isMuted\n        ? (restreamer.getMutedSdp() ?? restreamer.getSdp() ?? '')\n        : (restreamer.getSdp() ?? '')\n      const session = new RtspSession(socket, sdp, () => {\n        restreamer.removeSession(session.getSessionId())\n      }, isMuted)\n      restreamer.addSession(session)\n\n      // Re-inject the buffered data so the session processes the first request\n      socket.unshift(Buffer.from(requestBuffer, 'ascii'))\n    }\n\n    socket.on('data', onData)\n    socket.on('error', () => {})\n  }\n}\n","import type { IScopedLogger } 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      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      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","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} 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 { RtspRestreamProviderImpl, type RtspRestreamEntry } from '../rtsp/rtsp-restream-provider.js'\nimport { CAM_PROFILE_ORDER, DeviceFeature, EventCategory, errMsg, RingBuffer } from '@camstack/types'\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\ninterface 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\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 broker id from a `(deviceId, camStreamId)` tuple. */\nfunction brokerIdFor(deviceId: number, camStreamId: string): string {\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  /** 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  // 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  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  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  // ── 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      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      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      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  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    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  }\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  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  }): Promise<{ success: true }> {\n    const { deviceId, camStreamId, kind, url, codec, resolution, fps, label, metadata } = input\n\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    // 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.\n    const urlRefreshed = existing !== undefined\n      && url !== undefined\n      && existing.url !== url\n    // Re-publish updates the feature snapshot — battery detection can\n    // happen post-creation (live probe), so keeping the latest signal\n    // matters more than the first.\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\n    // Re-run auto-assignment when the device has never had a manual\n    // assignment AND (a) profileMap is empty, or (b) auto flag is true.\n    // Re-publish of a known camStreamId with the SAME id keeps map stable.\n    const current = this.assignments.get(deviceId)\n    const streams = [...streamMap.values()].sort((a, b) => a.order - b.order)\n    const isAuto = current?.auto ?? true\n    let newMap: Partial<Record<CamProfile, string>>\n    if (!current || (isAuto && this.wouldChangeAutoMap(current.map, streams))) {\n      newMap = this.computeInitialAssignment(streams)\n    } else {\n      newMap = current.map\n    }\n\n    // Snapshot whether the just-published stream is already in a profile\n    // slot before applyAssignmentUpdate runs. If it is — and the slot\n    // still points at it after the update (i.e. no transition will fire\n    // demand naturally) — we need to re-emit demand explicitly. A\n    // re-publish from a freshly-restarted provider wipes its in-memory\n    // demand state, so the broker must re-tell it which streams are\n    // wanted. Demand handlers are idempotent on the provider side\n    // (`if (active.has(id)) return`), so re-emitting is always safe.\n    const prePublishProfile = PUSH_KINDS.has(kind)\n      ? findProfileForStream(current?.map ?? {}, camStreamId)\n      : null\n    const wasAlreadyAssigned = prePublishProfile !== null && existing !== undefined\n\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: isAuto, reason: '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    this.emitCamStreamsChanged(deviceId)\n    return { success: true as const }\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  // ── 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  // ── Cap methods: broker runtime (stats + client inventory) ──────────\n\n  async getBroker(input: { brokerId: string }): Promise<IStreamBroker | null> {\n    return this.brokers.get(input.brokerId) ?? null\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: [], 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      const nodeId = broker.getDecoderNodeId()\n      if (!nodeId) continue\n      const brokerAgent = nodeId.includes('/') ? nodeId.split('/')[0]! : nodeId\n      if (brokerAgent !== agentNodeId) continue\n      try {\n        await broker.rotateDecoderSession(reason)\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    const stopPromises = [...this.brokers.values()].map((broker) => broker.stop())\n    await Promise.all(stopPromises)\n    this.brokers.clear()\n    this.streamHealthByBroker.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)\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)\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\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  // ── 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 \"Assignment\" — three profile→stream selectors ──────\n    const assignmentTab: ConfigSubTabDefinitionWithValue = {\n      id: 'assignment',\n      label: 'Assignment',\n      fields: 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 \"${label}\" profile slot`,\n        options: streamOptions,\n        span: 2 as const,\n        value: assignment.map[profile] ?? '',\n      })),\n    }\n\n    // ── One sub-tab per stream source with its broker settings ─────\n    // Every published cam stream owns a broker for as long as the\n    // source is published — there is no \"Active\" toggle. Pre-buffer\n    // is OFF by default and opt-in per stream.\n    const streamTabs: ConfigSubTabDefinitionWithValue[] = streams.map((stream) => {\n      const camStreamId = stream.camStreamId\n      const brokerId = brokerIdFor(deviceId, camStreamId)\n      const profile = findProfileForStream(assignment.map, camStreamId)\n      const rtspEnabled = this.rtspProvider.isEnabled(brokerId)\n      const rtspEntry = this.rtspProvider.getEntry(brokerId)\n      const perStream = override?.preBuffer?.[camStreamId]\n      const preBufferEnabled = perStream?.enabled ?? false\n      const preBufferSec = perStream?.seconds ?? this.defaultPreBufferSec\n\n      const fields: ConfigFieldWithValue[] = [\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: `rtspEnabled:${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 streamBrokerSection: ConfigSectionWithValues = {\n      id: 'stream-broker-streams',\n      title: '',\n      tab: 'stream-broker',\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: [assignmentTab, ...streamTabs],\n        },\n      ],\n    }\n\n    const diagnosticsSection: ConfigSectionWithValues = {\n      id: 'stream-broker-diagnostics',\n      title: 'Diagnostics',\n      tab: 'stream-broker',\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    }\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(`${deviceId}/`)) {\n            broker.setStreamingDebug(next === true)\n          }\n        }\n        continue\n      }\n    }\n\n    if (overrideDirty) {\n      if (\n        nextOverride.preBufferSecOverride === undefined &&\n        (!nextOverride.preBuffer || Object.keys(nextOverride.preBuffer).length === 0) &&\n        nextOverride.streamingDebug === undefined\n      ) {\n        this.deviceOverrides.delete(deviceId)\n      } else {\n        this.deviceOverrides.set(deviceId, nextOverride)\n      }\n      this.deviceOverridePersister?.(this.deviceOverrides)\n    }\n    return { success: true as const }\n  }\n\n  // ── Private helpers ─────────────────────────────────────────────────\n\n  private nextOrder(deviceId: number): number {\n    const next = (this.streamOrderSeq.get(deviceId) ?? 0) + 1\n    this.streamOrderSeq.set(deviceId, next)\n    return next\n  }\n\n  private toCameraStream(s: CameraStreamInternal): CameraStream {\n    const { order: _order, deviceFeatures: _df, ...rest } = s\n    return rest\n  }\n\n  private deviceHasFeature(deviceId: number, feature: string): boolean {\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) return false\n    for (const s of streamMap.values()) {\n      if (s.deviceFeatures.includes(feature)) return true\n    }\n    return false\n  }\n\n  private labelForDropdown(s: CameraStreamInternal): string {\n    const base = s.label ?? s.camStreamId\n    if (s.resolution) return `${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  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  }\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      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. Mirrors\n    // Scrypted's per-`getVideoStream()` `ensureRfcServer` resolution.\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    this.rtspServer.registerRestreamer(brokerId, broker.getRtspRestreamer(), existingToken)\n    if (!existingToken) this.rtspProvider.persistTokens()\n\n    await this.fanOutRegistration(deviceId, brokerId, cam, broker)\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.rtspProvider.persistTokens()\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    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) : `${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","/**\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 * 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  const replacements = await Promise.all(\n    [...hosts].map(async (host) => {\n      try {\n        const { address } = await dnsLookup(host, { family: 4 })\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', { 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 * Ported from Scrypted (BSD-2-Clause):\n *   https://github.com/koush/scrypted/blob/main/plugins/homekit/src/types/camera/jitter-buffer.ts\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 * Ported from Scrypted (BSD-2-Clause):\n *   https://github.com/koush/scrypted/blob/main/plugins/webrtc/src/h265-packetizer.ts\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 * 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 } from \"./h264-utils.js\";\nimport { convertH265ToAnnexB, isH265IrapAccessUnit } from \"./h265-utils.js\";\nimport { resolveMdnsCandidatesInSdp } from \"./mdns-resolve.js\";\nimport { H265Repacketizer, type H265CodecInfo } from \"./h265-repacketizer.js\";\nimport type {\n  WeriftModule,\n  WeriftPeerConnection,\n  WeriftMediaStreamTrack,\n  WeriftRtpSender,\n  WeriftTransceiver,\n  WeriftPcOptions,\n  WeriftIceServer,\n} from \"./werift-types.js\";\nimport { errMsg } from '@camstack/types'\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    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// 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  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  /** Callback for RTCP stats (called every ~3s). */\n  onStats?: (stats: SessionStats) => void;\n  /** Enable verbose frame/RTP logging. */\n  debug?: boolean;\n  /** Source video codec. Determines SDP negotiation + RTP packetization. */\n  sourceCodec?: 'H264' | 'H265';\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 source: FrameSource;\n  private readonly logger: Logger;\n  private readonly intercom: AdaptiveSessionOptions[\"intercom\"];\n  private readonly iceConfig: AdaptiveSessionOptions[\"iceConfig\"];\n  private readonly onStats: AdaptiveSessionOptions[\"onStats\"];\n  debug: boolean;\n  private readonly sourceCodec: 'H264' | 'H265';\n  /** Codec actually negotiated with the browser after SDP answer. */\n  private negotiatedCodec: 'H264' | 'H265' = 'H264';\n  /** True when source is H.265 but browser negotiated H.264 — needs transcode. */\n  get needsTranscode(): boolean { return this.sourceCodec === 'H265' && this.negotiatedCodec === 'H264' }\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  /** RTP sequence number counter (must increment per packet). */\n  private videoSeqNum = 0;\n  private audioSeqNum = 0;\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  private lastKeyframeRtpTs = 0;\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 — leave headroom for SRTP auth tag. */\n  private static readonly H265_REPACKETIZER_MTU = 1180;\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  constructor(options: AdaptiveSessionOptions) {\n    this.sessionId = options.sessionId;\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.createdAt = Date.now();\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    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\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      parameters: \"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\",\n      rtcpFeedback,\n    });\n    // H.265 fmtp — Scrypted ships the H.264-shaped fmtp string for H.265\n    // and it works. 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 the value Scrypted uses for parity with a known-\n    // good implementation.\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 required for BUNDLE demuxing and congestion control.\n      // Without sdes:mid, browsers cannot demux incoming RTP on a BUNDLE'd connection.\n      headerExtensions: {\n        video: [\n          { uri: \"urn:ietf:params:rtp-hdrext:sdes:mid\" },\n          { uri: \"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\" },\n          { uri: \"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\" },\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.iceConfig?.portRange) pcOptions.icePortRange = this.iceConfig.portRange;\n    if (this.iceConfig?.additionalHostAddresses?.length) {\n      pcOptions.iceAdditionalHostAddresses = [...this.iceConfig.additionalHostAddresses];\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, state } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\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    // Wait for ICE gathering to complete so the offer includes candidates.\n    // Without this, the offer has no candidates and the client can't connect.\n    await new Promise<void>((resolve) => {\n      if (this.pc?.iceGatheringState === \"complete\") { resolve(); return; }\n      this.pc?.iceGatheringStateChange.subscribe((state: string) => {\n        if (state === \"complete\") resolve();\n      });\n      // Timeout: don't wait forever if STUN is unreachable\n      setTimeout(resolve, 5000);\n    });\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.\n    const answerVideoCodec = resolvedSdp.match(/a=rtpmap:\\d+ (H264|H265)\\/90000/i)?.[1]?.toUpperCase()\n    if (answerVideoCodec === 'H264' || answerVideoCodec === 'H265') {\n      this.negotiatedCodec = answerVideoCodec as 'H264' | 'H265'\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    // 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      while (dtlsTransport.state !== \"connected\" && Date.now() < deadline) {\n        await new Promise<void>((r) => setTimeout(r, 100));\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(clientOffer: { sdp: string; type: \"offer\" }): Promise<{ sdp: string; type: \"answer\" }> {\n    const { werift, pcOptions } = await this.buildPcOptions();\n\n    this.pc = new werift.RTCPeerConnection(pcOptions);\n\n    this.pc.iceConnectionStateChange.subscribe((state: string) => {\n      this.logger.debug('ICE state', { meta: { phase: 'session', sessionId: this.sessionId, state } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\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    // 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    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 tracks.\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        this.videoTrack = new werift.MediaStreamTrack({ kind: \"video\" });\n        await t.sender.replaceTrack(this.videoTrack);\n      } else if (kind === \"audio\" && !this.audioTrack) {\n        this.audioTrack = new werift.MediaStreamTrack({ kind: \"audio\" });\n        await t.sender.replaceTrack(this.audioTrack);\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      this.pc.addTransceiver(this.videoTrack, { direction: \"sendonly\" });\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      this.pc.addTransceiver(this.audioTrack, { direction: \"sendonly\" });\n    }\n\n    const answerDesc = await this.pc.createAnswer();\n    await this.pc.setLocalDescription(answerDesc);\n    this.state = \"connecting\";\n    this.logger.info('WHEP answer created', { meta: { phase: 'session', sessionId: this.sessionId } });\n\n    this.startFeedingFrames();\n\n    return { sdp: answerDesc.sdp, type: \"answer\" };\n  }\n\n  /** Add ICE candidate. */\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   * 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    if (this.negotiatedCodec !== 'H265') return;\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      AdaptiveSession.H265_REPACKETIZER_MTU,\n      codecInfo,\n    );\n  }\n\n  /**\n   * Forward a source RTP video packet (raw on-wire bytes) through the\n   * H.265 repacketizer to the browser.  Used by the broker's H.265\n   * direct-RTP subscription — bypasses the AnnexB→writeVideoNals\n   * path entirely.\n   *\n   * Drops everything until the SDP answer has been negotiated\n   * (`negotiatedCodec`/`videoSender` populated) and silently no-ops on\n   * non-H.265 sessions.\n   */\n  forwardSourceRtpVideo(rtpData: Buffer): void {\n    if (this.closed) return;\n    if (this.negotiatedCodec !== 'H265') return;\n    if (!this.videoSender || !_werift) return;\n    const werift = _werift;\n    this.ensureH265Repacketizer();\n    const rep = this.h265Repacketizer!;\n\n    let srcPkt;\n    try {\n      srcPkt = werift.RtpPacket.deSerialize(rtpData);\n    } catch (err) {\n      this.logger.warn('H265 RTP deserialize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err), len: rtpData.length },\n      });\n      return;\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 ?? 97;\n\n    let outPkts;\n    try {\n      outPkts = rep.repacketize(srcPkt);\n    } catch (err) {\n      this.logger.warn('H265 repacketize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) },\n      });\n      return;\n    }\n\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        this.rtpPacketsSent++;\n        if (this.debug && (this.rtpPacketsSent === 1 || this.rtpPacketsSent % 500 === 0)) {\n          this.logger.info('H265 source-RTP forwarded', {\n            meta: { phase: 'session', sessionId: this.sessionId, count: this.rtpPacketsSent },\n          });\n        }\n      } catch (err) {\n        if (this.rtpPacketsSent <= 10) {\n          this.logger.error('sendRtp (h265 forward) error', {\n            meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) },\n          });\n        }\n      }\n    }\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    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    // Sessions running the H.265 repacketizer forward source RTP\n    // through it — the upstream camera will respond to PLI by emitting\n    // a fresh IDR access unit on its own. Re-sending a cached one\n    // through writeVideoNals would collide with the repacketizer's\n    // outbound RTP stream. Push-mode H.265 (no repacketizer) keeps the\n    // cached-keyframe re-send path.\n    if (this.h265Repacketizer) return;\n\n    const now = Date.now();\n    if (now - this.lastPliResendAt < AdaptiveSession.PLI_RESEND_COOLDOWN_MS) return;\n    this.lastPliResendAt = now;\n\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    // Re-send all NALs via writeVideoNals — same path the feed loop\n    // uses for keyframes (single-NAL packets for VPS/SPS/PPS, FU\n    // fragmentation for IDR slice). All share the same RTP timestamp;\n    // marker bit set only on the last FU of the last NAL.\n    this.writeVideoNals(this.lastKeyframeNals, this.lastKeyframeRtpTs, 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      let videoTimestampBase: number | null = null;\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            if (videoTimestampBase === null) videoTimestampBase = frame.timestampMicros;\n            const rtpTs = Math.floor(\n              ((frame.timestampMicros - videoTimestampBase) * 90000) / 1_000_000,\n            ) >>> 0;\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. Scrypted does the same and 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                    this.lastKeyframeRtpTs = rtpTs\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) this.lastSps = Buffer.from(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                  this.lastKeyframeRtpTs = rtpTs;\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            // For G.711 (PCMU/PCMA): timestamp increments by frame size (160 samples per 20ms)\n            // Using a simple counter is more reliable than wall-clock based timestamps\n            if (audioTimestampBase === null) audioTimestampBase = 0;\n            audioTimestampBase = ((audioTimestampBase as number) + frame.data.length) >>> 0;\n            const rtpTs = audioTimestampBase as number;\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  /** Max RTP payload size (MTU 1200 to stay under typical network MTU). */\n  /**\n   * Transcode feed — source is H.265 but browser only supports H.264.\n   * Pipes raw H.265 Annex-B to ffmpeg stdin, reads H.264 from stdout.\n   */\n  private startTranscodeFeed(signal: AbortSignal): void {\n    const { spawn } = require('node:child_process') as typeof import('node:child_process')\n\n    this.logger.info('Starting H.265→H.264 transcode feed', {\n      meta: { phase: 'session', sessionId: this.sessionId },\n    })\n\n    const ff = spawn('ffmpeg', [\n      '-hide_banner', '-loglevel', 'error',\n      '-f', 'hevc', '-i', 'pipe:0',\n      '-c:v', 'libx264', '-preset', 'ultrafast', '-tune', 'zerolatency',\n      '-profile:v', 'baseline', '-g', '30', '-bf', '0',\n      '-an', '-f', 'h264', 'pipe:1',\n    ], { stdio: ['pipe', 'pipe', 'pipe'] })\n\n    signal.addEventListener('abort', () => {\n      if (!ff.killed) ff.kill('SIGTERM')\n    })\n\n    ff.on('exit', (code) => {\n      this.logger.info('Transcode ffmpeg exited', { meta: { phase: 'session', sessionId: this.sessionId, code } })\n    })\n    ff.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    // Parse H.264 output from ffmpeg and send via RTP\n    let pendingBuf = Buffer.alloc(0)\n    let rtpTs = 0\n    const FRAME_INTERVAL_90K = 3000 // 30fps → 90kHz ticks\n\n    ff.stdout.on('data', (chunk: Buffer) => {\n      pendingBuf = Buffer.concat([pendingBuf, chunk])\n      // Look for keyframe boundaries to emit complete access units\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 completedNals = splitAnnexBToNals(complete)\n      if (completedNals.length === 0) return\n\n      rtpTs = (rtpTs + FRAME_INTERVAL_90K) >>> 0\n      this.writeVideoNals(completedNals, rtpTs, 'H264')\n    })\n\n    // Feed source H.265 data 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 (!ff.stdin.destroyed) {\n              ff.stdin.write(mediaFrame.frame.data)\n            }\n          } else if (mediaFrame.type === 'audio') {\n            // Audio passthrough (no transcode needed)\n            const frame = mediaFrame.frame\n            this.writeAudio(frame.data, frame.timestampMicros, 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 (!ff.stdin.destroyed) ff.stdin.end()\n        if (!this.closed) void this.close()\n      }\n    })()\n  }\n\n  private static readonly MAX_RTP_PAYLOAD = 1200;\n\n  private rtpPacketsSent = 0;\n  private rtpLogCounter = 0;\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    // 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 (like Scrypted'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        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    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    } catch (err) {\n      this.logger.debug('Audio write error', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) } });\n    }\n  }\n\n  // -----------------------------------------------------------------------\n  // RTCP stats collection\n  // -----------------------------------------------------------------------\n\n  private startStatsCollection(): void {\n    if (this.statsTimer || !this.onStats) 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 || !this.onStats) return;\n\n    try {\n      // werift exposes getReceivers() and getSenders() on RTCPeerConnection.\n      // For video sender stats, we check RTCP RR feedback from the receiver.\n      const senders = this.pc.getSenders?.() ?? [];\n\n      for (const sender of senders) {\n        const track = sender.track;\n        if (!track || track.kind !== \"video\") continue;\n\n        // werift tracks RTCP receiver reports via sender.rtcpReceiveReport\n        // or via the transport's RTCP event. The exact API depends on werift version.\n        // Fallback: use werift's built-in stats if available.\n        const report = sender.lastReceiverReport ?? sender.rtcpReport;\n        if (!report) continue;\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        const packetLoss = fractionLost / 256; // Fraction lost is 0–255\n\n        this.onStats({\n          sessionId: this.sessionId,\n          packetLoss,\n          jitterMs: jitter,\n          rttMs: rtt * 1000, // seconds → ms\n          packetsReceived: 0, // Not available from sender side\n          packetsLost,\n          timestamp: Date.now(),\n        });\n        return; // One video sender is enough\n      }\n\n      // Fallback: no RTCP report available yet — emit zeros\n      // (Client-reported stats will supplement this)\n    } catch {\n      // RTCP not yet available — normal during early connection\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","/**\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 } from '@camstack/types'\nimport { errMsg } from '@camstack/types'\nimport { AdaptiveSession, type IceServerEntry } from './session.js'\nimport type { FrameSource, MediaFrame } from './types.js'\nimport { convertH264ToAnnexB } from './h264-utils.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\ninterface SessionEntry {\n  readonly session: AdaptiveSession\n  readonly brokerId: string\n  /** Unsubscribe from broker's onEncodedData. */\n  readonly unsubBroker: Unsubscribe\n  /**\n   * Unsubscribe from broker's `onVideoRtp` for H.265 sessions that\n   * forward source RTP through `H265Repacketizer`. `null` for H.264\n   * sessions (which use the AnnexB feed-loop path).\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}\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\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: 'high' | 'mid' | 'low' | 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\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 (`'high' | 'mid' | 'low'`)\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) => 'high' | 'mid' | 'low' | null) | null = null\n  /** sessionId → session entry. */\n  private readonly sessions = new Map<string, SessionEntry>()\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  }\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) => 'high' | 'mid' | 'low' | 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 } = {},\n  ): Promise<{ sessionId: string; sdpOffer: string }> {\n    if (this.stopped) throw new Error('Server stopped')\n\n    const brokerId = this.resolveBrokerId(streamId, hints)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) throw new Error(`No broker for stream \"${streamId}\" (resolved: \"${brokerId}\")`)\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 deviceTags = slashIdx > 0\n      ? {\n          deviceId: Number.parseInt(brokerId.slice(0, slashIdx), 10),\n          camStreamId: brokerId.slice(slashIdx + 1),\n        }\n      : { deviceId: -1, camStreamId: brokerId }\n    const sessionLogger = this.logger.withTags(deviceTags)\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    const useH265Repacketizer = sessionCodec === 'H265' && 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=${useH265Repacketizer}`,\n      { meta: { brokerId, sessionCodec, brokerCodec, sourceType, isRtp, useH265Repacketizer } },\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    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 (useH265Repacketizer) 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                timestampMicros: packet.pts,\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            timestampMicros: packet.pts,\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            timestampMicros: packet.pts,\n          },\n        })\n      }\n    })\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 (useH265Repacketizer) 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            timestampMicros: pkt.pts,\n          },\n        })\n      }\n    }\n\n    const sessionId = crypto.randomUUID()\n    const iceServers = await this.resolveIceServers()\n\n    const session = new AdaptiveSession({\n      sessionId,\n      source,\n      sourceCodec: sessionCodec,\n      iceConfig: {\n        iceServers,\n        portRange: this.icePortRange,\n        additionalHostAddresses: this.iceAdditionalHostAddresses,\n      },\n      logger: this.logger,\n      debug: opts.streamingDebug === true,\n    })\n\n    // For RTSP+H.265 sessions: subscribe to source-RTP and forward\n    // through the H265Repacketizer in the session. Also seed the\n    // repacketizer codec info from the SDP-derived VPS/SPS/PPS so the\n    // AP codec packet can be sent ahead of the first IDR even when\n    // the camera doesn't emit param sets in-band (Reolink RTSP\n    // high-profile streams). Push-mode H.265 (Reolink Baichuan, etc.)\n    // skips this and uses the AnnexB feed path.\n    let unsubRtp: Unsubscribe | null = null\n    let unsubSdpParams: Unsubscribe | null = null\n    if (useH265Repacketizer) {\n      const sdpPs = broker.getSdpParameterSets()\n      if (sdpPs && sdpPs.length > 0) {\n        session.seedH265CodecInfoFromSdp(sdpPs)\n        sessionLogger.info('H.265 session seeded from SDP at create-time', {\n          meta: { sessionId, brokerId, 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 AP codec packet, so\n        // Chrome 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('H.265 session deferred: SDP params not ready, subscribing for late delivery', {\n          meta: { sessionId, brokerId },\n        })\n        unsubSdpParams = broker.onSdpParameterSets((ps) => {\n          try {\n            session.seedH265CodecInfoFromSdp(ps)\n            sessionLogger.info('H.265 session seeded from SDP late delivery', {\n              meta: { sessionId, brokerId, psCount: ps.length },\n            })\n          } catch (err) {\n            sessionLogger.warn('seedH265CodecInfoFromSdp threw', {\n              meta: { sessionId, 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('H.265 session: first source RTP forwarded to repacketizer', {\n            meta: { sessionId, brokerId, bytes: rtpData.length },\n          })\n        }\n        try {\n          session.forwardSourceRtpVideo(rtpData)\n        } catch (err) {\n          sessionLogger.warn('forwardSourceRtpVideo threw', {\n            meta: { sessionId, error: errMsg(err) },\n          })\n        }\n      })\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    try {\n      const offer = await session.createOffer()\n      sessionLogger.info('WebRTC session created', {\n        meta: { sessionId, brokerId, iceServers: iceServers.length, codec: sessionCodec, useH265Repacketizer },\n      })\n      return { sessionId, sdpOffer: offer.sdp }\n    } catch (err) {\n      this.cleanupSession(sessionId)\n      await session.close().catch(() => {})\n      throw err\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: ${sessionId}`)\n    await entry.session.handleAnswer({ sdp: sdpAnswer, type: 'answer' })\n  }\n\n  async closeSession(sessionId: string): Promise<void> {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.cleanupSession(sessionId)\n    await entry.session.close().catch(() => {})\n    this.logger.info('WebRTC session closed', { meta: { sessionId } })\n  }\n\n  async stop(): Promise<void> {\n    if (this.stopped) return\n    this.stopped = true\n    const closes: Promise<void>[] = []\n    for (const [sid, entry] of this.sessions) {\n      this.cleanupSession(sid)\n      closes.push(entry.session.close().catch(() => {}))\n    }\n    await Promise.all(closes)\n    this.brokers.clear()\n    this.logger.info('WebRTC server stopped')\n  }\n\n  getSessionCount(): number {\n    return this.sessions.size\n  }\n\n  // ── Private ─────────────────────────────────────────────────────────\n\n  private cleanupSession(sessionId: string): void {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.sessions.delete(sessionId)\n    entry.unsubBroker()\n    entry.unsubRtp?.()\n    entry.unsubSdpParams?.()\n    entry.closeSource()\n  }\n\n  private resolveBrokerId(streamId: string, hints: ClientHints = {}): string {\n    // Direct match — `webrtc-session.createSession` resolves the\n    // discriminated `WebrtcStreamTarget` to a canonical brokerId\n    // (`${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: 'high' | 'mid' | 'low' | 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: 'high' | 'mid' | 'low' | 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\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 type {\n  CameraStream, ProfileSlot, CamProfile,\n  RtspRestreamEntry,\n  WebrtcClientHints, WebrtcStreamChoice, WebrtcStreamTarget,\n} from '@camstack/types'\nimport type { StreamBrokerManager } from './stream-broker/stream-broker-manager.js'\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\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   * 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 ${deviceId}`)\n        }\n        return `${deviceId}/${slot.sourceCamStreamId}`\n      }\n      case 'cam-stream':\n        return `${deviceId}/${target.camStreamId}`\n    }\n  }\n\n  async listStreams(input: { deviceId: number }): Promise<readonly WebrtcStreamChoice[]> {\n    const { deviceId } = input\n    const slots = this.manager.getProfileSlotsForDevice(deviceId)\n    const assignedSlots = slots.filter((s) => s.sourceCamStreamId !== null)\n    const camStreams = this.manager.getCameraStreamsForDevice(deviceId)\n    const camStreamById = new Map<string, CameraStream>()\n    for (const c of camStreams) camStreamById.set(c.camStreamId, c)\n\n    const choices: WebrtcStreamChoice[] = []\n\n    if (assignedSlots.length > 0) {\n      choices.push({\n        id: 'adaptive',\n        label: 'Adaptive',\n        target: { kind: 'adaptive' },\n        codec: null,\n        resolution: null,\n        status: null,\n        inputFps: null,\n        decodeFps: null,\n        bitrateKbps: null,\n      })\n    }\n\n    // Profile-tier choices: pin a session to high / mid / low. Source\n    // can change as the operator reassigns the slot — the session\n    // follows.\n    for (const slot of assignedSlots) {\n      const profile = slot.profile as CamProfile\n      const camStreamId = slot.sourceCamStreamId\n      if (!camStreamId) continue\n      const broker = await this.manager.getBroker({ brokerId: `${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: `${deviceId}/${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  }): Promise<{ sessionId: string; sdpOffer: string }> {\n    const streamId = this.resolveTargetToStreamId(input.deviceId, input.target)\n    const streamingDebug = typeof this.manager.isStreamingDebug === 'function'\n      ? this.manager.isStreamingDebug(input.deviceId)\n      : false\n    return this.webrtcServer.createSession(streamId, input.hints, { streamingDebug })\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 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","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 { StreamBrokerManager, type ProfileMapPersister } from './stream-broker/stream-broker-manager'\nimport { CameraStreamsProvider, WebrtcSessionProvider } from './device-scoped-providers'\nimport { BrokerWebrtcServer } from './webrtc/broker-webrtc-server'\nimport type { CamProfile } from '@camstack/types'\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\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  /**\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    })\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\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    rtspProvider.setTokenPersister((tokens) => {\n      const obj: Record<string, string> = {}\n      for (const [k, v] of tokens) { obj[k] = v }\n      this.ctx.settings?.writeAddonStore({ rtspTokens: obj }).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist RTSP tokens', { meta: { error: errMsg(err) } })\n      })\n    })\n\n    rtspProvider.setEnabledPersister((states) => {\n      const obj: Record<string, boolean> = {}\n      for (const [k, v] of states) { obj[k] = v }\n      this.ctx.settings?.writeAddonStore({ rtspEnabled: 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) persisted as numeric\n    // device ids. JSON serialises numeric Map keys as strings, so on\n    // read we reparse and drop anything that isn't a valid number.\n    try {\n      const addonStore = (await this.ctx.settings?.readAddonStore()) ?? {}\n      const overrideData = asJsonObject(addonStore['deviceOverrides'])\n      if (overrideData) {\n        const overrides = new Map<number, {\n          readonly preBufferSecOverride?: number\n          readonly preBuffer?: Record<string, { enabled: boolean; seconds: number }>\n          readonly streamingDebug?: boolean\n        }>()\n        for (const [rawKey, raw] of Object.entries(overrideData)) {\n          const numericKey = Number(rawKey)\n          if (!Number.isFinite(numericKey) || !Number.isInteger(numericKey)) continue\n          const o = asJsonObject(raw)\n          if (!o) continue\n          const entry: Record<string, unknown> = {}\n          if (typeof o['preBufferSecOverride'] === 'number') {\n            entry['preBufferSecOverride'] = o['preBufferSecOverride']\n          }\n          if (typeof o['streamingDebug'] === 'boolean') {\n            entry['streamingDebug'] = o['streamingDebug']\n          }\n          const pbRaw = asJsonObject(o['preBuffer'])\n          if (pbRaw) {\n            const preBuffer: Record<string, { enabled: boolean; seconds: number }> = {}\n            for (const [profile, v] of Object.entries(pbRaw)) {\n              const sv = asJsonObject(v)\n              if (sv && typeof sv['seconds'] === 'number') {\n                preBuffer[profile] = { enabled: sv['enabled'] !== false, seconds: sv['seconds'] }\n              }\n            }\n            entry['preBuffer'] = preBuffer\n          }\n          overrides.set(numericKey, entry)\n        }\n        this.brokerManager.loadPersistedDeviceOverrides(overrides)\n        this.ctx.logger.info('Loaded persisted device overrides', { meta: { overrideCount: overrides.size } })\n      }\n    } catch { /* first boot — no persisted state yet */ }\n\n    this.brokerManager.setDeviceOverridePersister((overrides) => {\n      const obj: Record<string, Record<string, unknown>> = {}\n      for (const [deviceId, ov] of overrides) {\n        const entry: Record<string, unknown> = {}\n        if (ov.preBufferSecOverride !== undefined) entry['preBufferSecOverride'] = ov.preBufferSecOverride\n        if (ov.streamingDebug !== undefined) entry['streamingDebug'] = ov.streamingDebug\n        if (ov.preBuffer) entry['preBuffer'] = ov.preBuffer\n        obj[`${deviceId}`] = entry\n      }\n      this.ctx.settings?.writeAddonStore({ deviceOverrides: 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    try {\n      const addonStore = (await this.ctx.settings?.readAddonStore()) ?? {}\n      const raw = asJsonObject(addonStore['profileMap'])\n      if (raw) {\n        const loaded = new Map<number, PersistedAssignment>()\n        for (const [rawKey, rawVal] of Object.entries(raw)) {\n          const deviceId = Number(rawKey)\n          if (!Number.isFinite(deviceId) || !Number.isInteger(deviceId)) continue\n          const val = asJsonObject(rawVal)\n          if (!val) continue\n          const mapObj = asJsonObject(val['map']) ?? {}\n          const map: Partial<Record<CamProfile, string>> = {}\n          for (const profile of PROFILE_KEYS) {\n            const v = mapObj[profile]\n            if (typeof v === 'string' && v.length > 0) map[profile] = v\n          }\n          const auto = val['auto'] !== false\n          loaded.set(deviceId, { map, auto })\n        }\n        this.brokerManager.loadPersistedProfileMap(loaded)\n        this.ctx.logger.info('Loaded persisted profile map', { meta: { deviceCount: loaded.size } })\n      }\n    } catch { /* first boot — no persisted state yet */ }\n\n    const profileMapPersister: ProfileMapPersister = (assignments) => {\n      const obj: Record<string, PersistedAssignment> = {}\n      for (const [deviceId, assignment] of assignments) {\n        obj[`${deviceId}`] = { map: { ...assignment.map }, auto: assignment.auto }\n      }\n      this.ctx.settings?.writeAddonStore({ profileMap: obj }).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist profile map', { meta: { error: errMsg(err) } })\n      })\n    }\n    this.brokerManager.setProfileMapPersister(profileMapPersister)\n\n    // Start RTSP restream server\n    await this.brokerManager.startRtspServer(this.config.rtspPort)\n\n    // Per-agent hwaccel override changed — force-rotate any active shared\n    // decoder session whose owning node matches.\n    this.subscribe(\n      { category: EventCategory.PipelineAgentHwaccelChanged },\n      (event) => {\n        const { agentNodeId, reason } = event.data\n        if (typeof agentNodeId !== 'string' || agentNodeId.length === 0) return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = typeof reason === 'string' ? reason : 'hwaccel-changed'\n        void manager.rotateDecodersOnNode(agentNodeId, reasonLabel).then((rotated) => {\n          if (rotated > 0) {\n            this.ctx.logger.info('decoders rotated for hwaccel change', {\n              tags: { nodeId: agentNodeId },\n              meta: { rotated, reason: reasonLabel },\n            })\n          }\n        }).catch((err: unknown) => {\n          this.ctx.logger.warn('hwaccel-driven decoder rotation failed', {\n            tags: { nodeId: agentNodeId },\n            meta: { error: errMsg(err) },\n          })\n        })\n      },\n    )\n\n    // Decoder down → rotate active shared sessions to a healthy provider.\n    this.watchCapability(['decoder'], {\n      onDown: (nodeId, capName) => {\n        if (capName !== 'decoder') return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = `decoder-down:${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    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-orchestrator` cap (singleton). The orchestrator aggregates\n    // ICE servers from every enabled `turn-provider` (cloudflare-turn,\n    // future coturn / Twilio, …) — there's NO per-device round-trip;\n    // the orchestrator caches at the provider layer (cloudflare-turn\n    // refreshes ICE credentials in the background ~12 h before the\n    // 24 h TTL), so the per-session call costs a single cap dispatch +\n    // an in-process memory read. Without TURN configured the call\n    // returns `[]` and WebRTC falls back to host/srflx candidates.\n    const turnApi = this.ctx.api as unknown as {\n      turnOrchestrator?: {\n        getAllServers: { 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.turnOrchestrator) return []\n        try {\n          const servers = await turnApi.turnOrchestrator.getAllServers.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          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('turnOrchestrator.getAllServers failed — session will start without TURN', {\n            meta: { error: err instanceof Error ? err.message : String(err) },\n          })\n          return []\n        }\n      },\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\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          stableId: 'stream-broker-panel',\n          label: 'Stream Brokers',\n          description: 'Per-camera stream broker panel with adaptive controls.',\n          icon: 'radio',\n          remoteName: 'addon_stream_broker_widgets',\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    }\n\n    this.ctx.logger.info('Stream broker manager initialized')\n    const registrations: ProviderRegistration[] = [\n      { capability: streamBrokerCapability, provider: this.brokerManager },\n      {\n        capability: cameraStreamsCapability,\n        provider: cameraStreamsProvider,\n        kind: 'wrapper',\n        defaultActive: true,\n      },\n      {\n        capability: webrtcSessionCapability,\n        provider: webrtcSessionProvider,\n        kind: 'wrapper',\n        defaultActive: true,\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    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        },\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","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'\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    '-hide_banner', '-loglevel', '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}${toBase64Url(sig)}`\n}\n\nfunction verifyToken(secret: Buffer, token: string, expectedSessionId: string): AuthVerifyResult {\n  const parts = token.split(SEPARATOR)\n  if (parts.length !== 3) {\n    return { ok: false, reason: 'malformed' }\n  }\n  const [sessionId, expiryStr, signatureB64] = parts as [string, string, string]\n\n  const expiryEpochMs = Number(expiryStr)\n  if (!Number.isFinite(expiryEpochMs) || expiryEpochMs <= 0) {\n    return { ok: false, reason: 'malformed' }\n  }\n\n  if (sessionId !== expectedSessionId) {\n    return { ok: false, reason: 'wrong-session' }\n  }\n\n  const expectedSig = crypto.createHmac('sha256', secret).update(`${sessionId}${SEPARATOR}${expiryStr}`).digest()\n  const providedSig = fromBase64Url(signatureB64)\n\n  if (providedSig === null || providedSig.length !== expectedSig.length) {\n    return { ok: false, reason: 'bad-signature' }\n  }\n  if (!crypto.timingSafeEqual(providedSig, expectedSig)) {\n    return { ok: false, reason: 'bad-signature' }\n  }\n\n  if (Date.now() > expiryEpochMs) {\n    return { ok: false, reason: 'expired' }\n  }\n\n  return { ok: true, sessionId, expiresAt: expiryEpochMs }\n}\n\nfunction toBase64Url(buf: Buffer): string {\n  return buf.toString('base64').replace(/\\+/g, '-').replace(/\\//g, '_').replace(/=+$/, '')\n}\n\nfunction fromBase64Url(s: string): Buffer | null {\n  const normalized = s.replace(/-/g, '+').replace(/_/g, '/')\n  const padded = normalized + '='.repeat((4 - (normalized.length % 4)) % 4)\n  try {\n    return Buffer.from(padded, 'base64')\n  } catch {\n    return null\n  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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// Match AudioRtpDecoder's emit cadence so YAMNet/Apple SoundAnalysis\n// see the same window size regardless of source codec. A YAMNet hop is\n// ~0.48s and Apple SA classifies on rolling ~0.96s windows; 0.5s of\n// target-rate samples gives both classifiers a stable input. Codec\n// frames are typically 64–128ms so ~4–8 frames coalesce into one chunk.\nconst TARGET_CHUNK_MS = 500\nconst F32_BYTES = 4\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  // Accumulator for the ~0.5s emit window. The codec cap returns a PCM\n  // frame per decoded codec packet (~21–128ms depending on codec), which\n  // is too short for the audio classifiers downstream. We coalesce\n  // until we've buffered TARGET_CHUNK_MS worth of samples, then emit\n  // one chunk with the earliest packet's timestamp.\n  private pendingPcm: Buffer[] = []\n  private pendingBytes = 0\n  private pendingSampleRate = 0\n  private pendingChannels = 0\n  private pendingFirstTs = 0\n  private readonly chunkBytesTarget: number\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    this.chunkBytesTarget =\n      Math.max(1, Math.round(cfg.targetSampleRate * (TARGET_CHUNK_MS / 1000))) *\n      Math.max(1, cfg.targetChannels) *\n      F32_BYTES\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    // Emit whatever's in the accumulator so downstream audio\n    // analyzers don't miss the last partial window. Skipped if the\n    // accumulator is empty.\n    this.flushPending()\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   * Stage a freshly-pulled PCM frame into the accumulator. When we've\n   * buffered enough samples for the configured ~0.5s window we\n   * concatenate, copy out, and emit a single DecodedAudioChunk so\n   * downstream audio classifiers see a stable input length regardless\n   * of source codec frame size (G.711 packets ≈ 20ms, AAC ≈ 21–43ms,\n   * Opus ≈ 20ms).\n   *\n   * Format changes (sampleRate/channels) flush the previous window\n   * first so a chunk never spans two formats.\n   */\n  private appendPcm(pcm: { data: Uint8Array; sampleRate: number; channels: number; pts: number }): void {\n    const incoming = pcm.data.byteLength\n    if (incoming === 0) return\n\n    // Format change → flush the previous window before swapping params.\n    if (\n      this.pendingBytes > 0 &&\n      (pcm.sampleRate !== this.pendingSampleRate ||\n        pcm.channels !== this.pendingChannels)\n    ) {\n      this.flushPending()\n    }\n\n    if (this.pendingBytes === 0) {\n      this.pendingSampleRate = pcm.sampleRate\n      this.pendingChannels = pcm.channels\n      // pcm.pts is a codec-relative PTS (often starting at 0), not a\n      // wall-clock timestamp. Downstream consumers (audio-metrics\n      // snapshot, UI \"X seconds ago\" chip) interpret the chunk\n      // timestamp as ms-since-epoch — emitting `pts` produced\n      // bogus values like \"1777832862s ago\" because Date.now() - 0\n      // ≈ Date.now(). AudioRtpDecoder always uses Date.now() too,\n      // so the codec path now matches.\n      this.pendingFirstTs = Date.now()\n    }\n    // Buffer.from(uint8array) deep-copies — safe to keep across the\n    // RPC tick that owns the source buffer.\n    this.pendingPcm.push(Buffer.from(pcm.data))\n    this.pendingBytes += incoming\n\n    if (this.pendingBytes >= this.chunkBytesTarget) {\n      this.flushPending()\n    }\n  }\n\n  private flushPending(): void {\n    if (this.pendingBytes === 0) return\n    const merged = this.pendingPcm.length === 1\n      ? this.pendingPcm[0]!\n      : Buffer.concat(this.pendingPcm, this.pendingBytes)\n    const sampleRate = this.pendingSampleRate\n    const channels = this.pendingChannels\n    const timestamp = this.pendingFirstTs\n    this.pendingPcm = []\n    this.pendingBytes = 0\n    this.pendingSampleRate = 0\n    this.pendingChannels = 0\n    this.pendingFirstTs = 0\n    this.emit({\n      data: merged,\n      sampleRate,\n      channels,\n      timestamp,\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    // Drain whatever survived from the dead session — the new session\n    // may negotiate a different sampleRate/channels, and emitting the\n    // tail keeps timeline continuity for the classifier.\n    this.flushPending()\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","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 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","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 { 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  /** 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  }\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.sdp = buildSdp({ codec, parameterSets: all.slice(0, needed), streamName: this.streamName })\n        this.mutedSdp = buildSdp({ codec, parameterSets: all.slice(0, needed), streamName: `${this.streamName} (muted)` })\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` — same wire format the scrypted Reolink\n * plugin consumes through its `x-scrypted/x-rfc4571` MediaObject.\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 * Ported from `koush/scrypted/plugins/prebuffer-mixin/src/flv.ts` (ISC).\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 * Ported from `koush/scrypted/plugins/prebuffer-mixin/src/rtmp-client.ts`\n * (ISC). Adaptations:\n *   - `console` replaced with project `IScopedLogger`.\n *   - Inlined `readLength` helper (no dependency on @scrypted/common).\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 (replaces @scrypted/common/src/read-stream) ──\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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6Ci4ozoX7d5fn0zYMXFVF3N2rnh/WmSzDD7Q/Yx0ttU/siyEirAQqrQnEav6JK+uaQwUdK77oi4laVukHGWP2V2X1yuHVV8AlW0z1wpmPtUnu1MmeHIavzUFAZgWXD5++r6Ot/hOgJp0ZxPZZUCgbmvknx5guU/EmHwrKCjOjqIsvFYuC2yd0FpCAz1B6OI6V7WfG1P56Ln5aI06Mf46GEEjh8gK4KRznRdB/xXyJUnbQh3VRu022hXqpI439/25/8EyU+o6TjhSMpV2XnR83rGflzCfxYBa0eqm/oCDk5lpViQusu+1v7g6+/B8A3JkM78sO064c9DJHwBKcssLoncxh1rOZHqYmH7dyJcKvKl56c4yjMAPGoTu/dQ7DqUl+/1ZTfXnScfhvxLshgyc0P9fbDvKJngtx+FL7pEI3N0am5b7qNTDhVZT3KgbCynYRYKvL/yEzW67K2BJ5ZFl9CGtKA/Wuue66P7sZTogpNLy5bCzs0EBpxAz15JdP6o8IKrx8g4YPCirXwWayjOcdVY5Yzoh3EIABoL7JyIPJ+cOqOQRVbaNRSK3uqK0cDclgP7JNT6itDaY+eNuGZnsLka93Ojqnx0/OvvcM/VMOV3XNoBzjMCwBK62jL3BJ6Sjkzv9zgFSVmuyQ1AhGowTYPI7lEOqJeg9acM47dS50fO57M1ipbHahZzkNA7cArH+mpK6pofOUpxsEuAjwx0/oJZX7WPb+2nYRSd8bSa6KvwWZKrxcxDGUDRuPXs1qUL0bx5/nt4Rpdh+r541w9HBywK/EqnlL5DYgjOp4GzorITU/9i2NsPaK/Z6yc4RvPkpFu+lTXX15KedvWxUH8IMgm4DaPS9pdw0J6iZtJSap+C6kyRPgZtqZ9JVAU0V5i8gQY+L0jFX3PnWHcMOAzHQwFGZWBOtMq8pkyJ93N3kgst1DWGj6egLEMVuSRONH/3vN8qXpifVRo9g/NcuIYc8l/zYlWqIIB/WlttCvqZ/NE9cN6m6xU2WDfO+5QIHQBMGSHWZCN16hw86A0bJeS6SRsmWbN0LJO0V/YXTr2EiAtBYVawrglXrSVtdH8nEGXJVznoRPgYu5p4Llm0e3e6GvZnJYWJ97lrbWXcLvpXP54XUDyIEPuXlZTRmMUWoHFTl8OIgUPhTIKMoG/Ib89C5uyCXO415fSc6Ppq8MEiTT76ZAqXjfG7Dl5ACmBTbvX450/CB1Vi9xB8IIFFT7xD+OAmwewOGLaXBBO4pkl5r7Upopbpeoi/FAHZ0bMCdmOQgDTKnioiPzQYcq0KWVAoG5r5J8jULlPxJh8Kyxozo6iLLxWLfvItk+Dk1uOEUQG0sxnSpTCtftlKRmLEP1+zyJErrDz9yres+EgsDhno7sW7bOYXOuph4PB+Ra87C7UpcF7QoQ4Rq7+QtOobzeYS5V2OR5dJxNkiYFzyiARK0HojeNvZPsDPukDb/rr8DQ14H4kMoXqBt5PD6qjJfl4JKPpPL4SRx2PfF7oETuqmVkkm6si4m2Yxa551yCAs2KFO04guWSpDFtgl80mCDMLqMoSphaJF3HEX71YqU7ZLyY8Zdk8ruJMIrvZvlE8/RxyFAjlGSVYrv6wO2FPVx0houMsMwB5IpBY4ki9hC5edchcPQZNwM878Wt/o+bYxR+wIqJgK1nMqybFzcZEapW0k0SBbKXlHosilK/N6AAACngAAADAbwcg0sfHQt65f6mnxtP2r+08fGJCSWncC9bRMWkwVBgpQR9hbHN4iZZRH0Z8rW+xzoBFYLNKy3xIereHgLNTMOv62N2JOpDqN++sH7wAcc3e9Fo2nP31QA4zC527jTsgdwKgJaisVqq83vKsLGD+mX0effufZ3T8Mg5/QoFOcWvVE4u51cXJYr+7QsOOiBhz1wGGb7n5Riy+uIE70aDf4chPltnlwTlOPeaZba/RfYvXb3LTvBBXvXF2/vt/uNbdw73//djuXVcc2u778jWxY8yXBlDP1Lh7gFkKU9LsFJc+gnHXU7W3jI8fFwuxeJKAfRlWNwTZmYANsBtJlzM4qbyhxXYGOOzWlDyICJTKcnkQssORsQVM5bsaHBXPm4piT9sxg19wlCBp6K5MigVX+HFnevX/Y55tonjRrEoHmGCV9kKRV6ZUIESRziBozfic6j2ftv4oLu5bI/fQtkvAnj3rtJ1XWbfdQsBxRBf6dZjWewdrMtZy7s2Wqf9d6W9GrHRKH44rS+gtFLk8h0FJLn6zQLoVdsrtdTnOdG+3OB2nJ8Ll1ZrqsyrSK0KBWtBthMS90/8GbrLJyds6h7ctDw791GN2aqDJ4h42RhTazZhiZ2XJjw6PtucSfsCR0iPYPlOm9W7uSVsQwgcfzevIWBhU/4K4IxYJohDWirZbT7QciZuXwTC574DOGso6Ddu3iRhYFYzPOcWvgxxKvDr8UBw//4Jnc4JfKbvTGbc3k50ML5IZCuQJxyj+K22xRf2fBinLjfiXt1ncXjhKC7lzlgYXWi7dNZL917wYytylf0nvn0ABCs9XijfD8jvNVUhzn7FyKsLJbgHr24rz1hCw/YBipkxDP39vYAiBmNX2uG5IuVVnJZ5fGHEdy09nClEnyPYgLxuQs9UBSwoUcd1awDoeG0l/+NyE4HnszKMX3+4VyQwKWqek0dspBqT2meYK0sbomqG3nRKxUANy0l4iqNX/xwPjuCyd/7Xdxz93mT5yVFNbci5YUea9g6jhB/KT8HdwnvDe/DMilt1HPZpgB0wgXCNAaIqLjVB+8kxgLrHmYXtvo2wbzl5+7FVN78Rm8uMHl/AobG9C2ph/tMv6LoCfkq6pRm4x5IMB0kModAdXa3qjA92hYeX2AAkTYfvAGU0jYpUtcE6IUmjyt8t79IA3do96y3l/Yrpo5p67dN3Xmp+3lkuj8pcIfCqZBUyQlrrXNgaj3zli2u4StYzLJgLhQ3OKrP/sQZ14caam0hWYDNJkEgN3Ydj6InQ1Ymx0Aedt/SUXSbw1w5oiZahKrkub8bHe3MAMed981rnNunKdC18+ckf3z1FaLECw0A5R3vJEfVyRPkyBxckOY59j+suEa5MpVNZalf+UxyzmlqUbgmkaXgHe9PbtWLkdHWXn5HRQeWABU0ky7jrKktuGVmiHhjtyjzPBq4ZEZxfciWLnJRfLzzue4v+azInxca0i8DzyTb2eWXtl9yVEz2yxJ1mX7tdQej9o0nYsHpN4EBE5SHK5o32tCjxpYdr2cufCrq9sl0bHxRDKZceNwgKGCTSPXYIwC04S5qBjOrxDVexlbFVvOpSrh6KxSamB9ZGgkKGqP2RKVci7zuI9wEMNLjkO7elLpXa4KU8NIwAHNp7SJoIpMYxBaD/iK4ZDq2AP3G0kIWJRKvZl2whIMguF9R7pDDFNn9WoFJ021BWo8DMFyrxvNQCoZlopDA7/sgJ16VBsGKOXDltXigk5APGWqSjaeezweoevWhT+M8th4+2yF862EsNo36UF0xB16BSZOaMSB8vTsH0u3+Lc6Czm0A7UiUvNlkwJ5z0lLwIxmI/+LSWEmzHIsjeXWQHde30gUHvTvOMIhWqHvjiSvLjw17T6tijkO47h8RI9bWzmG9DNEn9a8bmlJrHh/tGMgAwGu4DUUotoHfaHegkYIrWzkfVH13yHTTGrcY82KdmS6lT5E8uXIcyGoQ6LJYmDZu3WX2GS66RN8oehSzHcH58HgYoTpYKIDweSZcEiWYz/tZAWMYZgWXnWQZPfaABFbbCns0twaNs0Sn41f8gkjAc3lm4pqFMTAj9xaMBcmLefXcrkWTHw691m21Xnc9pQdMbWmlorU8Boq5AbDnckloxu0Z74DJi9j6A0n9LVh6dlzvodocp71Er/hGdNJRi+Vcl+kHo1ic3iIOHsS2Yqhs0RNFjo/fjbxeaOBgWRLxNRkx/AIkiJVHNV0XAv98N3FxA1oWE28krbjksU0KMrB6rkKwUfNcfbACNDAtzuKkSRpw07CcY+Zpnp00QTqET6boRpUMsnfaZdk21i1g0aIOiDRr2dpZ5PoUh+x3t9oWEgAsb4tJ+WL2NgGdhlL9ArpbJCkoYxz1XC143XWlkVey9DkztysoGKuHxs1kS/741nQt+prrl/yhEsVDmZ/Z5MMAj9O3hJyAkT44zRhjJLh3tWJoeFlbIzQq1uS5d0BE3kG6Qr6QNycPBhMCmaPgehdmJY+pOPHDi7GKK5rgDhTTxTtxWHyLLtfZ0uvhCyP2f/vgdf//AW22MA3F7+0FcPd/9PQKUGX/d4OreG7Oo6JiAmP2IarCYG1AQh9D8GlhytEMDqFqrMb+m0c0dcyEfdriIIgDPTFNJ50aGVMu1lYMBAeTw5jwhtYMuiogKIPkiwBbuxxmY24CzXbT4fPEXiuSTi3xI+QpK6pQKD303zumZnAWYRe3RoU+5hh2mqFB4XpO7X67q4pTDvY2iIBJgNeVaNgkLRA5AgiUAq9OnZZ6DW/qsT3aLY1XqDqoX0OQqltPj2HGVs3TjfcD74AvF6ESnvX9RzOG9LDlya+1j7GxfQB8dY96fBq7EdJtwBRND9wTMC7kswFAClGZNPZ5FhthY9JGLV3GPjO2Pa1kZwp9IvZZMROq0ELt2ZAfv+Ue5IoQO7pEUCsjOnNStt3pWwP6Vpi5XsQVkGI+NARaVSl7Nudb0UnrG5gbENDRtcZIXAxGdIgwOt6e8vysvL+b5Yi8IRqaaQ89puymmtR6vrVec2CcaYWm5nZdP6HLbqgFDHujcz0NOqV71H1GCiaNrWy/t35pRVfq640VGg52Hqg8tTaaqsXG2uuThvW3MRANtcHJMvdaV8I9d+Vb28UNf47VvosuDkJxZYexaYsVvpxtvlHsbGPbfsUm+ucN6qbHw/4nWSJgPd9K2t+LpU6dW01zs7bZUA0ZWBh8KbVP8neck5aIq4IsoFGxGdJRugUY2kc1o6QKsSFpzqsUO8DxRZ9uWp/d88Uhl33NB99BZD64HBGT5R6s3X/NwvIehHtrdWJ1CQexCTVFv3Eb0CuuWaglxVhyrp+p3xVxoVlxqBIRw1ct155Gf4rLGzRWhi/lD/h6eJ6I6Gh3DZ/waZwb/4jhIjHRYemhw12YoNzyTqV340QHjgBALKzT6EYu0BCp3k05aopnwZDqqCC6uc4H2//2hUxaKOv36N7VdYluXSj1nq4fr5Xn+mvFRheiMKxalcBxAPkgiRIO3V2M8WZp05C0d7dovQabSceQ7zzX5Kis3fOr4IQvGA1dsmQYeHClVm749yGxVPhITPaIxhFGdKsnqp/N5oBrOs08kkmO4KsIQ4YVjQp3SDyekaMbdh42ZKxmzF7CDszcChDbzxKi/47kzTL+XxnDI6npJCUA/lvh6P/3ooqWRAZ+ohyjnqz/83aeJjC0d2toQFL4EIajAiRul9hQypMhS0XnmHZ15XbBiLRGpbDdZQ6bWbe/6+fbakKDMUP5Ewz8DK0PijXVkvCQYaBrVQqwKhquutLM69iT5h9cVApJCw+Nllb0UHOiobf4oGg8lIC3H/FN62IA/vHZcetduokSAeNxWOKvrPx6q4aEXZEqEJU/5RxL2EPUidxh1T7aRM3MGJM8f3SkvCq+bq1EmJthptURTAgWP/WpjVjvdcYz2L9qxOe83/8Gw8//0js8i4pOvGOcmCVLlb4Im0GZJQfCQD3bdwBFnTf0blZtn+4pOOyWKF5kvPUqsNLRIYwRaZxrtVx11WJI938QDTMzbi1RAh3yZRRfc5+zf1Gr2LTnThAaUv17gs5KGGbGfd0Uuc/Ky7Cdjbnj6KmLHSCCrFDBzQAMHtbNdI/mGg9ut9ZcCsKZngnEFCvoQaLf66strIWjhDB7bqnxq8cqq4mATh7t98ES8NnEOWXT9z0zhDfRMZ8e7CQOk3nc3+p5zyNXA0TXCtjAP6sE0ShNfMQ/tTguB5Zj1E+zUb7sOjZANfmHFtjJlzY34YLRBGUg5o5sR+nCG+kaOtdxHUSq8C3YtzqQT+e5sxuLeme3xCkLSaC9OUO8cmIzNmtFnYk353mnh8Wgg4UdK5SI1oF2vRRsjcGL571LepYc8lwkS/WmcdAhR379sIjgQT14ysZ8k9eLssO9UjB/7q2tG1AL0KYlwjlCuqjR6pAdrjmOpq2kVVY045O/7L+Jhpn7PqvQDgFhsWOrMEOm1EoTFMgRuPtoWHTeizXe9lcCYVsCt/5rG6LxIPBz4ZVFBF0Cj8Or6q2yftONDIY/UXBy0KnWmbHRu4he3qt7vWLVh7/3NxyGNNxn9X1b9D50txYJA15Itn3rsHOBNYb94xeX3niF0VIWMbgv+9wCqYqRlwoGtH0CFyNFo4+2qOJpXAoXM937ry/eSK+Z8c40wn9AIPg8C1T5qscDGAvYolWSVJqa0imWz+UnlYPGjnQCMII4mY9IWqUOKRC2N8NKHvw/EwFYAAAMCqTu4+23fGyAAABLQAAADAAADAAADACtgAAADAAADAAADAH+AAAADAAADAAADAWMAAAMAAAMAAAMDsgAAAwAAAwAABuQAAAMAAAMAAA4YAAADAAADAAAY0AAAAwAAAwAAKSAAAAMAAAMAADzghwAAAwAAAwAAAwAAAwAAAwAAAwFp',\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  DecodedFrame,\n  DecodedAudioChunk,\n  DecodeOptions,\n  BrokerStatus,\n  BrokerStats,\n  Unsubscribe,\n  IRestreamer,\n  IScopedLogger,\n  FrameFormat,\n} from '@camstack/types'\n\nimport { DecoderSessionProxy, type DecoderCapApi } from './decoder-session-proxy.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 { FrameDropper } from './frame-dropper'\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'\nimport type { Sharp } from 'sharp'\n\ntype SharpFn = (input: Buffer | Uint8Array, opts?: Record<string, unknown>) => Sharp\n\nlet cachedSharp: SharpFn | null = null\nasync function getSharp(): Promise<SharpFn> {\n  if (cachedSharp) return cachedSharp\n  const mod = await import('sharp')\n  cachedSharp = mod.default as unknown as SharpFn\n  return cachedSharp\n}\n\n/**\n * In-place R↔B swap on a packed RGB24 buffer. Used by the broker's\n * per-frame conversion cache to satisfy `bgr` subscribers when the\n * decoder's canonical output is `rgb`. Returns a new Buffer so the\n * source frame stays untouched (other subscribers may still need it).\n */\nfunction swapRedBlue(rgb: Buffer): Buffer {\n  const out = Buffer.allocUnsafe(rgb.length)\n  for (let i = 0; i + 2 < rgb.length; i += 3) {\n    out[i] = rgb[i + 2]!\n    out[i + 1] = rgb[i + 1]!\n    out[i + 2] = rgb[i]!\n  }\n  return out\n}\n\nconst DEFAULT_MAX_FPS = 5\nconst DEFAULT_SCALE = 1\n\nconst INITIAL_RECONNECT_DELAY_MS = 1_000\nconst MAX_RECONNECT_DELAY_MS = 30_000\nconst DECODER_TEARDOWN_GRACE_MS = 2_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\ninterface DecodedSubscriber {\n  readonly callback: (frame: DecodedFrame) => void\n  readonly frameDropper: FrameDropper\n  /** Caller-supplied identity — used by `listClients` for diagnostics. */\n  readonly tag: string\n  readonly maxFps: number\n  /** Requested output format. Gray consumers don't need JPEG encoding. */\n  readonly format: FrameFormat\n  readonly subscribedAt: number\n  framesDelivered: number\n  framesDropped: number\n}\n\ninterface AudioSubscriber {\n  readonly callback: (chunk: DecodedAudioChunk) => void\n  readonly tag: string\n  readonly subscribedAt: number\n  chunksDelivered: number\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 — mirrors Scrypted's\n   * `ensureRfcServer`-on-each-`getVideoStream` pattern. `null` for\n   * tests and stand-alone callers; falls back to the source passed\n   * 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  private decoderProxy: DecoderSessionProxy | null = null\n  /** Current output format of the shared decoder session. */\n  private decoderOutputFormat: FrameFormat | null = null\n  /**\n   * Moleculer nodeID of the decoder provider currently hosting\n   * `decoderProxy`. Set from the `{ nodeId }` returned by\n   * `decoderApi.createSession`; used by the broker-manager's\n   * hwaccel-change subscription to decide which brokers need a forced\n   * decoder rotation. Null when no shared decoder session exists.\n   */\n  private decoderNodeId: string | 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  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  private manualStop = false\n  private stopping = false\n  private decoderTeardownTimer: ReturnType<typeof setTimeout> | undefined\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 Set<(packet: EncodedPacket) => void>()\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 Set<(rtpData: Buffer) => void>()\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  private readonly decodedSubscribers = new Map<symbol, DecodedSubscriber>()\n  private readonly audioSubscribers = new Map<symbol, AudioSubscriber>()\n  /** Stored options from first onDecodedFrame call when codec was not yet known */\n  private pendingDecodeOptions: DecodeOptions | undefined\n  /** Pull mode decoder needs to wait for the first keyframe before opening */\n  private pendingPullModeOpen: (() => void) | null = null\n  private firstKeyframeReceived = false\n  /** Retry handle for \"no decoder provider yet\" — addons register in\n   *  arbitrary order, the broker must wait rather than fail. Cancelled\n   *  on decoder attach or broker destroy. */\n  private decoderResolveRetryTimer: ReturnType<typeof setTimeout> | undefined\n  /** Attempt counter for the retry loop above — feeds bounded backoff. */\n  private decoderResolveAttempts = 0\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  /** Debug counters for decoder troubleshooting */\n  private decoderPushCount = 0\n  private decoderFrameCount = 0\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  /** Cached decoder stats — updated by the polling loop, read synchronously by getStats(). */\n  private cachedDecoderStats: import('@camstack/types').DecoderStats | undefined\n\n  /** Tracking flags set synchronously by the RTP depacketizer callback. */\n  private _lastNalKeyframe = false\n  private _lastNalParamSet = false\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\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 — same Scrypted prebuffer-\n    //     mixin contract. Holding the loopback open 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\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.decoderTeardownTimer) {\n      clearTimeout(this.decoderTeardownTimer)\n      this.decoderTeardownTimer = 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.decoderProxy) {\n      await this.destroyDecoder()\n    }\n\n    await this.pipeServer.stop()\n    this.rtspRestreamer.destroy()\n    this.preBuffer.clear()\n\n    this.pendingParamNals = []\n    this.pendingPullModeOpen = null\n    this.firstKeyframeReceived = false\n    this.encodedCallbacks.clear()\n    this.decodedSubscribers.clear()\n    this.audioSubscribers.clear()\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    if (this.decodedSubscribers.size > 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.audioSubscribers.size > 0) return true\n    if (this.rtspRestreamer.getSessionCount() > 0) return true\n    if (this.pipeServer.getClientCount() > 0) return true\n    if (this._preBufferEnabled && this.preBuffer.getDuration() > 0) 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  /**\n   * Pick the decoder's native output format given the current subscriber\n   * mix. Phase 3 collapsed three concerns into this one decision:\n   *\n   *   1. Avoid wasted CPU when subscribers are homogeneous — `gray`-\n   *      only stays `gray` (motion-only camera), `jpeg`-only stays\n   *      `jpeg` (no detection consumer joined yet, broker doesn't have\n   *      to encode either).\n   *   2. Pick the cheapest *canonical* mode for mixed subscribers —\n   *      `rgb` is the universal source for the broker-side conversion\n   *      cache: gray is one luminance pass, jpeg is one sharp encode,\n   *      bgr is a channel swap. Producing JPEG decoder-side and then\n   *      *decoding back* to feed a raw consumer would cost more than\n   *      both consumers combined.\n   *   3. Stay stable when there are no subscribers — defaults to\n   *      `jpeg` so the legacy \"no subscriber yet\" code path keeps\n   *      behaving as it did pre-refactor.\n   */\n  private resolveDecoderFormat(): FrameFormat {\n    if (this.decodedSubscribers.size === 0) return 'jpeg'\n    let allGray = true\n    let allJpeg = true\n    for (const sub of this.decodedSubscribers.values()) {\n      if (sub.format !== 'gray') allGray = false\n      if (sub.format !== 'jpeg') allJpeg = false\n    }\n    if (allGray) return 'gray'\n    if (allJpeg) return 'jpeg'\n    return 'rgb'\n  }\n\n  /**\n   * Check if the decoder's output format needs upgrading because a new\n   * subscriber requires a richer format (e.g. gray → jpeg when detection\n   * joins after motion). Calls updateConfig on the decoder proxy to\n   * reinitialize the scaler without recreating the session.\n   */\n  private maybeUpgradeDecoderFormat(): void {\n    if (!this.decoderProxy || !this.decoderOutputFormat) return\n    const needed = this.resolveDecoderFormat()\n    if (needed === this.decoderOutputFormat) return\n    this.logger?.info('upgrading decoder output format', {\n      meta: { from: this.decoderOutputFormat, to: needed },\n    })\n    this.decoderOutputFormat = needed\n    this.decoderProxy.updateConfig({ outputFormat: needed }).catch((err: unknown) => {\n      this.logger?.warn('decoder format upgrade failed', { meta: { error: errMsg(err) } })\n    })\n  }\n\n  /**\n   * Fan a single decoded frame out to every active subscriber. Each\n   * subscriber may have requested a different `format` than the one the\n   * decoder is currently producing — the per-frame `convertedCache` map\n   * memoises any format the decoder didn't already supply, so multiple\n   * subscribers asking for the same target share the conversion.\n   *\n   * Conversion paths covered today:\n   *   - rgb → jpeg (sharp encode)\n   *   - rgb → bgr  (R/B channel swap, in-process)\n   *   - rgb → gray (BT.601 luminance)\n   *   - rgb → yuv420 (sharp toFormat)\n   *   - gray → jpeg (sharp encode 1ch)\n   *   - jpeg → rgb / bgr / gray / yuv420 — not supported in Phase 3.\n   *     Decoder picks `jpeg` as canonical only when ALL subscribers\n   *     want jpeg, so this branch should never trigger. Such a request\n   *     is logged and the subscriber receives the JPEG passthrough as a\n   *     graceful fallback (caller is expected to re-decode).\n   */\n  private fanoutDecodedFrame(frame: DecodedFrame): void {\n    const convertedCache = new Map<FrameFormat, DecodedFrame | Promise<DecodedFrame>>()\n    convertedCache.set(frame.format, frame)\n    for (const subscriber of this.decodedSubscribers.values()) {\n      if (!subscriber.frameDropper.shouldKeep()) {\n        subscriber.framesDropped++\n        continue\n      }\n      this.deliverConvertedFrame(subscriber, frame, convertedCache)\n    }\n  }\n\n  private deliverConvertedFrame(\n    subscriber: DecodedSubscriber,\n    frame: DecodedFrame,\n    cache: Map<FrameFormat, DecodedFrame | Promise<DecodedFrame>>,\n  ): void {\n    if (subscriber.format === frame.format) {\n      subscriber.callback(frame)\n      subscriber.framesDelivered++\n      return\n    }\n    const cached = cache.get(subscriber.format)\n    if (cached && !(cached instanceof Promise)) {\n      subscriber.callback(cached)\n      subscriber.framesDelivered++\n      return\n    }\n    if (cached instanceof Promise) {\n      cached.then((converted) => {\n        subscriber.callback(converted)\n        subscriber.framesDelivered++\n      }).catch((err: unknown) => {\n        subscriber.framesDropped++\n        this.logger?.warn('frame conversion failed', {\n          meta: { tag: subscriber.tag, target: subscriber.format, error: errMsg(err) },\n        })\n      })\n      return\n    }\n    const conversion = this.convertFrame(frame, subscriber.format)\n    cache.set(subscriber.format, conversion)\n    conversion.then((converted) => {\n      cache.set(subscriber.format, converted)\n      subscriber.callback(converted)\n      subscriber.framesDelivered++\n    }).catch((err: unknown) => {\n      subscriber.framesDropped++\n      this.logger?.warn('frame conversion failed', {\n        meta: { tag: subscriber.tag, target: subscriber.format, error: errMsg(err) },\n      })\n    })\n  }\n\n  private async convertFrame(source: DecodedFrame, target: FrameFormat): Promise<DecodedFrame> {\n    if (source.format === target) return source\n    const sharp = await getSharp()\n    if (source.format === 'rgb' || source.format === 'gray') {\n      const channels = source.format === 'gray' ? 1 : 3\n      const raw = { width: source.width, height: source.height, channels: channels as 1 | 3 }\n      const pipeline = sharp(source.data, { raw })\n      if (target === 'jpeg') {\n        const data = await pipeline.jpeg({ quality: 80, mozjpeg: false }).toBuffer()\n        return { ...source, data, format: 'jpeg' }\n      }\n      if (target === 'bgr') {\n        // Channel swap stays in-process — sharp's extractChannel + join\n        // would require an extra encode/decode, so handle it directly.\n        if (source.format !== 'rgb') {\n          throw new Error(`bgr conversion requires rgb source, got ${source.format}`)\n        }\n        const data = swapRedBlue(source.data)\n        return { ...source, data, format: 'bgr' }\n      }\n      if (target === 'gray') {\n        const data = await pipeline.toColorspace('b-w').raw().toBuffer()\n        return { ...source, data, format: 'gray' }\n      }\n      if (target === 'rgb') {\n        // gray → rgb: replicate the single luminance channel into three\n        // identical planes so consumers expecting RGB24 don't crash.\n        const data = await pipeline.toColorspace('srgb').raw().toBuffer()\n        return { ...source, data, format: 'rgb' }\n      }\n      if (target === 'yuv420') {\n        const data = await pipeline.toColorspace('yuv').raw().toBuffer()\n        return { ...source, data, format: 'yuv420' }\n      }\n    }\n    if (source.format === 'jpeg') {\n      // Decoder only picks `jpeg` when every subscriber wants jpeg,\n      // so a non-jpeg request here is a misconfiguration. Log and pass\n      // the JPEG through so the caller can decide what to do.\n      this.logger?.warn('jpeg-source frame requested as non-jpeg — passthrough', {\n        meta: { target },\n      })\n      return source\n    }\n    throw new Error(`unsupported conversion ${source.format} → ${target}`)\n  }\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\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 (clip creation, late-join)\n    if (packet.type === 'video') {\n      this.preBuffer.push(packet)\n    }\n\n    for (const cb of this.encodedCallbacks) {\n      cb(packet)\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 decoder or opening pull mode.\n      // H264 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        if (this.pendingPullModeOpen) {\n          this.pendingPullModeOpen()\n          this.pendingPullModeOpen = null\n        }\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 decoder after the first keyframe — the decoder needs SPS/PPS to start.\n    // Without this, streams that start mid-GOP silently produce 0 decoded frames.\n    if (packet.type === 'video' && this.firstKeyframeReceived) {\n      if (this.decoderProxy) {\n        this.decoderPushCount++\n        if (this.decoderPushCount === 1 || this.decoderPushCount % 500 === 0) {\n          this.logger?.info('decoder push', {\n            meta: {\n              pushCount: this.decoderPushCount,\n              size: packet.data.length,\n              keyframe: packet.keyframe,\n              decodedCount: this.decoderFrameCount,\n            },\n          })\n        }\n        this.decoderProxy.pushPacket(packet).catch((err: unknown) => {\n          this.logger?.warn('decoder push error', { meta: { error: errMsg(err) } })\n        })\n      }\n    }\n  }\n\n  onEncodedData(callback: (packet: EncodedPacket) => void): Unsubscribe {\n    this.encodedCallbacks.add(callback)\n    if (this.streamingDebug) {\n      this.logger?.info('encoded subscriber added', { meta: { total: this.encodedCallbacks.size } })\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', { meta: { total: this.encodedCallbacks.size } })\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) 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(callback: (rtpData: Buffer) => void): Unsubscribe {\n    this.rtpVideoCallbacks.add(callback)\n    if (this.streamingDebug) {\n      this.logger?.info('rtp video subscriber added', {\n        meta: { total: this.rtpVideoCallbacks.size },\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 },\n        })\n      }\n      this.checkDemand()\n    }\n  }\n\n  getSdpParameterSets(): ReadonlyArray<Buffer> | null {\n    return this.sdpParameterSets\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. Fans\n   * out to every `onVideoRtp` subscriber. Mirrors the dispatch the\n   * native RTSP reader performs in its `onVideoRtp` callback. No-ops\n   * when no subscribers are attached so providers can fire-and-forget.\n   */\n  pushVideoRtp(rtpData: Buffer): void {\n    if (this.stopping) return\n    if (this.rtpVideoCallbacks.size === 0) return\n    for (const cb of this.rtpVideoCallbacks) {\n      try {\n        cb(rtpData)\n      } catch (err) {\n        this.logger?.warn('rtp video subscriber threw', {\n          meta: { error: errMsg(err) },\n        })\n      }\n    }\n  }\n\n  /** Toggle the per-device streaming debug flag. */\n  setStreamingDebug(enabled: boolean): void {\n    this.streamingDebug = enabled\n  }\n\n  onDecodedFrame(callback: (frame: DecodedFrame) => void, options?: DecodeOptions): Unsubscribe {\n    const maxFps = options?.maxFps ?? DEFAULT_MAX_FPS\n    const subscriberId = Symbol('decoded-subscriber')\n    const frameDropper = new FrameDropper(maxFps)\n    const tag = options?.tag\n    if (!tag) {\n      this.logger?.warn(`onDecodedFrame called without tag — listClients will show 'unknown' (Phase 10: every caller must pass options.tag)`)\n    }\n\n    const subscriber: DecodedSubscriber = {\n      callback,\n      frameDropper,\n      tag: tag ?? 'unknown',\n      maxFps,\n      format: options?.format ?? 'jpeg',\n      subscribedAt: Date.now(),\n      framesDelivered: 0,\n      framesDropped: 0,\n    }\n    this.decodedSubscribers.set(subscriberId, subscriber)\n    this.logger?.info('decoded subscriber added', {\n      meta: { tag: subscriber.tag, total: this.decodedSubscribers.size, maxFps, format: subscriber.format },\n    })\n    this.checkDemand()\n\n    // Cancel pending teardown if a new subscriber joins while grace period is active\n    if (this.decoderTeardownTimer) {\n      clearTimeout(this.decoderTeardownTimer)\n      this.decoderTeardownTimer = undefined\n    }\n\n    // Create shared decoder on first subscriber (or reuse existing).\n    // If codec is not yet known (probe in progress), defer creation until codec is detected.\n    if (!this.decoderProxy && this.source) {\n      const codec = this.detectedCodec ?? this.source.videoCodec\n      if (codec) {\n        this.createSharedDecoderSession(options)\n      } else {\n        this.pendingDecodeOptions = options\n        this.logger?.info('Decoder creation deferred — waiting for codec probe')\n      }\n    } else if (this.decoderProxy) {\n      // Decoder already running — check if new subscriber requires a format upgrade\n      this.maybeUpgradeDecoderFormat()\n    }\n\n    return () => {\n      this.decodedSubscribers.delete(subscriberId)\n      this.logger?.info('decoded subscriber removed', { meta: { total: this.decodedSubscribers.size } })\n      this.checkDemand()\n\n      if (this.decodedSubscribers.size === 0 && this.decoderProxy) {\n        this.logger?.info('last decoded subscriber left — scheduling decoder teardown', {\n          meta: { graceMs: DECODER_TEARDOWN_GRACE_MS },\n        })\n        this.decoderTeardownTimer = setTimeout(() => {\n          this.decoderTeardownTimer = undefined\n          if (this.decodedSubscribers.size === 0 && this.decoderProxy) {\n            this.logger?.info('decoder teardown — no subscribers reconnected')\n            this.destroyDecoder()\n          }\n        }, DECODER_TEARDOWN_GRACE_MS)\n      }\n    }\n  }\n\n  onDecodedAudioChunk(\n    callback: (chunk: DecodedAudioChunk) => void,\n    options?: { readonly tag?: string },\n  ): Unsubscribe {\n    const tag = options?.tag\n    if (!tag) {\n      this.logger?.warn(`onDecodedAudioChunk called without tag — listClients will show 'unknown' (Phase 10)`)\n    }\n    const subscriberId = Symbol('audio-subscriber')\n    const subscriber: AudioSubscriber = {\n      callback,\n      tag: tag ?? 'unknown',\n      subscribedAt: Date.now(),\n      chunksDelivered: 0,\n    }\n    this.audioSubscribers.set(subscriberId, subscriber)\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\n    return () => {\n      this.audioSubscribers.delete(subscriberId)\n      this.checkDemand()\n    }\n  }\n\n  getStats(): BrokerStats {\n    const decoderStats = this._suspended ? undefined : this.cachedDecoderStats\n\n    return {\n      status: this._suspended ? 'idle' : this._status,\n      inputFps: decoderStats?.inputFps ?? this.encodedInputFps,\n      decodeFps: decoderStats?.outputFps ?? 0,\n      encodedSubscribers: this.encodedCallbacks.size,\n      decodedSubscribers: this.decodedSubscribers.size,\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.decoderNodeId,\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 pipeClients: number\n    readonly encodedSubscribers: number\n  } {\n    return {\n      rtsp: this.rtspRestreamer.listSessionInfos(),\n      decoded: [...this.decodedSubscribers.values()].map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        maxFps: s.maxFps,\n        framesDelivered: s.framesDelivered,\n        framesDropped: s.framesDropped,\n      })),\n      audio: [...this.audioSubscribers.values()].map((s) => ({\n        tag: s.tag,\n        subscribedAt: s.subscribedAt,\n        chunksDelivered: s.chunksDelivered,\n      })),\n      pipeClients: this.pipeServer.getClientCount(),\n      encodedSubscribers: this.encodedCallbacks.size,\n    }\n  }\n\n  /**\n   * Force-disconnect a single consumer. Matches by channel:\n   *   - `rtsp`: by `sessionId` (calls `RtspRestreamer.killSession`).\n   *   - `decoded` / `audio`: by `tag` — drops every subscriber whose tag\n   *     matches, then re-evaluates demand so the decoder can wind down\n   *     if nothing else is holding it.\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    const subscribers = channel === 'decoded' ? this.decodedSubscribers : this.audioSubscribers\n    const victims: symbol[] = []\n    for (const [key, sub] of subscribers) {\n      if (sub.tag === handle) victims.push(key)\n    }\n    if (victims.length === 0) return false\n    for (const key of victims) subscribers.delete(key)\n    this.logger?.info('subscribers force-disconnected', {\n      meta: { channel, handle, killed: victims.length, remaining: subscribers.size },\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          // Create deferred decoder session if subscribers were waiting\n          if (!this.decoderProxy && this.decodedSubscribers.size > 0 && this.pendingDecodeOptions !== undefined) {\n            this.logger?.info('native: codec detected — creating deferred decoder', {\n              meta: { codec: track.codec },\n            })\n            this.createSharedDecoderSession(this.pendingDecodeOptions)\n            this.pendingDecodeOptions = undefined\n          }\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\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          // 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 of this.rtpVideoCallbacks) {\n              try {\n                cb(rtpData)\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              for (const sub of this.audioSubscribers.values()) {\n                sub.callback(chunk)\n                sub.chunksDelivered++\n              }\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          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 of this.encodedCallbacks) {\n                cb(packet)\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  // ── 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  private destroyDecoder(): Promise<void> {\n    if (this.decoderResolveRetryTimer) {\n      clearTimeout(this.decoderResolveRetryTimer)\n      this.decoderResolveRetryTimer = undefined\n    }\n    this.decoderResolveAttempts = 0\n    const proxy = this.decoderProxy\n    this.decoderProxy = null\n    this.decoderNodeId = null\n    this.cachedDecoderStats = undefined\n    return proxy?.destroy() ?? Promise.resolve()\n  }\n\n  /**\n   * Moleculer nodeID of the decoder provider currently owning this broker's\n   * shared session. `null` when no session is active. Used by the\n   * broker-manager's `rotateDecodersOnNode` method to drop only the\n   * sessions affected by a per-agent event (e.g. hwaccel override change).\n   */\n  getDecoderNodeId(): string | null {\n    return this.decoderNodeId\n  }\n\n  /**\n   * Force the current shared decoder session to tear down. The next\n   * subscriber will trigger a fresh `createSharedDecoderSession` which\n   * re-pulls the latest per-agent preferences (hwaccel backend, …). Safe\n   * to call when there's no active session — resolves to a no-op.\n   */\n  async rotateDecoderSession(reason: string): Promise<void> {\n    if (!this.decoderProxy) return\n    this.logger?.info('rotating decoder session', { meta: { reason, decoderNodeId: this.decoderNodeId } })\n    await this.destroyDecoder()\n    // If subscribers are still attached, rebuild immediately so streaming\n    // resumes without waiting for a new subscribe event.\n    if (this.decodedSubscribers.size > 0 && this.source) {\n      const codec = this.detectedCodec ?? this.source.videoCodec\n      if (codec) {\n        this.createSharedDecoderSession(this.pendingDecodeOptions).catch((err: unknown) => {\n          this.logger?.warn('decoder session rebuild after rotation failed', { meta: { error: errMsg(err) } })\n        })\n      }\n    }\n  }\n\n  /**\n   * Poll the decoder resolver until a provider for `codec` shows up, then\n   * build the session. Addons register asynchronously and out-of-order:\n   * the broker must NOT fail when the decoder addon hasn't yet finished\n   * spawning its isolated process — it must wait.\n   *\n   * Bounded backoff: 500ms → 1s → 2s → 4s → 8s (capped), with a hard\n   * ceiling of ~60s total wait. If the decoder still isn't registered\n   * after that, log once at error level and stop retrying — at that\n   * point something is genuinely wrong with the addon config.\n   *\n   * The first failure is logged at debug so a healthy boot (decoder\n   * arriving within the first few hundred ms) doesn't pollute the log\n   * with noisy errors.\n   */\n  private scheduleDecoderResolveRetry(codec: string): void {\n    if (this.decoderResolveRetryTimer) return\n    const attempt = this.decoderResolveAttempts++\n    const backoffMs = Math.min(500 * Math.pow(2, attempt), 8000)\n    const cumulativeMs = (Math.pow(2, this.decoderResolveAttempts) - 1) * 500\n    if (cumulativeMs > 60000) {\n      this.logger?.error(\n        'No decoder provider — giving up. Check that addon-decoder-nodeav or addon-decoder-ffmpeg is installed and enabled.',\n        { meta: { codec, attempts: this.decoderResolveAttempts } },\n      )\n      this.decoderResolveAttempts = 0\n      return\n    }\n    if (attempt === 0) {\n      this.logger?.debug(\n        'Decoder not yet registered — will retry (addon boot race)',\n        { meta: { codec } },\n      )\n    }\n    this.decoderResolveRetryTimer = setTimeout(() => {\n      this.decoderResolveRetryTimer = undefined\n      // Only retry if we still have subscribers and no session yet — a\n      // teardown between schedule and fire is valid and should no-op.\n      if (this.decoderProxy) return\n      if (this.decodedSubscribers.size === 0) {\n        this.decoderResolveAttempts = 0\n        this.pendingDecodeOptions = undefined\n        return\n      }\n      this.createSharedDecoderSession(this.pendingDecodeOptions)\n    }, backoffMs)\n  }\n\n\n  private destroyNativeClient(): void {\n    if (this.nativeClient) {\n      this.nativeClient.destroy()\n      this.nativeClient = null\n    }\n    this.resetRtpPipeline()\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  }\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.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  private async createSharedDecoderSession(options?: DecodeOptions): Promise<void> {\n    const codec = this.detectedCodec ?? this.source?.videoCodec ?? 'h264'\n\n    if (!this.decoderApi) {\n      this.logger?.warn('no decoder API available — decoded frames will not be produced')\n      return\n    }\n\n    // Check if the decoder supports this codec before creating a session.\n    const supported = await this.decoderApi.supportsCodec({ codec })\n    if (!supported) {\n      // Decoder addon may not be registered yet (boot race).\n      this.pendingDecodeOptions = options\n      this.scheduleDecoderResolveRetry(codec)\n      return\n    }\n\n    // A provider is now available — clear any pending retry so we don't\n    // double-create the session on the next tick.\n    if (this.decoderResolveRetryTimer) {\n      clearTimeout(this.decoderResolveRetryTimer)\n      this.decoderResolveRetryTimer = undefined\n    }\n    this.decoderResolveAttempts = 0\n\n    // Shared decoder runs at unlimited FPS — each subscriber throttles locally.\n    // Use resolveDecoderFormat() to start with the cheapest format that\n    // satisfies all current subscribers (e.g. gray when only motion is active).\n    const resolvedFormat = this.resolveDecoderFormat()\n    // Stream-broker `deviceId` is actually the brokerId (`<numeric>/<profile>`).\n    // Parse out the numeric for the cap config so decoder logs land with\n    // a clean integer `deviceId` tag — the full brokerId goes through `tag`.\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    const config = {\n      codec,\n      maxFps: 0,\n      outputFormat: resolvedFormat,\n      scale: options?.scale ?? DEFAULT_SCALE,\n      ...(Number.isFinite(numericDeviceId) ? { deviceId: numericDeviceId } : {}),\n      tag: `broker:${this.deviceId}`,\n    }\n    this.decoderOutputFormat = resolvedFormat\n\n    const { sessionId, nodeId } = await this.decoderApi.createSession(config)\n    const proxy = new DecoderSessionProxy(this.decoderApi, sessionId)\n    this.decoderProxy = proxy\n    this.decoderNodeId = nodeId\n\n    // Fan-out decoded frames to all subscribers with per-subscriber\n    // throttling. In Phase 3 the decoder produces one *canonical*\n    // output (gray / rgb / jpeg per `resolveDecoderFormat`) and the\n    // broker derives any other format on demand via a per-frame\n    // conversion cache, so that an `rgb`-source frame fanned out to\n    // five `jpeg` subscribers performs exactly one sharp encode and\n    // pushes the same buffer to each. Subscribers that match the\n    // canonical format short-circuit the cache entirely.\n    const onFrame = (frame: DecodedFrame) => {\n      this.decoderFrameCount++\n      if (this.decoderFrameCount === 1) {\n        this.logger?.info('first decoded frame', {\n          meta: { width: frame.width, height: frame.height, format: frame.format },\n        })\n      }\n      // Update cached stats periodically (every 30 frames)\n      if (this.decoderFrameCount % 30 === 0) {\n        proxy.getStats().then((stats) => {\n          this.cachedDecoderStats = stats\n        }).catch(() => { /* stats refresh failure is non-fatal */ })\n      }\n      this.fanoutDecodedFrame(frame)\n    }\n\n    // Start the polling loop in the background — it runs until stopPolling or destroy\n    proxy.startPolling(onFrame).catch((err: unknown) => {\n      this.logger?.warn('decoder polling error', { meta: { error: errMsg(err) } })\n    })\n\n    // Pull mode: decoder reads from the broker's local pipe server (no duplicate RTSP connection)\n    // Must wait for the first keyframe so the pipe contains SPS/PPS for the decoder.\n    const info = await this.decoderApi.getInfo()\n    if (info.isPullMode) {\n      const localUrl = this.pipeServer.getUrl()\n      const isHevc = codec === 'h265' || codec === 'hevc'\n      const inputUrl = `${localUrl}?format=${isHevc ? 'hevc' : 'h264'}`\n\n      const doOpen = () => {\n        this.logger?.info('Pull mode decoder: opening local pipe', {\n          meta: { localUrl, codec },\n        })\n        proxy.openStream(inputUrl).catch((err: unknown) => {\n          this.logger?.error('Pull mode decoder failed', { meta: { error: errMsg(err) } })\n        })\n      }\n\n      if (this.firstKeyframeReceived) {\n        doOpen()\n      } else {\n        this.logger?.info('Pull mode decoder: waiting for first keyframe before opening')\n        this.pendingPullModeOpen = doOpen\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 `audioSubscribers` unchanged.\n      for (const sub of this.audioSubscribers.values()) {\n        sub.callback(chunk)\n        sub.chunksDelivered++\n      }\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 of this.encodedCallbacks) {\n            cb(packet)\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 * 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\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(brokerId: string, restreamer: RtspRestreamer, existingToken?: string): 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    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  }\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\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      const sdp = isMuted\n        ? (restreamer.getMutedSdp() ?? restreamer.getSdp() ?? '')\n        : (restreamer.getSdp() ?? '')\n      const session = new RtspSession(socket, sdp, () => {\n        restreamer.removeSession(session.getSessionId())\n      }, isMuted)\n      restreamer.addSession(session)\n\n      // Re-inject the buffered data so the session processes the first request\n      socket.unshift(Buffer.from(requestBuffer, 'ascii'))\n    }\n\n    socket.on('data', onData)\n    socket.on('error', () => {})\n  }\n}\n","import type { IScopedLogger } 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      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      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 * Transcode-pipeline manager — backs `streamBroker.getStreamWithCodec`.\n *\n * Responsibilities:\n *   1. Source-select: when the device already publishes a stream in the\n *      requested codec (and within `maxResolution`), return its URL\n *      without spawning anything.\n *   2. Spawn + refcount transcode pipelines: when transcode is needed,\n *      one ffmpeg child per unique (deviceId, videoCodec, audioCodec,\n *      resolution) tuple, multi-consumer via refcount. Stops the child\n *      N seconds after the refcount drops to 0 (`releaseGraceMs`).\n *   3. Pick the right encoder: consults\n *      `platform-probe.getHardwareEncoders` and falls back to\n *      `libx264` / `libx265`.\n *\n * Egress today: ffmpeg writes to a `rtsp://127.0.0.1:<port>/<token>`\n * loopback exposed by the broker's RTSP restream server.\n */\nimport { spawn, type ChildProcess } from 'node:child_process'\nimport { randomUUID } from 'node:crypto'\nimport { errMsg } from '@camstack/types'\nimport type {\n  CameraStream,\n  IScopedLogger,\n  AddonApi,\n  GetStreamWithCodecInput,\n  RtpSource,\n  VideoCodecTarget,\n  AudioCodecTarget,\n  HardwareEncoders,\n  HardwareEncoderId,\n} from '@camstack/types'\n\ntype CameraStreamLookup = (deviceId: number) => readonly CameraStream[]\ntype RtspEntryLookup = (brokerId: string) => { url: string } | null\n\ninterface PipelineEntry {\n  readonly key: string\n  readonly source: RtpSource\n  readonly child: ChildProcess | null\n  refcount: number\n  releaseTimer: NodeJS.Timeout | null\n}\n\nexport interface TranscodePipelineManagerOptions {\n  readonly logger: IScopedLogger\n  readonly api: AddonApi | null\n  readonly cameraStreamLookup: CameraStreamLookup\n  readonly rtspEntryLookup: RtspEntryLookup\n  readonly localRtspPort: () => number\n  readonly releaseGraceMs?: number\n  readonly ffmpegPath?: string\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 pipelineKeyFor(\n  deviceId: number,\n  videoCodec: 'H264' | 'H265',\n  audioCodec: AudioCodecTarget,\n  width: number,\n  height: number,\n): string {\n  return `${deviceId}|v=${videoCodec}|a=${audioCodec}|${width}x${height}`\n}\n\nfunction pickVideoEncoder(\n  target: 'H264' | 'H265',\n  encoders: HardwareEncoders | null,\n): HardwareEncoderId {\n  if (!encoders) return target === 'H264' ? 'libx264' : 'libx265'\n  return target === 'H264' ? encoders.defaultH264 : encoders.defaultH265\n}\n\nfunction pickAudioEncoderArgs(audio: AudioCodecTarget): readonly string[] {\n  switch (audio) {\n    case 'AAC':\n      return ['-c:a', 'aac', '-b:a', '128k', '-ar', '48000', '-ac', '2']\n    case 'Opus':\n      return ['-c:a', 'libopus', '-b:a', '64k', '-ar', '48000', '-ac', '2']\n    case 'PCMU':\n      return ['-c:a', 'pcm_mulaw', '-ar', '8000', '-ac', '1']\n    case 'none':\n      return ['-an']\n    default:\n      return ['-c:a', 'aac', '-b:a', '128k']\n  }\n}\n\nexport class TranscodePipelineManager {\n  private readonly entries = new Map<string, PipelineEntry>()\n  private readonly logger: IScopedLogger\n  private readonly api: AddonApi | null\n  private readonly cameraStreamLookup: CameraStreamLookup\n  private readonly rtspEntryLookup: RtspEntryLookup\n  private readonly localRtspPort: () => number\n  private readonly releaseGraceMs: number\n  private readonly ffmpegPath: string\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.localRtspPort = opts.localRtspPort\n    this.releaseGraceMs = opts.releaseGraceMs ?? 5_000\n    this.ffmpegPath = opts.ffmpegPath ?? 'ffmpeg'\n  }\n\n  async acquire(input: GetStreamWithCodecInput): Promise<RtpSource> {\n    const wantedVideo: VideoCodecTarget = input.videoCodec\n    const wantedAudio: AudioCodecTarget = input.audioCodec ?? 'AAC'\n\n    const direct = this.findDirectMatch(input.deviceId, wantedVideo, input.maxResolution)\n    if (direct) {\n      const key = pipelineKeyFor(\n        input.deviceId,\n        direct.videoCodec,\n        wantedAudio,\n        direct.resolution.width,\n        direct.resolution.height,\n      )\n      return this.shareOrCreate(key, () => ({\n        url: direct.url,\n        videoCodec: direct.videoCodec,\n        audioCodec: 'passthrough',\n        resolution: direct.resolution,\n        transcoded: false,\n        encoder: 'copy',\n        pipelineKey: key,\n      }), null)\n    }\n\n    const sourceStream = this.pickTranscodeSource(input.deviceId, input.maxResolution)\n    if (!sourceStream || !sourceStream.url) {\n      throw new Error(\n        `getStreamWithCodec: no source stream available for device ${input.deviceId}`,\n      )\n    }\n\n    const targetCodec: 'H264' | 'H265' = wantedVideo === 'auto'\n      ? (normaliseCodec(sourceStream.codec) ?? 'H264')\n      : wantedVideo\n    const targetResolution = input.maxResolution\n      ?? sourceStream.resolution\n      ?? { width: 1280, height: 720 }\n\n    const key = pipelineKeyFor(\n      input.deviceId,\n      targetCodec,\n      wantedAudio,\n      targetResolution.width,\n      targetResolution.height,\n    )\n\n    const existing = this.entries.get(key)\n    if (existing) {\n      return this.shareEntry(existing).source\n    }\n\n    const encoders = await this.loadHardwareEncoders()\n    const encoder = pickVideoEncoder(targetCodec, encoders)\n    const dialableUrl = this.allocateLoopbackUrl(input.deviceId, key)\n    const child = this.spawnFfmpeg({\n      sourceUrl: sourceStream.url,\n      outputUrl: dialableUrl,\n      encoder,\n      targetCodec,\n      width: targetResolution.width,\n      height: targetResolution.height,\n      audio: wantedAudio,\n      tag: input.tag ?? `stream-with-codec:${input.deviceId}`,\n    })\n\n    const source: RtpSource = {\n      url: dialableUrl,\n      videoCodec: targetCodec,\n      audioCodec: wantedAudio,\n      resolution: targetResolution,\n      transcoded: true,\n      encoder,\n      pipelineKey: key,\n    }\n    const entry: PipelineEntry = { key, source, child, refcount: 1, releaseTimer: null }\n    this.entries.set(key, entry)\n    this.logger.info('Transcode pipeline started', {\n      meta: { key, encoder, sourceUrl: sourceStream.url, outputUrl: dialableUrl },\n    })\n\n    if (child) {\n      child.once('exit', (code, signal) => {\n        this.logger.info('Transcode pipeline exited', {\n          meta: { key, code, signal },\n        })\n        const cur = this.entries.get(key)\n        if (cur && cur.child === child) {\n          this.entries.delete(key)\n        }\n      })\n    }\n    return source\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      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      this.shutdownEntry(entry)\n    }\n    this.entries.clear()\n  }\n\n  private findDirectMatch(\n    deviceId: number,\n    wanted: VideoCodecTarget,\n    maxRes: { width: number; height: number } | undefined,\n  ): { url: string; videoCodec: 'H264' | 'H265'; resolution: { width: number; height: number } } | null {\n    if (wanted === 'auto') return null\n    const streams = this.cameraStreamLookup(deviceId)\n    for (const s of streams) {\n      if (!s.url) continue\n      const codec = normaliseCodec(s.codec)\n      if (codec !== wanted) continue\n      if (maxRes && s.resolution) {\n        if (s.resolution.width > maxRes.width || s.resolution.height > maxRes.height) continue\n      }\n      const resolution = s.resolution ?? { width: 1280, height: 720 }\n      return { url: s.url, videoCodec: codec, resolution }\n    }\n    return null\n  }\n\n  private pickTranscodeSource(\n    deviceId: number,\n    maxRes: { width: number; height: number } | undefined,\n  ): CameraStream | null {\n    const streams = this.cameraStreamLookup(deviceId)\n    if (streams.length === 0) return null\n    if (!maxRes) return streams[0] ?? null\n    let best: CameraStream | null = null\n    let bestArea = -1\n    for (const s of streams) {\n      if (!s.url) continue\n      const w = s.resolution?.width ?? 1280\n      const h = s.resolution?.height ?? 720\n      if (w > maxRes.width || h > maxRes.height) continue\n      const area = w * h\n      if (area > bestArea) {\n        bestArea = area\n        best = s\n      }\n    }\n    if (best) return best\n    let smallest: CameraStream | null = null\n    let smallestArea = Number.MAX_SAFE_INTEGER\n    for (const s of streams) {\n      if (!s.url) continue\n      const w = s.resolution?.width ?? 1280\n      const h = s.resolution?.height ?? 720\n      const area = w * h\n      if (area < smallestArea) {\n        smallestArea = area\n        smallest = s\n      }\n    }\n    return smallest\n  }\n\n  private allocateLoopbackUrl(deviceId: number, pipelineKey: string): string {\n    const port = this.localRtspPort()\n    const token = `t-${deviceId}-${randomUUID().slice(0, 8)}`\n    void pipelineKey\n    if (!port) {\n      return `rtsp://0.0.0.0:0/${token}`\n    }\n    return `rtsp://127.0.0.1:${port}/${token}`\n  }\n\n  private async loadHardwareEncoders(): Promise<HardwareEncoders | null> {\n    if (!this.api) return null\n    try {\n      const api = this.api as Record<string, unknown>\n      const probe = api['platformProbe']\n      if (!probe || typeof probe !== 'object') return null\n      const fn = (probe as Record<string, unknown>)['getHardwareEncoders']\n      if (!fn || typeof fn !== 'object') return null\n      const query = (fn as Record<string, unknown>)['query']\n      if (typeof query !== 'function') return null\n      const callable = query as (input?: void) => Promise<HardwareEncoders>\n      return await callable()\n    } catch (err) {\n      this.logger.warn('platformProbe.getHardwareEncoders failed — using software fallback', {\n        meta: { error: errMsg(err) },\n      })\n      return null\n    }\n  }\n\n  private spawnFfmpeg(opts: {\n    sourceUrl: string\n    outputUrl: string\n    encoder: HardwareEncoderId\n    targetCodec: 'H264' | 'H265'\n    width: number\n    height: number\n    audio: AudioCodecTarget\n    tag: string\n  }): ChildProcess | null {\n    const args: string[] = [\n      '-hide_banner',\n      '-loglevel', 'warning',\n      '-rtsp_transport', 'tcp',\n      '-i', opts.sourceUrl,\n      '-vf', `scale=${opts.width}:${opts.height}`,\n      '-c:v', opts.encoder,\n      ...pickAudioEncoderArgs(opts.audio),\n      '-f', 'rtsp',\n      '-rtsp_transport', 'tcp',\n      opts.outputUrl,\n    ]\n\n    try {\n      const child = spawn(this.ffmpegPath, args, { stdio: ['ignore', 'ignore', 'pipe'] })\n      child.stderr?.setEncoding('utf8')\n      child.stderr?.on('data', (chunk: string) => {\n        this.logger.debug('ffmpeg', { meta: { tag: opts.tag, line: chunk.trim() } })\n      })\n      child.once('error', (err) => {\n        this.logger.error('ffmpeg spawn error', {\n          meta: { tag: opts.tag, error: errMsg(err) },\n        })\n      })\n      return child\n    } catch (err) {\n      this.logger.error('Failed to spawn ffmpeg', {\n        meta: { tag: opts.tag, error: errMsg(err) },\n      })\n      return null\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 shareOrCreate(\n    key: string,\n    factory: () => RtpSource,\n    child: ChildProcess | null,\n  ): 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, child, refcount: 1, releaseTimer: null }\n    this.entries.set(key, entry)\n    return source\n  }\n\n  private shutdownEntry(entry: PipelineEntry): void {\n    if (entry.releaseTimer) {\n      clearTimeout(entry.releaseTimer)\n      entry.releaseTimer = null\n    }\n    if (entry.child && !entry.child.killed) {\n      try {\n        entry.child.kill('SIGTERM')\n      } catch (err) {\n        this.logger.warn('ffmpeg kill error', {\n          meta: { key: entry.key, error: errMsg(err) },\n        })\n      }\n    }\n    this.entries.delete(entry.key)\n  }\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} 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 { RtspRestreamProviderImpl, type RtspRestreamEntry } from '../rtsp/rtsp-restream-provider.js'\nimport { TranscodePipelineManager } from './transcode-pipeline.js'\nimport { CAM_PROFILE_ORDER, DeviceFeature, EventCategory, errMsg, RingBuffer } from '@camstack/types'\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\ninterface 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\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 broker id from a `(deviceId, camStreamId)` tuple. */\nfunction brokerIdFor(deviceId: number, camStreamId: string): string {\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  /** 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  // 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  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  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        localRtspPort: () => this.rtspProvider.getRtspPort(),\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      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      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      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  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    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  }\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  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  }): Promise<{ success: true }> {\n    const { deviceId, camStreamId, kind, url, codec, resolution, fps, label, metadata } = input\n\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    // 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.\n    const urlRefreshed = existing !== undefined\n      && url !== undefined\n      && existing.url !== url\n    // Re-publish updates the feature snapshot — battery detection can\n    // happen post-creation (live probe), so keeping the latest signal\n    // matters more than the first.\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\n    // Re-run auto-assignment when the device has never had a manual\n    // assignment AND (a) profileMap is empty, or (b) auto flag is true.\n    // Re-publish of a known camStreamId with the SAME id keeps map stable.\n    const current = this.assignments.get(deviceId)\n    const streams = [...streamMap.values()].sort((a, b) => a.order - b.order)\n    const isAuto = current?.auto ?? true\n    let newMap: Partial<Record<CamProfile, string>>\n    if (!current || (isAuto && this.wouldChangeAutoMap(current.map, streams))) {\n      newMap = this.computeInitialAssignment(streams)\n    } else {\n      newMap = current.map\n    }\n\n    // Snapshot whether the just-published stream is already in a profile\n    // slot before applyAssignmentUpdate runs. If it is — and the slot\n    // still points at it after the update (i.e. no transition will fire\n    // demand naturally) — we need to re-emit demand explicitly. A\n    // re-publish from a freshly-restarted provider wipes its in-memory\n    // demand state, so the broker must re-tell it which streams are\n    // wanted. Demand handlers are idempotent on the provider side\n    // (`if (active.has(id)) return`), so re-emitting is always safe.\n    const prePublishProfile = PUSH_KINDS.has(kind)\n      ? findProfileForStream(current?.map ?? {}, camStreamId)\n      : null\n    const wasAlreadyAssigned = prePublishProfile !== null && existing !== undefined\n\n    await this.applyAssignmentUpdate(deviceId, newMap, { auto: isAuto, reason: '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    this.emitCamStreamsChanged(deviceId)\n    return { success: true as const }\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  // ── 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  // ── Cap methods: broker runtime (stats + client inventory) ──────────\n\n  async getBroker(input: { brokerId: string }): Promise<IStreamBroker | null> {\n    return this.brokers.get(input.brokerId) ?? null\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: [], 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      const nodeId = broker.getDecoderNodeId()\n      if (!nodeId) continue\n      const brokerAgent = nodeId.includes('/') ? nodeId.split('/')[0]! : nodeId\n      if (brokerAgent !== agentNodeId) continue\n      try {\n        await broker.rotateDecoderSession(reason)\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    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)\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)\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\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  // ── 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 \"Assignment\" — three profile→stream selectors ──────\n    const assignmentTab: ConfigSubTabDefinitionWithValue = {\n      id: 'assignment',\n      label: 'Assignment',\n      fields: 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 \"${label}\" profile slot`,\n        options: streamOptions,\n        span: 2 as const,\n        value: assignment.map[profile] ?? '',\n      })),\n    }\n\n    // ── One sub-tab per stream source with its broker settings ─────\n    // Every published cam stream owns a broker for as long as the\n    // source is published — there is no \"Active\" toggle. Pre-buffer\n    // is OFF by default and opt-in per stream.\n    const streamTabs: ConfigSubTabDefinitionWithValue[] = streams.map((stream) => {\n      const camStreamId = stream.camStreamId\n      const brokerId = brokerIdFor(deviceId, camStreamId)\n      const profile = findProfileForStream(assignment.map, camStreamId)\n      const rtspEnabled = this.rtspProvider.isEnabled(brokerId)\n      const rtspEntry = this.rtspProvider.getEntry(brokerId)\n      const perStream = override?.preBuffer?.[camStreamId]\n      const preBufferEnabled = perStream?.enabled ?? false\n      const preBufferSec = perStream?.seconds ?? this.defaultPreBufferSec\n\n      const fields: ConfigFieldWithValue[] = [\n        {\n          type: 'boolean' as const,\n          style: 'switch' as const,\n          key: `rtspEnabled:${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 streamBrokerSection: ConfigSectionWithValues = {\n      id: 'stream-broker-streams',\n      title: '',\n      tab: 'stream-broker',\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: [assignmentTab, ...streamTabs],\n        },\n      ],\n    }\n\n    const diagnosticsSection: ConfigSectionWithValues = {\n      id: 'stream-broker-diagnostics',\n      title: 'Diagnostics',\n      tab: 'stream-broker',\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    }\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(`${deviceId}/`)) {\n            broker.setStreamingDebug(next === true)\n          }\n        }\n        continue\n      }\n    }\n\n    if (overrideDirty) {\n      if (\n        nextOverride.preBufferSecOverride === undefined &&\n        (!nextOverride.preBuffer || Object.keys(nextOverride.preBuffer).length === 0) &&\n        nextOverride.streamingDebug === undefined\n      ) {\n        this.deviceOverrides.delete(deviceId)\n      } else {\n        this.deviceOverrides.set(deviceId, nextOverride)\n      }\n      this.deviceOverridePersister?.(this.deviceOverrides)\n    }\n    return { success: true as const }\n  }\n\n  // ── Private helpers ─────────────────────────────────────────────────\n\n  private nextOrder(deviceId: number): number {\n    const next = (this.streamOrderSeq.get(deviceId) ?? 0) + 1\n    this.streamOrderSeq.set(deviceId, next)\n    return next\n  }\n\n  private toCameraStream(s: CameraStreamInternal): CameraStream {\n    const { order: _order, deviceFeatures: _df, ...rest } = s\n    return rest\n  }\n\n  private deviceHasFeature(deviceId: number, feature: string): boolean {\n    const streamMap = this.cameraStreams.get(deviceId)\n    if (!streamMap) return false\n    for (const s of streamMap.values()) {\n      if (s.deviceFeatures.includes(feature)) return true\n    }\n    return false\n  }\n\n  private labelForDropdown(s: CameraStreamInternal): string {\n    const base = s.label ?? s.camStreamId\n    if (s.resolution) return `${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  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  }\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      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. Mirrors\n    // Scrypted's per-`getVideoStream()` `ensureRfcServer` resolution.\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    this.rtspServer.registerRestreamer(brokerId, broker.getRtspRestreamer(), existingToken)\n    if (!existingToken) this.rtspProvider.persistTokens()\n\n    await this.fanOutRegistration(deviceId, brokerId, cam, broker)\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.rtspProvider.persistTokens()\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    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) : `${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","/**\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 * 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  const replacements = await Promise.all(\n    [...hosts].map(async (host) => {\n      try {\n        const { address } = await dnsLookup(host, { family: 4 })\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', { 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 * Ported from Scrypted (BSD-2-Clause):\n *   https://github.com/koush/scrypted/blob/main/plugins/homekit/src/types/camera/jitter-buffer.ts\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 * Ported from Scrypted (BSD-2-Clause):\n *   https://github.com/koush/scrypted/blob/main/plugins/webrtc/src/h265-packetizer.ts\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 * 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 } from \"./h264-utils.js\";\nimport { convertH265ToAnnexB, isH265IrapAccessUnit } from \"./h265-utils.js\";\nimport { resolveMdnsCandidatesInSdp } from \"./mdns-resolve.js\";\nimport { H265Repacketizer, type H265CodecInfo } from \"./h265-repacketizer.js\";\nimport type {\n  WeriftModule,\n  WeriftPeerConnection,\n  WeriftMediaStreamTrack,\n  WeriftRtpSender,\n  WeriftTransceiver,\n  WeriftPcOptions,\n  WeriftIceServer,\n} from \"./werift-types.js\";\nimport { errMsg } from '@camstack/types'\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    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// 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  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  /** Callback for RTCP stats (called every ~3s). */\n  onStats?: (stats: SessionStats) => void;\n  /** Enable verbose frame/RTP logging. */\n  debug?: boolean;\n  /** Source video codec. Determines SDP negotiation + RTP packetization. */\n  sourceCodec?: 'H264' | 'H265';\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 source: FrameSource;\n  private readonly logger: Logger;\n  private readonly intercom: AdaptiveSessionOptions[\"intercom\"];\n  private readonly iceConfig: AdaptiveSessionOptions[\"iceConfig\"];\n  private readonly onStats: AdaptiveSessionOptions[\"onStats\"];\n  debug: boolean;\n  private readonly sourceCodec: 'H264' | 'H265';\n  /** Codec actually negotiated with the browser after SDP answer. */\n  private negotiatedCodec: 'H264' | 'H265' = 'H264';\n  /** True when source is H.265 but browser negotiated H.264 — needs transcode. */\n  get needsTranscode(): boolean { return this.sourceCodec === 'H265' && this.negotiatedCodec === 'H264' }\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  /** RTP sequence number counter (must increment per packet). */\n  private videoSeqNum = 0;\n  private audioSeqNum = 0;\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  private lastKeyframeRtpTs = 0;\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 — leave headroom for SRTP auth tag. */\n  private static readonly H265_REPACKETIZER_MTU = 1180;\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  constructor(options: AdaptiveSessionOptions) {\n    this.sessionId = options.sessionId;\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.createdAt = Date.now();\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    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\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      parameters: \"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\",\n      rtcpFeedback,\n    });\n    // H.265 fmtp — Scrypted ships the H.264-shaped fmtp string for H.265\n    // and it works. 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 the value Scrypted uses for parity with a known-\n    // good implementation.\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 required for BUNDLE demuxing and congestion control.\n      // Without sdes:mid, browsers cannot demux incoming RTP on a BUNDLE'd connection.\n      headerExtensions: {\n        video: [\n          { uri: \"urn:ietf:params:rtp-hdrext:sdes:mid\" },\n          { uri: \"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\" },\n          { uri: \"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\" },\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.iceConfig?.portRange) pcOptions.icePortRange = this.iceConfig.portRange;\n    if (this.iceConfig?.additionalHostAddresses?.length) {\n      pcOptions.iceAdditionalHostAddresses = [...this.iceConfig.additionalHostAddresses];\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, state } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\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    // Wait for ICE gathering to complete so the offer includes candidates.\n    // Without this, the offer has no candidates and the client can't connect.\n    await new Promise<void>((resolve) => {\n      if (this.pc?.iceGatheringState === \"complete\") { resolve(); return; }\n      this.pc?.iceGatheringStateChange.subscribe((state: string) => {\n        if (state === \"complete\") resolve();\n      });\n      // Timeout: don't wait forever if STUN is unreachable\n      setTimeout(resolve, 5000);\n    });\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.\n    const answerVideoCodec = resolvedSdp.match(/a=rtpmap:\\d+ (H264|H265)\\/90000/i)?.[1]?.toUpperCase()\n    if (answerVideoCodec === 'H264' || answerVideoCodec === 'H265') {\n      this.negotiatedCodec = answerVideoCodec as 'H264' | 'H265'\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    // 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      while (dtlsTransport.state !== \"connected\" && Date.now() < deadline) {\n        await new Promise<void>((r) => setTimeout(r, 100));\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(clientOffer: { sdp: string; type: \"offer\" }): Promise<{ sdp: string; type: \"answer\" }> {\n    const { werift, pcOptions } = await this.buildPcOptions();\n\n    this.pc = new werift.RTCPeerConnection(pcOptions);\n\n    this.pc.iceConnectionStateChange.subscribe((state: string) => {\n      this.logger.debug('ICE state', { meta: { phase: 'session', sessionId: this.sessionId, state } });\n      if (state === \"connected\") {\n        this.state = \"connected\";\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    // 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    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 tracks.\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        this.videoTrack = new werift.MediaStreamTrack({ kind: \"video\" });\n        await t.sender.replaceTrack(this.videoTrack);\n      } else if (kind === \"audio\" && !this.audioTrack) {\n        this.audioTrack = new werift.MediaStreamTrack({ kind: \"audio\" });\n        await t.sender.replaceTrack(this.audioTrack);\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      this.pc.addTransceiver(this.videoTrack, { direction: \"sendonly\" });\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      this.pc.addTransceiver(this.audioTrack, { direction: \"sendonly\" });\n    }\n\n    const answerDesc = await this.pc.createAnswer();\n    await this.pc.setLocalDescription(answerDesc);\n    this.state = \"connecting\";\n    this.logger.info('WHEP answer created', { meta: { phase: 'session', sessionId: this.sessionId } });\n\n    this.startFeedingFrames();\n\n    return { sdp: answerDesc.sdp, type: \"answer\" };\n  }\n\n  /** Add ICE candidate. */\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   * 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    if (this.negotiatedCodec !== 'H265') return;\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      AdaptiveSession.H265_REPACKETIZER_MTU,\n      codecInfo,\n    );\n  }\n\n  /**\n   * Forward a source RTP video packet (raw on-wire bytes) through the\n   * H.265 repacketizer to the browser.  Used by the broker's H.265\n   * direct-RTP subscription — bypasses the AnnexB→writeVideoNals\n   * path entirely.\n   *\n   * Drops everything until the SDP answer has been negotiated\n   * (`negotiatedCodec`/`videoSender` populated) and silently no-ops on\n   * non-H.265 sessions.\n   */\n  forwardSourceRtpVideo(rtpData: Buffer): void {\n    if (this.closed) return;\n    if (this.negotiatedCodec !== 'H265') return;\n    if (!this.videoSender || !_werift) return;\n    const werift = _werift;\n    this.ensureH265Repacketizer();\n    const rep = this.h265Repacketizer!;\n\n    let srcPkt;\n    try {\n      srcPkt = werift.RtpPacket.deSerialize(rtpData);\n    } catch (err) {\n      this.logger.warn('H265 RTP deserialize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err), len: rtpData.length },\n      });\n      return;\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 ?? 97;\n\n    let outPkts;\n    try {\n      outPkts = rep.repacketize(srcPkt);\n    } catch (err) {\n      this.logger.warn('H265 repacketize failed', {\n        meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) },\n      });\n      return;\n    }\n\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        this.rtpPacketsSent++;\n        if (this.debug && (this.rtpPacketsSent === 1 || this.rtpPacketsSent % 500 === 0)) {\n          this.logger.info('H265 source-RTP forwarded', {\n            meta: { phase: 'session', sessionId: this.sessionId, count: this.rtpPacketsSent },\n          });\n        }\n      } catch (err) {\n        if (this.rtpPacketsSent <= 10) {\n          this.logger.error('sendRtp (h265 forward) error', {\n            meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) },\n          });\n        }\n      }\n    }\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    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    // Sessions running the H.265 repacketizer forward source RTP\n    // through it — the upstream camera will respond to PLI by emitting\n    // a fresh IDR access unit on its own. Re-sending a cached one\n    // through writeVideoNals would collide with the repacketizer's\n    // outbound RTP stream. Push-mode H.265 (no repacketizer) keeps the\n    // cached-keyframe re-send path.\n    if (this.h265Repacketizer) return;\n\n    const now = Date.now();\n    if (now - this.lastPliResendAt < AdaptiveSession.PLI_RESEND_COOLDOWN_MS) return;\n    this.lastPliResendAt = now;\n\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    // Re-send all NALs via writeVideoNals — same path the feed loop\n    // uses for keyframes (single-NAL packets for VPS/SPS/PPS, FU\n    // fragmentation for IDR slice). All share the same RTP timestamp;\n    // marker bit set only on the last FU of the last NAL.\n    this.writeVideoNals(this.lastKeyframeNals, this.lastKeyframeRtpTs, 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      let videoTimestampBase: number | null = null;\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            if (videoTimestampBase === null) videoTimestampBase = frame.timestampMicros;\n            const rtpTs = Math.floor(\n              ((frame.timestampMicros - videoTimestampBase) * 90000) / 1_000_000,\n            ) >>> 0;\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. Scrypted does the same and 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                    this.lastKeyframeRtpTs = rtpTs\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) this.lastSps = Buffer.from(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                  this.lastKeyframeRtpTs = rtpTs;\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            // For G.711 (PCMU/PCMA): timestamp increments by frame size (160 samples per 20ms)\n            // Using a simple counter is more reliable than wall-clock based timestamps\n            if (audioTimestampBase === null) audioTimestampBase = 0;\n            audioTimestampBase = ((audioTimestampBase as number) + frame.data.length) >>> 0;\n            const rtpTs = audioTimestampBase as number;\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  /** Max RTP payload size (MTU 1200 to stay under typical network MTU). */\n  /**\n   * Transcode feed — source is H.265 but browser only supports H.264.\n   * Pipes raw H.265 Annex-B to ffmpeg stdin, reads H.264 from stdout.\n   */\n  private startTranscodeFeed(signal: AbortSignal): void {\n    const { spawn } = require('node:child_process') as typeof import('node:child_process')\n\n    this.logger.info('Starting H.265→H.264 transcode feed', {\n      meta: { phase: 'session', sessionId: this.sessionId },\n    })\n\n    const ff = spawn('ffmpeg', [\n      '-hide_banner', '-loglevel', 'error',\n      '-f', 'hevc', '-i', 'pipe:0',\n      '-c:v', 'libx264', '-preset', 'ultrafast', '-tune', 'zerolatency',\n      '-profile:v', 'baseline', '-g', '30', '-bf', '0',\n      '-an', '-f', 'h264', 'pipe:1',\n    ], { stdio: ['pipe', 'pipe', 'pipe'] })\n\n    signal.addEventListener('abort', () => {\n      if (!ff.killed) ff.kill('SIGTERM')\n    })\n\n    ff.on('exit', (code) => {\n      this.logger.info('Transcode ffmpeg exited', { meta: { phase: 'session', sessionId: this.sessionId, code } })\n    })\n    ff.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    // Parse H.264 output from ffmpeg and send via RTP\n    let pendingBuf = Buffer.alloc(0)\n    let rtpTs = 0\n    const FRAME_INTERVAL_90K = 3000 // 30fps → 90kHz ticks\n\n    ff.stdout.on('data', (chunk: Buffer) => {\n      pendingBuf = Buffer.concat([pendingBuf, chunk])\n      // Look for keyframe boundaries to emit complete access units\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 completedNals = splitAnnexBToNals(complete)\n      if (completedNals.length === 0) return\n\n      rtpTs = (rtpTs + FRAME_INTERVAL_90K) >>> 0\n      this.writeVideoNals(completedNals, rtpTs, 'H264')\n    })\n\n    // Feed source H.265 data 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 (!ff.stdin.destroyed) {\n              ff.stdin.write(mediaFrame.frame.data)\n            }\n          } else if (mediaFrame.type === 'audio') {\n            // Audio passthrough (no transcode needed)\n            const frame = mediaFrame.frame\n            this.writeAudio(frame.data, frame.timestampMicros, 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 (!ff.stdin.destroyed) ff.stdin.end()\n        if (!this.closed) void this.close()\n      }\n    })()\n  }\n\n  private static readonly MAX_RTP_PAYLOAD = 1200;\n\n  private rtpPacketsSent = 0;\n  private rtpLogCounter = 0;\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    // 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 (like Scrypted'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        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    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    } catch (err) {\n      this.logger.debug('Audio write error', { meta: { phase: 'session', sessionId: this.sessionId, error: errMsg(err) } });\n    }\n  }\n\n  // -----------------------------------------------------------------------\n  // RTCP stats collection\n  // -----------------------------------------------------------------------\n\n  private startStatsCollection(): void {\n    if (this.statsTimer || !this.onStats) 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 || !this.onStats) return;\n\n    try {\n      // werift exposes getReceivers() and getSenders() on RTCPeerConnection.\n      // For video sender stats, we check RTCP RR feedback from the receiver.\n      const senders = this.pc.getSenders?.() ?? [];\n\n      for (const sender of senders) {\n        const track = sender.track;\n        if (!track || track.kind !== \"video\") continue;\n\n        // werift tracks RTCP receiver reports via sender.rtcpReceiveReport\n        // or via the transport's RTCP event. The exact API depends on werift version.\n        // Fallback: use werift's built-in stats if available.\n        const report = sender.lastReceiverReport ?? sender.rtcpReport;\n        if (!report) continue;\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        const packetLoss = fractionLost / 256; // Fraction lost is 0–255\n\n        this.onStats({\n          sessionId: this.sessionId,\n          packetLoss,\n          jitterMs: jitter,\n          rttMs: rtt * 1000, // seconds → ms\n          packetsReceived: 0, // Not available from sender side\n          packetsLost,\n          timestamp: Date.now(),\n        });\n        return; // One video sender is enough\n      }\n\n      // Fallback: no RTCP report available yet — emit zeros\n      // (Client-reported stats will supplement this)\n    } catch {\n      // RTCP not yet available — normal during early connection\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","/**\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 } from '@camstack/types'\nimport { errMsg } from '@camstack/types'\nimport { AdaptiveSession, type IceServerEntry } from './session.js'\nimport type { FrameSource, MediaFrame } from './types.js'\nimport { convertH264ToAnnexB } from './h264-utils.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\ninterface SessionEntry {\n  readonly session: AdaptiveSession\n  readonly brokerId: string\n  /** Unsubscribe from broker's onEncodedData. */\n  readonly unsubBroker: Unsubscribe\n  /**\n   * Unsubscribe from broker's `onVideoRtp` for H.265 sessions that\n   * forward source RTP through `H265Repacketizer`. `null` for H.264\n   * sessions (which use the AnnexB feed-loop path).\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}\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\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: 'high' | 'mid' | 'low' | 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\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 (`'high' | 'mid' | 'low'`)\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) => 'high' | 'mid' | 'low' | null) | null = null\n  /** sessionId → session entry. */\n  private readonly sessions = new Map<string, SessionEntry>()\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  }\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) => 'high' | 'mid' | 'low' | 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 } = {},\n  ): Promise<{ sessionId: string; sdpOffer: string }> {\n    const setup = await this.setupSessionForBroker(streamId, hints, opts)\n    try {\n      const offer = await setup.session.createOffer()\n      setup.sessionLogger.info('WebRTC session created', {\n        meta: {\n          sessionId: setup.sessionId,\n          brokerId: setup.brokerId,\n          iceServers: setup.iceServerCount,\n          codec: setup.sessionCodec,\n          useH265Repacketizer: setup.useH265Repacketizer,\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 } = {},\n  ): Promise<{ sessionId: string; sdpAnswer: string }> {\n    const setup = await this.setupSessionForBroker(streamId, hints, opts)\n    try {\n      const answer = await setup.session.handleOffer({ sdp: clientOfferSdp, type: 'offer' })\n      setup.sessionLogger.info('WebRTC session created (client-offer)', {\n        meta: {\n          sessionId: setup.sessionId,\n          brokerId: setup.brokerId,\n          iceServers: setup.iceServerCount,\n          codec: setup.sessionCodec,\n          useH265Repacketizer: setup.useH265Repacketizer,\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 },\n  ): Promise<{\n    sessionId: string\n    session: AdaptiveSession\n    sessionLogger: IScopedLogger\n    brokerId: string\n    sessionCodec: 'H264' | 'H265'\n    useH265Repacketizer: boolean\n    iceServerCount: number\n  }> {\n    if (this.stopped) throw new Error('Server stopped')\n\n    const brokerId = this.resolveBrokerId(streamId, hints)\n    const broker = this.brokers.get(brokerId)\n    if (!broker) throw new Error(`No broker for stream \"${streamId}\" (resolved: \"${brokerId}\")`)\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 deviceTags = slashIdx > 0\n      ? {\n          deviceId: Number.parseInt(brokerId.slice(0, slashIdx), 10),\n          camStreamId: brokerId.slice(slashIdx + 1),\n        }\n      : { deviceId: -1, camStreamId: brokerId }\n    const sessionLogger = this.logger.withTags(deviceTags)\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    const useH265Repacketizer = sessionCodec === 'H265' && 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=${useH265Repacketizer}`,\n      { meta: { brokerId, sessionCodec, brokerCodec, sourceType, isRtp, useH265Repacketizer } },\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    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 (useH265Repacketizer) 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                timestampMicros: packet.pts,\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            timestampMicros: packet.pts,\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            timestampMicros: packet.pts,\n          },\n        })\n      }\n    })\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 (useH265Repacketizer) 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            timestampMicros: pkt.pts,\n          },\n        })\n      }\n    }\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. Collide-guard against a stale session\n    // already keyed on the same id — the caller should `closeSession`\n    // first if it really wants to reuse the id.\n    const requestedSessionId = opts.sessionId\n    if (requestedSessionId !== undefined && this.sessions.has(requestedSessionId)) {\n      throw new Error(`Session id already in use: ${requestedSessionId}`)\n    }\n    const sessionId = requestedSessionId ?? crypto.randomUUID()\n    const iceServers = await this.resolveIceServers()\n\n    const session = new AdaptiveSession({\n      sessionId,\n      source,\n      sourceCodec: sessionCodec,\n      iceConfig: {\n        iceServers,\n        portRange: this.icePortRange,\n        additionalHostAddresses: this.iceAdditionalHostAddresses,\n      },\n      logger: this.logger,\n      debug: opts.streamingDebug === true,\n    })\n\n    // For RTSP+H.265 sessions: subscribe to source-RTP and forward\n    // through the H265Repacketizer in the session. Also seed the\n    // repacketizer codec info from the SDP-derived VPS/SPS/PPS so the\n    // AP codec packet can be sent ahead of the first IDR even when\n    // the camera doesn't emit param sets in-band (Reolink RTSP\n    // high-profile streams). Push-mode H.265 (Reolink Baichuan, etc.)\n    // skips this and uses the AnnexB feed path.\n    let unsubRtp: Unsubscribe | null = null\n    let unsubSdpParams: Unsubscribe | null = null\n    if (useH265Repacketizer) {\n      const sdpPs = broker.getSdpParameterSets()\n      if (sdpPs && sdpPs.length > 0) {\n        session.seedH265CodecInfoFromSdp(sdpPs)\n        sessionLogger.info('H.265 session seeded from SDP at create-time', {\n          meta: { sessionId, brokerId, 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 AP codec packet, so\n        // Chrome 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('H.265 session deferred: SDP params not ready, subscribing for late delivery', {\n          meta: { sessionId, brokerId },\n        })\n        unsubSdpParams = broker.onSdpParameterSets((ps) => {\n          try {\n            session.seedH265CodecInfoFromSdp(ps)\n            sessionLogger.info('H.265 session seeded from SDP late delivery', {\n              meta: { sessionId, brokerId, psCount: ps.length },\n            })\n          } catch (err) {\n            sessionLogger.warn('seedH265CodecInfoFromSdp threw', {\n              meta: { sessionId, 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('H.265 session: first source RTP forwarded to repacketizer', {\n            meta: { sessionId, brokerId, bytes: rtpData.length },\n          })\n        }\n        try {\n          session.forwardSourceRtpVideo(rtpData)\n        } catch (err) {\n          sessionLogger.warn('forwardSourceRtpVideo threw', {\n            meta: { sessionId, error: errMsg(err) },\n          })\n        }\n      })\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    return {\n      sessionId,\n      session,\n      sessionLogger,\n      brokerId,\n      sessionCodec,\n      useH265Repacketizer,\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: ${sessionId}`)\n    await entry.session.handleAnswer({ sdp: sdpAnswer, type: 'answer' })\n  }\n\n  async closeSession(sessionId: string): Promise<void> {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.cleanupSession(sessionId)\n    await entry.session.close().catch(() => {})\n    this.logger.info('WebRTC session closed', { meta: { sessionId } })\n  }\n\n  async stop(): Promise<void> {\n    if (this.stopped) return\n    this.stopped = true\n    const closes: Promise<void>[] = []\n    for (const [sid, entry] of this.sessions) {\n      this.cleanupSession(sid)\n      closes.push(entry.session.close().catch(() => {}))\n    }\n    await Promise.all(closes)\n    this.brokers.clear()\n    this.logger.info('WebRTC server stopped')\n  }\n\n  getSessionCount(): number {\n    return this.sessions.size\n  }\n\n  // ── Private ─────────────────────────────────────────────────────────\n\n  private cleanupSession(sessionId: string): void {\n    const entry = this.sessions.get(sessionId)\n    if (!entry) return\n    this.sessions.delete(sessionId)\n    entry.unsubBroker()\n    entry.unsubRtp?.()\n    entry.unsubSdpParams?.()\n    entry.closeSource()\n  }\n\n  private resolveBrokerId(streamId: string, hints: ClientHints = {}): string {\n    // Direct match — `webrtc-session.createSession` resolves the\n    // discriminated `WebrtcStreamTarget` to a canonical brokerId\n    // (`${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: 'high' | 'mid' | 'low' | 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: 'high' | 'mid' | 'low' | 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\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 type {\n  CameraStream, ProfileSlot, CamProfile,\n  RtspRestreamEntry,\n  WebrtcClientHints, WebrtcStreamChoice, WebrtcStreamTarget,\n} from '@camstack/types'\nimport type { StreamBrokerManager } from './stream-broker/stream-broker-manager.js'\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\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   * 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 ${deviceId}`)\n        }\n        return `${deviceId}/${slot.sourceCamStreamId}`\n      }\n      case 'cam-stream':\n        return `${deviceId}/${target.camStreamId}`\n    }\n  }\n\n  async listStreams(input: { deviceId: number }): Promise<readonly WebrtcStreamChoice[]> {\n    const { deviceId } = input\n    const slots = this.manager.getProfileSlotsForDevice(deviceId)\n    const assignedSlots = slots.filter((s) => s.sourceCamStreamId !== null)\n    const camStreams = this.manager.getCameraStreamsForDevice(deviceId)\n    const camStreamById = new Map<string, CameraStream>()\n    for (const c of camStreams) camStreamById.set(c.camStreamId, c)\n\n    const choices: WebrtcStreamChoice[] = []\n\n    if (assignedSlots.length > 0) {\n      choices.push({\n        id: 'adaptive',\n        label: 'Adaptive',\n        target: { kind: 'adaptive' },\n        codec: null,\n        resolution: null,\n        status: null,\n        inputFps: null,\n        decodeFps: null,\n        bitrateKbps: null,\n      })\n    }\n\n    // Profile-tier choices: pin a session to high / mid / low. Source\n    // can change as the operator reassigns the slot — the session\n    // follows.\n    for (const slot of assignedSlots) {\n      const profile = slot.profile as CamProfile\n      const camStreamId = slot.sourceCamStreamId\n      if (!camStreamId) continue\n      const broker = await this.manager.getBroker({ brokerId: `${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: `${deviceId}/${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  }): Promise<{ sessionId: string; sdpOffer: string }> {\n    const streamId = this.resolveTargetToStreamId(input.deviceId, input.target)\n    const streamingDebug = typeof this.manager.isStreamingDebug === 'function'\n      ? this.manager.isStreamingDebug(input.deviceId)\n      : false\n    return this.webrtcServer.createSession(streamId, input.hints, { streamingDebug })\n  }\n\n  async handleOffer(input: {\n    deviceId: number\n    target?: WebrtcStreamTarget\n    sdpOffer: string\n    sessionId?: string\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 = typeof this.manager.isStreamingDebug === 'function'\n      ? this.manager.isStreamingDebug(input.deviceId)\n      : false\n    return this.webrtcServer.handleOffer(streamId, input.sdpOffer, undefined, {\n      streamingDebug,\n      sessionId: input.sessionId,\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 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","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 { StreamBrokerManager, type ProfileMapPersister } from './stream-broker/stream-broker-manager'\nimport { CameraStreamsProvider, WebrtcSessionProvider } from './device-scoped-providers'\nimport { BrokerWebrtcServer } from './webrtc/broker-webrtc-server'\nimport type { CamProfile } from '@camstack/types'\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\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  /**\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    })\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\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    rtspProvider.setTokenPersister((tokens) => {\n      const obj: Record<string, string> = {}\n      for (const [k, v] of tokens) { obj[k] = v }\n      this.ctx.settings?.writeAddonStore({ rtspTokens: obj }).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist RTSP tokens', { meta: { error: errMsg(err) } })\n      })\n    })\n\n    rtspProvider.setEnabledPersister((states) => {\n      const obj: Record<string, boolean> = {}\n      for (const [k, v] of states) { obj[k] = v }\n      this.ctx.settings?.writeAddonStore({ rtspEnabled: 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) persisted as numeric\n    // device ids. JSON serialises numeric Map keys as strings, so on\n    // read we reparse and drop anything that isn't a valid number.\n    try {\n      const addonStore = (await this.ctx.settings?.readAddonStore()) ?? {}\n      const overrideData = asJsonObject(addonStore['deviceOverrides'])\n      if (overrideData) {\n        const overrides = new Map<number, {\n          readonly preBufferSecOverride?: number\n          readonly preBuffer?: Record<string, { enabled: boolean; seconds: number }>\n          readonly streamingDebug?: boolean\n        }>()\n        for (const [rawKey, raw] of Object.entries(overrideData)) {\n          const numericKey = Number(rawKey)\n          if (!Number.isFinite(numericKey) || !Number.isInteger(numericKey)) continue\n          const o = asJsonObject(raw)\n          if (!o) continue\n          const entry: Record<string, unknown> = {}\n          if (typeof o['preBufferSecOverride'] === 'number') {\n            entry['preBufferSecOverride'] = o['preBufferSecOverride']\n          }\n          if (typeof o['streamingDebug'] === 'boolean') {\n            entry['streamingDebug'] = o['streamingDebug']\n          }\n          const pbRaw = asJsonObject(o['preBuffer'])\n          if (pbRaw) {\n            const preBuffer: Record<string, { enabled: boolean; seconds: number }> = {}\n            for (const [profile, v] of Object.entries(pbRaw)) {\n              const sv = asJsonObject(v)\n              if (sv && typeof sv['seconds'] === 'number') {\n                preBuffer[profile] = { enabled: sv['enabled'] !== false, seconds: sv['seconds'] }\n              }\n            }\n            entry['preBuffer'] = preBuffer\n          }\n          overrides.set(numericKey, entry)\n        }\n        this.brokerManager.loadPersistedDeviceOverrides(overrides)\n        this.ctx.logger.info('Loaded persisted device overrides', { meta: { overrideCount: overrides.size } })\n      }\n    } catch { /* first boot — no persisted state yet */ }\n\n    this.brokerManager.setDeviceOverridePersister((overrides) => {\n      const obj: Record<string, Record<string, unknown>> = {}\n      for (const [deviceId, ov] of overrides) {\n        const entry: Record<string, unknown> = {}\n        if (ov.preBufferSecOverride !== undefined) entry['preBufferSecOverride'] = ov.preBufferSecOverride\n        if (ov.streamingDebug !== undefined) entry['streamingDebug'] = ov.streamingDebug\n        if (ov.preBuffer) entry['preBuffer'] = ov.preBuffer\n        obj[`${deviceId}`] = entry\n      }\n      this.ctx.settings?.writeAddonStore({ deviceOverrides: 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    try {\n      const addonStore = (await this.ctx.settings?.readAddonStore()) ?? {}\n      const raw = asJsonObject(addonStore['profileMap'])\n      if (raw) {\n        const loaded = new Map<number, PersistedAssignment>()\n        for (const [rawKey, rawVal] of Object.entries(raw)) {\n          const deviceId = Number(rawKey)\n          if (!Number.isFinite(deviceId) || !Number.isInteger(deviceId)) continue\n          const val = asJsonObject(rawVal)\n          if (!val) continue\n          const mapObj = asJsonObject(val['map']) ?? {}\n          const map: Partial<Record<CamProfile, string>> = {}\n          for (const profile of PROFILE_KEYS) {\n            const v = mapObj[profile]\n            if (typeof v === 'string' && v.length > 0) map[profile] = v\n          }\n          const auto = val['auto'] !== false\n          loaded.set(deviceId, { map, auto })\n        }\n        this.brokerManager.loadPersistedProfileMap(loaded)\n        this.ctx.logger.info('Loaded persisted profile map', { meta: { deviceCount: loaded.size } })\n      }\n    } catch { /* first boot — no persisted state yet */ }\n\n    const profileMapPersister: ProfileMapPersister = (assignments) => {\n      const obj: Record<string, PersistedAssignment> = {}\n      for (const [deviceId, assignment] of assignments) {\n        obj[`${deviceId}`] = { map: { ...assignment.map }, auto: assignment.auto }\n      }\n      this.ctx.settings?.writeAddonStore({ profileMap: obj }).catch((err: unknown) => {\n        this.ctx.logger.warn('Failed to persist profile map', { meta: { error: errMsg(err) } })\n      })\n    }\n    this.brokerManager.setProfileMapPersister(profileMapPersister)\n\n    // Start RTSP restream server\n    await this.brokerManager.startRtspServer(this.config.rtspPort)\n\n    // Per-agent hwaccel override changed — force-rotate any active shared\n    // decoder session whose owning node matches.\n    this.subscribe(\n      { category: EventCategory.PipelineAgentHwaccelChanged },\n      (event) => {\n        const { agentNodeId, reason } = event.data\n        if (typeof agentNodeId !== 'string' || agentNodeId.length === 0) return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = typeof reason === 'string' ? reason : 'hwaccel-changed'\n        void manager.rotateDecodersOnNode(agentNodeId, reasonLabel).then((rotated) => {\n          if (rotated > 0) {\n            this.ctx.logger.info('decoders rotated for hwaccel change', {\n              tags: { nodeId: agentNodeId },\n              meta: { rotated, reason: reasonLabel },\n            })\n          }\n        }).catch((err: unknown) => {\n          this.ctx.logger.warn('hwaccel-driven decoder rotation failed', {\n            tags: { nodeId: agentNodeId },\n            meta: { error: errMsg(err) },\n          })\n        })\n      },\n    )\n\n    // Decoder down → rotate active shared sessions to a healthy provider.\n    this.watchCapability(['decoder'], {\n      onDown: (nodeId, capName) => {\n        if (capName !== 'decoder') return\n        const manager = this.brokerManager\n        if (!manager) return\n        const reasonLabel = `decoder-down:${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    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-orchestrator` cap (singleton). The orchestrator aggregates\n    // ICE servers from every enabled `turn-provider` (cloudflare-turn,\n    // future coturn / Twilio, …) — there's NO per-device round-trip;\n    // the orchestrator caches at the provider layer (cloudflare-turn\n    // refreshes ICE credentials in the background ~12 h before the\n    // 24 h TTL), so the per-session call costs a single cap dispatch +\n    // an in-process memory read. Without TURN configured the call\n    // returns `[]` and WebRTC falls back to host/srflx candidates.\n    const turnApi = this.ctx.api as unknown as {\n      turnOrchestrator?: {\n        getAllServers: { 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.turnOrchestrator) return []\n        try {\n          const servers = await turnApi.turnOrchestrator.getAllServers.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          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('turnOrchestrator.getAllServers failed — session will start without TURN', {\n            meta: { error: err instanceof Error ? err.message : String(err) },\n          })\n          return []\n        }\n      },\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\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          stableId: 'stream-broker-panel',\n          label: 'Stream Brokers',\n          description: 'Per-camera stream broker panel with adaptive controls.',\n          icon: 'radio',\n          remoteName: 'addon_stream_broker_widgets',\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    }\n\n    this.ctx.logger.info('Stream broker manager initialized')\n    const registrations: ProviderRegistration[] = [\n      { capability: streamBrokerCapability, provider: this.brokerManager },\n      {\n        capability: cameraStreamsCapability,\n        provider: cameraStreamsProvider,\n        kind: 'wrapper',\n        defaultActive: true,\n      },\n      {\n        capability: webrtcSessionCapability,\n        provider: webrtcSessionProvider,\n        kind: 'wrapper',\n        defaultActive: true,\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    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        },\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","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'\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    '-hide_banner', '-loglevel', '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}${toBase64Url(sig)}`\n}\n\nfunction verifyToken(secret: Buffer, token: string, expectedSessionId: string): AuthVerifyResult {\n  const parts = token.split(SEPARATOR)\n  if (parts.length !== 3) {\n    return { ok: false, reason: 'malformed' }\n  }\n  const [sessionId, expiryStr, signatureB64] = parts as [string, string, string]\n\n  const expiryEpochMs = Number(expiryStr)\n  if (!Number.isFinite(expiryEpochMs) || expiryEpochMs <= 0) {\n    return { ok: false, reason: 'malformed' }\n  }\n\n  if (sessionId !== expectedSessionId) {\n    return { ok: false, reason: 'wrong-session' }\n  }\n\n  const expectedSig = crypto.createHmac('sha256', secret).update(`${sessionId}${SEPARATOR}${expiryStr}`).digest()\n  const providedSig = fromBase64Url(signatureB64)\n\n  if (providedSig === null || providedSig.length !== expectedSig.length) {\n    return { ok: false, reason: 'bad-signature' }\n  }\n  if (!crypto.timingSafeEqual(providedSig, expectedSig)) {\n    return { ok: false, reason: 'bad-signature' }\n  }\n\n  if (Date.now() > expiryEpochMs) {\n    return { ok: false, reason: 'expired' }\n  }\n\n  return { ok: true, sessionId, expiresAt: expiryEpochMs }\n}\n\nfunction toBase64Url(buf: Buffer): string {\n  return buf.toString('base64').replace(/\\+/g, '-').replace(/\\//g, '_').replace(/=+$/, '')\n}\n\nfunction fromBase64Url(s: string): Buffer | null {\n  const normalized = s.replace(/-/g, '+').replace(/_/g, '/')\n  const padded = normalized + '='.repeat((4 - (normalized.length % 4)) % 4)\n  try {\n    return Buffer.from(padded, 'base64')\n  } catch {\n    return null\n  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