@pipecat-ai/moq-transport 0.1.0

package/dist/index.js.map

@@ -0,0 +1 @@
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* from \"./moqTransport\";\n","/*\n * Copyright (c) 2024-2026, Daily\n *\n * SPDX-License-Identifier: BSD-2-Clause\n */\n\nimport * as Json from \"@moq/json\";\nimport * as Moq from \"@moq/net\";\nimport * as Publish from \"@moq/publish\";\nimport { Effect, Signal } from \"@moq/signals\";\nimport * as Watch from \"@moq/watch\";\nimport {\n  type PipecatClientOptions,\n  type RTVIEventCallbacks,\n  RTVIMessage,\n  RTVIError,\n  Transport,\n  type Tracks,\n  type TransportState,\n} from \"@pipecat-ai/client-js\";\n\nconst DEFAULT_NAMESPACE = \"pipecat\";\nconst DEFAULT_CLIENT_ID = \"client0\";\nconst DEFAULT_BOT_ID = \"bot0\";\nconst DEFAULT_TRANSCRIPT_TRACK = \"transcript.json.z\";\n// Bounded jitter buffer on the audio decoder. Lower = more interactive\n// but more drops on bad networks. Matches the bot's audio_in_max_latency_ms\n// in spirit (each side enforces its own deadline).\nconst DEFAULT_AUDIO_LATENCY_MS = 80;\n// Latency ceiling for buffered TTS playback (ms). A finite cap (vs\n// uncapped) because the player's group/jitter buffer needs a concrete\n// maximum span to retain before dropping. The bot paces its writes a\n// little under this (~25s, see the Python transport's\n// `audio_out_max_buffer_ms`) so the producer self-limits below this drop\n// ceiling and the player never actually has to drop.\nconst DEFAULT_AUDIO_BUFFER_MAX_MS = 30 * 1000;\n// Sample rate we pin on the publish-side Opus encoder. Opus only supports\n// {8, 12, 16, 24, 48} kHz; 48 kHz is the upstream default and what most\n// browsers report as the mic's native rate. The bot reads this from our\n// catalog and resamples to its `audio_in_sample_rate` after decode, so we\n// don't need to match it exactly — but pinning it keeps the catalog\n// unambiguous.\nconst DEFAULT_AUDIO_SAMPLE_RATE = 48000;\n\n/**\n * Constructor options for the MoQ transport.\n *\n * The browser dials a MOQ peer (relay or bot in serve mode) at\n * ``relayUrl`` and publishes its mic under ``<namespace>/<clientId>``\n * while consuming the bot under ``<namespace>/<botId>``. Audio tracks\n * inside each broadcast are catalog-driven (codec, sample rate, channel\n * count are discovered at connect time), so they aren't pinned here.\n */\nexport interface MoqTransportOptions {\n  /**\n   * Full URL of the MOQ peer (e.g. ``https://relay.example.com:4080/moq``).\n   * WebTransport (HTTPS/HTTP3); @moq/net races WebSocket as a fallback\n   * if WebTransport isn't reachable.\n   */\n  relayUrl: string;\n\n  /**\n   * Pinned cert hashes for self-signed dev setups. Same shape as the\n   * native WebTransport ``serverCertificateHashes`` option.\n   */\n  serverCertificateHashes?: WebTransportHash[];\n\n  /**\n   * This client's participant id. Combined with ``namespace`` it forms\n   * the broadcast path the client publishes under: ``<namespace>/<clientId>``.\n   */\n  clientId?: string;\n\n  /**\n   * The peer (bot) participant id to consume. Subscriptions target\n   * ``<namespace>/<botId>``; track names come from the bot's catalog.\n   */\n  botId?: string;\n\n  /**\n   * Top-level namespace (analogous to a room name). Defaults to ``\"pipecat\"``.\n   */\n  namespace?: string;\n\n  /**\n   * Track name for RTVI server→client messages (transcripts, bot-ready,\n   * speech events, etc.). The client subscribes at\n   * ``<namespace>/<botId>/<transcriptTrack>``. Defaults to\n   * ``\"transcript.json.z\"``.\n   *\n   * Stays pinned because the transcript is a ``@moq/json`` stream track\n   * (an ordered, lossless append-log, wire-compatible with the bot's Rust\n   * ``moq_json::stream``), not a catalog rendition — the catalog only\n   * describes media tracks (audio/video). The ``.z`` suffix marks that the\n   * stream is compressed.\n   */\n  transcriptTrack?: string;\n\n  /**\n   * Latency floor (ms) — the jitter buffer the player keeps before\n   * playback. Lower = more interactive, more drops; higher = smoother,\n   * more glass-to-glass delay. Maps to the `min` bound of the\n   * ``@moq/watch`` ``Sync.latency`` range.\n   */\n  audioLatencyMs?: number;\n\n  /**\n   * Latency ceiling for buffered playback (the `max` bound of the\n   * ``@moq/watch`` ``Sync.latency`` range).\n   *\n   * The bot writes TTS audio faster than real-time with future-dated\n   * timestamps; the player buffers it and plays at the encoded pace\n   * instead of skipping ahead. This sets how much it's allowed to\n   * build up before re-anchoring:\n   *\n   * - a number (default: 30 s) — cap the buffer at that many ms. The\n   *   player retains up to this much faster-than-real-time audio before\n   *   dropping; an interruption (``user-started-speaking``) flushes early\n   *   via ``reset()``. The bot paces a little under this so it never\n   *   actually overruns the cap.\n   * - ``\"real-time\"`` — collapse to the floor (minimize latency, the old\n   *   skip-ahead behavior; only useful with a live, real-time publisher).\n   */\n  audioBufferMaxMs?: number | \"real-time\";\n\n  /**\n   * Sample rate (Hz) the client publishes its mic audio at. Must be one\n   * of Opus's supported rates: 8000, 12000, 16000, 24000, or 48000.\n   * Defaults to ``48000`` to match the typical browser mic rate.\n   *\n   * The bot reads this from the client's catalog and resamples to its\n   * own ``audio_in_sample_rate`` after Opus decode, so exact agreement\n   * isn't required — pinning a known value just keeps the catalog\n   * unambiguous.\n   *\n   * Bot-side playback sample rate isn't an option here: ``MoqTransport``\n   * reads whatever the bot advertises in its catalog and configures its\n   * playback ``AudioContext`` accordingly.\n   */\n  audioSampleRate?: number;\n}\n\ninterface ResolvedOptions {\n  relayUrl: string;\n  serverCertificateHashes?: WebTransportHash[];\n  clientId: string;\n  botId: string;\n  namespace: string;\n  transcriptTrack: string;\n  audioLatencyMs: number;\n  audioBufferMaxMs: number | \"real-time\";\n  audioSampleRate: number;\n}\n\nfunction applyDefaults(opts: MoqTransportOptions): ResolvedOptions {\n  return {\n    relayUrl: opts.relayUrl,\n    serverCertificateHashes: opts.serverCertificateHashes,\n    clientId: opts.clientId ?? DEFAULT_CLIENT_ID,\n    botId: opts.botId ?? DEFAULT_BOT_ID,\n    namespace: opts.namespace ?? DEFAULT_NAMESPACE,\n    transcriptTrack: opts.transcriptTrack ?? DEFAULT_TRANSCRIPT_TRACK,\n    audioLatencyMs: opts.audioLatencyMs ?? DEFAULT_AUDIO_LATENCY_MS,\n    audioBufferMaxMs: opts.audioBufferMaxMs ?? DEFAULT_AUDIO_BUFFER_MAX_MS,\n    audioSampleRate: opts.audioSampleRate ?? DEFAULT_AUDIO_SAMPLE_RATE,\n  };\n}\n\n/**\n * Shape of the bot's `/start` response when it's running the MoQ transport\n * (``pipecat.transports.moq.transport``). The bot nests its config under\n * ``moq`` so a single `/start` endpoint can serve multiple transport types\n * — apps pass this straight to ``PipecatClient.connect()`` with no\n * transport-specific glue on the app side.\n */\ninterface MoqStartResponse {\n  moq?: {\n    relayUrl: string;\n    certHash: string | null;\n    namespace: string;\n    clientId: string;\n    botId: string;\n    transcriptTrack: string;\n  };\n}\n\nfunction isMoqStartResponse(value: object): value is MoqStartResponse {\n  return \"moq\" in value;\n}\n\n/** Decode the bot's base64 SHA-256 cert hash into the ArrayBuffer shape\n *  ``WebTransport.serverCertificateHashes`` expects. */\nfunction decodeCertHash(b64: string): ArrayBuffer {\n  const bin = atob(b64);\n  const bytes = new Uint8Array(bin.length);\n  for (let i = 0; i < bin.length; i++) bytes[i] = bin.charCodeAt(i);\n  return bytes.buffer;\n}\n\n/**\n * ``MoqTransport`` — Pipecat Client SDK transport plugin for Media-over-QUIC.\n *\n * Built on the official ``moq`` library family:\n *\n * - ``@moq/net`` for connection management (``Connection.Reload``\n *   auto-reconnects on drops; races WebTransport + WebSocket).\n * - ``@moq/publish`` for mic capture and Opus encoding via\n *   ``Publish.Source.Microphone`` + ``Publish.Broadcast``.\n * - ``@moq/watch`` for bot audio playback: ``Watch.Broadcast`` discovers\n *   the catalog, ``Watch.Audio.Source`` picks a rendition, and\n *   ``Watch.Audio.Decoder`` + ``Watch.Audio.Emitter`` drive an\n *   ``AudioWorklet`` with a ring buffer (loss-tolerant, low-latency).\n * - ``@moq/json`` for the transcript tracks — the ``Stream`` mode: an\n *   ordered, lossless append-log (wire-compatible with the bot's Rust\n *   ``moq_json::stream``), so no RTVI event is dropped. Not the snapshot\n *   mode, which collapses to the latest value.\n * - ``@moq/signals`` for reactive plumbing.\n *\n * Two transcript tracks flow in opposite directions, both named\n * ``transcript.json.z`` (or whatever the shared ``transcriptTrack`` option\n * is set to) on their respective broadcasts:\n *\n * - Bot → client: bot publishes RTVI events (``bot-output``,\n *   ``bot-transcription``, ``bot-started-speaking``, ``metrics``, …);\n *   client subscribes and delivers to ``PipecatClient`` via ``_onMessage``.\n * - Client → bot: ``sendMessage`` writes each RTVI message on the client's\n *   own transcript track; the bot's Python transport\n *   (``pipecat.transports.moq.transport``) subscribes and dispatches them\n *   through ``RTVIProcessor`` the same way any other transport does. This\n *   is what carries ``client-ready`` for protocol-version negotiation,\n *   typed text input, function-call results, and any other client→server\n *   RTVI traffic.\n */\nexport class MoqTransport extends Transport {\n  public static SERVICE_NAME = \"moq-transport\";\n\n  // Resolved options (defaults applied), captured at construction time\n  // and overridden by `_validateConnectionParams` at connect time.\n  private _moqOptions: ResolvedOptions;\n\n  // Connection + reactive root.\n  private _reload: Moq.Connection.Reload | null = null;\n  private _signals: Effect | null = null;\n\n  // Publish side (mic → bot).\n  private _publishBroadcast: Publish.Broadcast | null = null;\n  private _microphone: Publish.Source.Microphone | null = null;\n  // RTVI JSON side-channel from client → bot, symmetric with the bot's\n  // outbound `transcript` track. Published as a lossless `@moq/json`\n  // append-stream (compression on); `sendMessage` appends each RTVI\n  // message and the bot's `_forward_peer_transcript` on the Python side\n  // drains it. The stream Producer binds to a single track with no\n  // built-in fan-out, and `publishTrack` serves each subscriber its own\n  // track, so we hold one Producer per subscription (`_transcriptOut`)\n  // and replay the message log (`_transcriptLog`) into each — so a bot\n  // that subscribes late still gets every message, in order.\n  private _transcriptOut: Set<Json.Stream.Producer<RTVIMessage>> | null = null;\n  private _transcriptLog: RTVIMessage[] | null = null;\n  private _micEnabled = new Signal(true);\n  private _micConstraints = new Signal<MediaTrackConstraints | undefined>(\n    undefined,\n  );\n  private _preferredMicId = new Signal<string | undefined>(undefined);\n  // Encoder sample rate is wired into the catalog the bot reads. We\n  // expose this as a Signal so `_connect` can update it from the resolved\n  // options at connect time.\n  private _audioSampleRate = new Signal<number | undefined>(undefined);\n\n  // Watch side (bot → playback). Buffered playback via @moq/watch:\n  // Broadcast -> Audio.Source -> Decoder -> Emitter, with `Sync.latencyMax`\n  // letting faster-than-real-time TTS build up instead of the player\n  // skipping ahead, and `reset()` flushing the buffer on interruption.\n  private _watchBroadcast: Watch.Broadcast | null = null;\n  private _sync: Watch.Sync | null = null;\n  private _audioSource: Watch.Audio.Source | null = null;\n  private _audioDecoder: Watch.Audio.Decoder | null = null;\n  private _audioEmitter: Watch.Audio.Emitter | null = null;\n\n  // Synthesized `MediaStreamTrack` for the bot, so `tracks().bot.audio`\n  // returns something a `<audio>` element / visualizer can consume.\n  // Tapped off `Watch.Audio.Decoder.root` (an `AudioNode`) into a\n  // `MediaStreamAudioDestinationNode`.\n  private _botAudioTrack: MediaStreamTrack | undefined;\n\n  // Transport lifecycle state. Mirrored to `_callbacks.onTransportStateChanged`.\n  declare protected _state: TransportState;\n\n  constructor(options: MoqTransportOptions) {\n    super();\n    this._moqOptions = applyDefaults(options);\n    this._state = \"disconnected\";\n  }\n\n  // --------------------------------------------------------------------\n  // Lifecycle\n  // --------------------------------------------------------------------\n\n  initialize(\n    options: PipecatClientOptions,\n    messageHandler: (ev: RTVIMessage) => void,\n  ): void {\n    this._options = options;\n    this._callbacks = options.callbacks ?? ({} as RTVIEventCallbacks);\n    this._onMessage = messageHandler;\n    this.state = \"initialized\";\n  }\n\n  async initDevices(): Promise<void> {\n    // Spin up the Microphone with permission requested so labels are\n    // available for the picker before the user clicks Connect. Real\n    // capture constraints (incl. sample rate) get applied in `_connect`\n    // once we have the resolved options.\n    if (!this._microphone) {\n      this._microphone = new Publish.Source.Microphone({\n        enabled: this._micEnabled,\n        constraints: this._micConstraints,\n        device: { preferred: this._preferredMicId },\n      });\n    }\n    this._microphone.device.requestPermission();\n  }\n\n  _validateConnectionParams(connectParams?: unknown): MoqTransportOptions {\n    if (connectParams === undefined || connectParams === null) {\n      return this._optionsAsInput();\n    }\n    if (typeof connectParams !== \"object\") {\n      throw new RTVIError(\"MoqTransport connect params must be an object\");\n    }\n    // The bot's `/start` response is `{ moq: {...} }`, not already-shaped\n    // `MoqTransportOptions` — unwrap and decode it here so apps can pass\n    // the raw response straight through with no transport-specific glue.\n    if (isMoqStartResponse(connectParams)) {\n      const { moq } = connectParams;\n      if (!moq) {\n        throw new RTVIError(\n          \"MoqTransport connect params did not include `moq` config (server may not be running with `-t moq`)\",\n        );\n      }\n      return {\n        ...this._optionsAsInput(),\n        relayUrl: moq.relayUrl,\n        clientId: moq.clientId,\n        botId: moq.botId,\n        namespace: moq.namespace,\n        transcriptTrack: moq.transcriptTrack,\n        serverCertificateHashes: moq.certHash\n          ? [{ algorithm: \"sha-256\", value: decodeCertHash(moq.certHash) }]\n          : undefined,\n      };\n    }\n    return { ...this._optionsAsInput(), ...(connectParams as Partial<MoqTransportOptions>) };\n  }\n\n  /** Return the resolved options re-shaped to the public input type\n   *  (used when merging with connect-time params). */\n  private _optionsAsInput(): MoqTransportOptions {\n    return { ...this._moqOptions };\n  }\n\n  async _connect(connectParams?: MoqTransportOptions): Promise<void> {\n    const merged = connectParams\n      ? { ...this._moqOptions, ...applyDefaults({ ...this._moqOptions, ...connectParams }) }\n      : this._moqOptions;\n    if (!merged.relayUrl) {\n      throw new RTVIError(\"MoqTransport requires `relayUrl`\");\n    }\n    this._moqOptions = merged;\n\n    this.state = \"connecting\";\n\n    let url: URL;\n    try {\n      url = new URL(merged.relayUrl);\n    } catch (err) {\n      this.state = \"error\";\n      throw new RTVIError(\n        `MoqTransport invalid relayUrl ${JSON.stringify(merged.relayUrl)}: ${\n          err instanceof Error ? err.message : String(err)\n        }`,\n      );\n    }\n\n    // @moq/net narrows `algorithm` to the literal `\"sha-256\"`, while\n    // the browser's `WebTransportHash` types it as `string`. Cast at\n    // the boundary; SHA-256 is the only algorithm WebTransport accepts.\n    const webtransport = merged.serverCertificateHashes\n      ? {\n          serverCertificateHashes: merged.serverCertificateHashes.flatMap(\n            (h) =>\n              h.value ? [{ algorithm: \"sha-256\" as const, value: h.value }] : [],\n          ),\n        }\n      : undefined;\n\n    // Reload auto-reconnects on disconnect; Publish.Broadcast and\n    // Watch.Broadcast both react to its `established` signal.\n    this._reload = new Moq.Connection.Reload({\n      enabled: new Signal(true),\n      url: new Signal(url),\n      webtransport,\n    });\n\n    // One reactive root for status mirroring. Connect/Watch are\n    // self-driving via their own internal effects.\n    this._signals = new Effect();\n    this._signals.run((eff) => {\n      const status = eff.get(this._reload!.status);\n      if (status === \"connected\") {\n        if (this._state === \"connecting\") this.state = \"connected\";\n      } else if (status === \"connecting\") {\n        this.state = \"connecting\";\n      } else if (status === \"disconnected\") {\n        if (this._state !== \"disconnecting\" && this._state !== \"disconnected\") {\n          this.state = \"disconnected\";\n        }\n      }\n    });\n\n    const ourPath = Moq.Path.from(merged.namespace, merged.clientId);\n    const botPath = Moq.Path.from(merged.namespace, merged.botId);\n\n    // ----------------------------------------------------------------\n    // Publish — Microphone owns getUserMedia, device selection, and\n    // produces a reactive `source` signal that Publish.Broadcast\n    // consumes. Constraints (channelCount:1, sampleRate) and preferred\n    // deviceId flow through the signals we hold a reference to, so\n    // updateMic() re-routes audio without re-creating the broadcast.\n    // The encoder's sampleRate Signal pins what the catalog advertises.\n    // ----------------------------------------------------------------\n    this._micConstraints.set({\n      channelCount: { exact: 1 },\n      sampleRate: { ideal: merged.audioSampleRate },\n      echoCancellation: true,\n      noiseSuppression: true,\n    });\n    this._audioSampleRate.set(merged.audioSampleRate);\n\n    if (!this._microphone) {\n      this._microphone = new Publish.Source.Microphone({\n        enabled: this._micEnabled,\n        constraints: this._micConstraints,\n        device: { preferred: this._preferredMicId },\n      });\n    }\n\n    this._publishBroadcast = new Publish.Broadcast({\n      connection: this._reload.established,\n      enabled: new Signal(true),\n      name: new Signal(ourPath),\n      audio: {\n        source: this._microphone.source,\n        enabled: this._micEnabled,\n        sampleRate: this._audioSampleRate,\n      },\n    });\n\n    // Client-side transcript: `sendMessage` appends each RTVI message to\n    // a lossless JSON append-stream. `publishTrack` hands us a fresh track\n    // per subscription (only one, in the normal single-bot flow), so we\n    // spin up a `Json.Stream.Producer` per subscriber and replay the\n    // message log into it — a bot that subscribes after we've already sent\n    // messages still gets the full log, in order. The bot subscribes to\n    // this track by its name — same convention as the bot's own transcript\n    // track — so no catalog entry is needed.\n    this._transcriptLog = [];\n    this._transcriptOut = new Set<Json.Stream.Producer<RTVIMessage>>();\n    this._publishBroadcast.publishTrack(\n      merged.transcriptTrack,\n      (track, effect) => {\n        const producer = new Json.Stream.Producer<RTVIMessage>(track, {\n          compression: true,\n        });\n        for (const msg of this._transcriptLog ?? []) producer.append(msg);\n        this._transcriptOut?.add(producer);\n        effect.cleanup(() => {\n          this._transcriptOut?.delete(producer);\n          producer.finish();\n        });\n      },\n    );\n\n    // Log the mic settings the browser actually granted, so we can see\n    // when a UA ignores the constraint (e.g. macOS often pins 48k\n    // regardless of `sampleRate.ideal`).\n    this._signals.run((eff) => {\n      const src = eff.get(this._microphone!.source);\n      if (!src) return;\n      const track = \"track\" in src ? src.track : src;\n      const s = track.getSettings();\n      console.log(\n        `[MoqTransport] publish: requested=${merged.audioSampleRate}Hz, ` +\n          `mic granted=${s.sampleRate}Hz, channels=${s.channelCount}, ` +\n          `deviceId=${s.deviceId}`,\n      );\n    });\n\n    // ----------------------------------------------------------------\n    // Watch — Broadcast handles catalog discovery and rendition\n    // tracking; Audio.Source picks the active audio rendition;\n    // Audio.Decoder runs the WebCodecs decode loop and feeds an\n    // AudioWorklet ring buffer; Audio.Emitter routes that to the\n    // speakers. `Sync.latencyMax` lets faster-than-real-time TTS build\n    // up a buffer instead of the player skipping ahead; `reset()`\n    // (invoked on `user-started-speaking`) flushes it on interruption.\n    // We also tap Decoder.root → MediaStreamAudioDestinationNode so\n    // tracks().bot.audio returns a MediaStreamTrack.\n    //\n    // `catalogFormat: \"hang\"` is pinned because the pipecat bot publishes\n    // a hang-format catalog (camelCase `sampleRate`). The auto-detector\n    // would also land on \"hang\" here (no suffix on the broadcast name,\n    // and hang is the DEFAULT_FORMAT) but pinning removes ambiguity if\n    // a future publisher adds a suffix or a different default ships.\n    // ----------------------------------------------------------------\n    this._watchBroadcast = new Watch.Broadcast({\n      connection: this._reload.established,\n      enabled: new Signal(true),\n      name: new Signal(botPath),\n      catalogFormat: new Signal<Watch.CatalogFormat>(\"hang\"),\n    });\n\n    // Latency range: floor = interactive jitter buffer (audioLatencyMs),\n    // ceiling = how much faster-than-real-time TTS the player will hold\n    // before dropping (audioBufferMaxMs). A number-typed max opens the\n    // buffer; \"real-time\" collapses to the floor (skip-ahead behavior).\n    const sync = new Watch.Sync({\n      connection: this._reload.established,\n      latency: new Signal<Watch.Latency>({\n        min: merged.audioLatencyMs as Moq.Time.Milli,\n        max:\n          merged.audioBufferMaxMs === \"real-time\"\n            ? \"real-time\"\n            : (merged.audioBufferMaxMs as Moq.Time.Milli),\n      }),\n    });\n    this._sync = sync;\n    this._audioSource = new Watch.Audio.Source(sync, {\n      broadcast: this._watchBroadcast,\n    });\n    this._audioDecoder = new Watch.Audio.Decoder(this._audioSource, {\n      enabled: new Signal(true),\n    });\n    this._audioEmitter = new Watch.Audio.Emitter(this._audioDecoder);\n\n    // Bridge Decoder.root (AudioNode) → MediaStreamTrack for tracks().bot.audio.\n    this._signals.run((eff) => {\n      const ctx = eff.get(this._audioDecoder!.context);\n      const root = eff.get(this._audioDecoder!.root);\n      if (!ctx || !root) return;\n      const dest = ctx.createMediaStreamDestination();\n      root.connect(dest);\n      this._botAudioTrack = dest.stream.getAudioTracks()[0];\n      eff.cleanup(() => {\n        try {\n          root.disconnect(dest);\n        } catch {\n          // best-effort.\n        }\n        this._botAudioTrack?.stop();\n        this._botAudioTrack = undefined;\n      });\n    });\n\n    // Log the bot's announced audio config + the AudioContext rate we\n    // ended up at. If these disagree the voice will play at the wrong\n    // pitch — the catalog drives the AudioContext, so a mismatch means\n    // either a parser bug or the bot is advertising a rate that doesn't\n    // match its actual Opus stream.\n    this._signals.run((eff) => {\n      const config = eff.get(this._audioSource!.config);\n      const ctx = eff.get(this._audioDecoder!.context);\n      if (!config && !ctx) return;\n      console.log(\n        `[MoqTransport] consume: catalog codec=${config?.codec}, ` +\n          `catalog sampleRate=${config?.sampleRate}Hz, ` +\n          `catalog channels=${config?.numberOfChannels}, ` +\n          `AudioContext rate=${ctx?.sampleRate}Hz`,\n      );\n    });\n\n    // Transcript — a lossless JSON append-stream over a single track. We\n    // re-subscribe on each (re)connect; @moq/json's stream Consumer yields\n    // every appended record in order, losslessly.\n    this._signals.run((eff) => {\n      const conn = eff.get(this._reload!.established);\n      if (!conn) return;\n      const botBroadcast = conn.consume(botPath);\n      const track = botBroadcast.subscribe(merged.transcriptTrack, 0);\n      const consumer = new Json.Stream.Consumer<RTVIMessage>(track, {\n        compression: true,\n      });\n      const ac = new AbortController();\n      this._drainTranscript(consumer, ac.signal).catch((e) => {\n        if (!ac.signal.aborted) {\n          console.warn(\"MoqTransport bot-transcript loop:\", e);\n        }\n      });\n      eff.cleanup(() => {\n        ac.abort();\n        try {\n          track.close();\n        } catch {\n          // best-effort.\n        }\n      });\n    });\n  }\n\n  async _disconnect(): Promise<void> {\n    if (this._state === \"disconnected\") return;\n    this.state = \"disconnecting\";\n    try {\n      this._audioEmitter?.close();\n      this._audioDecoder?.close();\n      this._audioSource?.close();\n      this._sync?.close();\n      this._watchBroadcast?.close();\n      // Each producer's own effect.cleanup (registered in the `publishTrack`\n      // callback above) already calls producer.finish() when this cascades\n      // through the broadcast's internal effect scope — no need to also\n      // finish them explicitly here.\n      this._publishBroadcast?.close();\n      this._microphone?.close();\n      this._signals?.close();\n      this._reload?.close();\n    } finally {\n      this._audioEmitter = null;\n      this._audioDecoder = null;\n      this._audioSource = null;\n      this._sync = null;\n      this._watchBroadcast = null;\n      this._transcriptOut = null;\n      this._transcriptLog = null;\n      this._publishBroadcast = null;\n      this._microphone = null;\n      this._signals = null;\n      this._reload = null;\n      this.state = \"disconnected\";\n      // Fire the SDK's disconnected callback. `state = \"disconnected\"`\n      // above already triggers `onTransportStateChanged`, but the SDK\n      // wires UI teardown (voice-ui-kit's connection-state hook, the\n      // `Disconnected` event, message-dispatcher cleanup) through\n      // `onDisconnected` — without this call the transport goes idle\n      // but the UI stays stuck in \"connected\" and never restores its\n      // \"Connect\" button. Matches the pattern used by\n      // small-webrtc-transport, openai-realtime-webrtc-transport, and\n      // gemini-live-websocket-transport.\n      this._callbacks?.onDisconnected?.();\n    }\n  }\n\n  sendReadyMessage(): void {\n    // Flip transport state to `ready` and dispatch the RTVI `client-ready`\n    // message on the client→bot transcript track so the bot's RTVIProcessor\n    // can negotiate protocol version. Mirrors the pattern used by\n    // small-webrtc-transport and daily-transport.\n    this.state = \"ready\";\n    this.sendMessage(RTVIMessage.clientReady());\n  }\n\n  // --------------------------------------------------------------------\n  // State\n  // --------------------------------------------------------------------\n\n  get state(): TransportState {\n    return this._state;\n  }\n\n  set state(next: TransportState) {\n    if (this._state === next) return;\n    this._state = next;\n    this._callbacks?.onTransportStateChanged?.(next);\n  }\n\n  // --------------------------------------------------------------------\n  // Devices\n  // --------------------------------------------------------------------\n\n  async getAllMics(): Promise<MediaDeviceInfo[]> {\n    return this._microphone?.device.available.peek() ?? [];\n  }\n\n  async getAllCams(): Promise<MediaDeviceInfo[]> {\n    return [];\n  }\n\n  async getAllSpeakers(): Promise<MediaDeviceInfo[]> {\n    return [];\n  }\n\n  updateMic(micId: string): void {\n    this._preferredMicId.set(micId);\n  }\n\n  updateCam(_camId: string): void {}\n\n  updateSpeaker(_speakerId: string): void {}\n\n  get selectedMic(): MediaDeviceInfo | Record<string, never> {\n    const id = this._microphone?.device.active.peek();\n    if (!id) return {};\n    return (\n      this._microphone?.device.available.peek()?.find((d) => d.deviceId === id) ??\n      {}\n    );\n  }\n\n  get selectedCam(): MediaDeviceInfo | Record<string, never> {\n    return {};\n  }\n\n  get selectedSpeaker(): MediaDeviceInfo | Record<string, never> {\n    return {};\n  }\n\n  enableMic(enable: boolean): void {\n    this._micEnabled.set(enable);\n  }\n\n  enableCam(_enable: boolean): void {}\n\n  enableScreenShare(_enable: boolean): void {}\n\n  get isCamEnabled(): boolean {\n    return false;\n  }\n\n  get isMicEnabled(): boolean {\n    return this._micEnabled.get();\n  }\n\n  get isSharingScreen(): boolean {\n    return false;\n  }\n\n  // --------------------------------------------------------------------\n  // Messaging + tracks\n  // --------------------------------------------------------------------\n\n  sendMessage(message: RTVIMessage): void {\n    if (!this._transcriptOut || !this._transcriptLog) {\n      console.warn(\n        \"[MoqTransport] sendMessage called before connect; dropping\",\n        message,\n      );\n      return;\n    }\n    // Append to the lossless stream. Keep a log too, so a bot that\n    // subscribes after this point is replayed every message in order\n    // (see the `publishTrack` serve callback above).\n    this._transcriptLog.push(message);\n    for (const producer of this._transcriptOut) producer.append(message);\n  }\n\n  tracks(): Tracks {\n    const localSource = this._microphone?.source.peek();\n    const localAudio =\n      localSource && \"track\" in localSource ? localSource.track : localSource;\n    return {\n      local: localAudio ? { audio: localAudio } : {},\n      bot: this._botAudioTrack ? { audio: this._botAudioTrack } : {},\n    };\n  }\n\n  // --------------------------------------------------------------------\n  // Internals — transcript\n  // --------------------------------------------------------------------\n\n  /** Pull RTVI messages off the @moq/json stream consumer and hand each\n   *  one to `PipecatClient` via `_onMessage`. The handler resolves the\n   *  SDK's connect promise when bot-ready arrives. */\n  private async _drainTranscript(\n    consumer: Json.Stream.Consumer<RTVIMessage>,\n    signal: AbortSignal,\n  ): Promise<void> {\n    for (;;) {\n      const message = await consumer.next();\n      if (!message || signal.aborted) break;\n      if (typeof message !== \"object\") continue;\n      // Intra-transport shutdown signal from the bot: bot is about to\n      // close its MoQ session. Disable auto-reconnect, tear down the\n      // audio decoder + subscribers on our side while WebTransport is\n      // still healthy — without this the connection just vanishes\n      // underneath Chrome's WebCodecs decoder and can crash the\n      // renderer. Not an RTVI event; don't forward it to `_onMessage`.\n      const label = (message as { label?: string }).label;\n      const type = (message as { type?: string }).type;\n      if (label === \"moq-transport\" && type === \"session-ending\") {\n        console.log(\"[MoqTransport] bot signaled session-ending — closing cleanly\");\n        // Fire-and-forget; we're returning from the drain loop right\n        // after, and _disconnect() aborts the consumer that owns us.\n        void this._disconnect();\n        return;\n      }\n      // Interruption: the user started talking, so flush the\n      // already-buffered TTS for the bot's previous utterance instead\n      // of letting it drain. With the bot writing faster than\n      // real-time, that buffer can be seconds long.\n      if (type === \"user-started-speaking\") {\n        this._resetBotAudio();\n      }\n      this._onMessage?.(message);\n    }\n  }\n\n  /** Flush buffered bot audio and re-anchor playback at an utterance\n   *  boundary. Called on interruption (`user-started-speaking`) so the\n   *  already-buffered TTS for the previous utterance stops immediately\n   *  instead of draining: re-anchor the sync reference and flush the\n   *  decoder's ring buffer. */\n  private _resetBotAudio(): void {\n    this._sync?.reset();\n    this._audioDecoder?.reset();\n  }\n}\n"],"names":[],"version":3,"file":"index.js.map"}

package/dist/index.module.js

@@ -0,0 +1,557 @@
+import {Stream as $3Y32T$Stream} from "@moq/json";
+import {Connection as $3Y32T$Connection, Path as $3Y32T$Path} from "@moq/net";
+import {Source as $3Y32T$Source, Broadcast as $3Y32T$Broadcast} from "@moq/publish";
+import {Signal as $3Y32T$Signal, Effect as $3Y32T$Effect} from "@moq/signals";
+import {Broadcast as $3Y32T$Broadcast1, Sync as $3Y32T$Sync, Audio as $3Y32T$Audio} from "@moq/watch";
+import {RTVIError as $3Y32T$RTVIError, RTVIMessage as $3Y32T$RTVIMessage, Transport as $3Y32T$Transport} from "@pipecat-ai/client-js";
+
+
+function $parcel$export(e, n, v, s) {
+  Object.defineProperty(e, n, {get: v, set: s, enumerable: true, configurable: true});
+}
+var $549dd95e7ad3e81f$exports = {};
+
+$parcel$export($549dd95e7ad3e81f$exports, "MoqTransport", () => $549dd95e7ad3e81f$export$739cde4b077543d1);
+/*
+ * Copyright (c) 2024-2026, Daily
+ *
+ * SPDX-License-Identifier: BSD-2-Clause
+ */ 
+
+
+
+
+
+const $549dd95e7ad3e81f$var$DEFAULT_NAMESPACE = "pipecat";
+const $549dd95e7ad3e81f$var$DEFAULT_CLIENT_ID = "client0";
+const $549dd95e7ad3e81f$var$DEFAULT_BOT_ID = "bot0";
+const $549dd95e7ad3e81f$var$DEFAULT_TRANSCRIPT_TRACK = "transcript.json.z";
+// Bounded jitter buffer on the audio decoder. Lower = more interactive
+// but more drops on bad networks. Matches the bot's audio_in_max_latency_ms
+// in spirit (each side enforces its own deadline).
+const $549dd95e7ad3e81f$var$DEFAULT_AUDIO_LATENCY_MS = 80;
+// Latency ceiling for buffered TTS playback (ms). A finite cap (vs
+// uncapped) because the player's group/jitter buffer needs a concrete
+// maximum span to retain before dropping. The bot paces its writes a
+// little under this (~25s, see the Python transport's
+// `audio_out_max_buffer_ms`) so the producer self-limits below this drop
+// ceiling and the player never actually has to drop.
+const $549dd95e7ad3e81f$var$DEFAULT_AUDIO_BUFFER_MAX_MS = 30000;
+// Sample rate we pin on the publish-side Opus encoder. Opus only supports
+// {8, 12, 16, 24, 48} kHz; 48 kHz is the upstream default and what most
+// browsers report as the mic's native rate. The bot reads this from our
+// catalog and resamples to its `audio_in_sample_rate` after decode, so we
+// don't need to match it exactly — but pinning it keeps the catalog
+// unambiguous.
+const $549dd95e7ad3e81f$var$DEFAULT_AUDIO_SAMPLE_RATE = 48000;
+function $549dd95e7ad3e81f$var$applyDefaults(opts) {
+    return {
+        relayUrl: opts.relayUrl,
+        serverCertificateHashes: opts.serverCertificateHashes,
+        clientId: opts.clientId ?? $549dd95e7ad3e81f$var$DEFAULT_CLIENT_ID,
+        botId: opts.botId ?? $549dd95e7ad3e81f$var$DEFAULT_BOT_ID,
+        namespace: opts.namespace ?? $549dd95e7ad3e81f$var$DEFAULT_NAMESPACE,
+        transcriptTrack: opts.transcriptTrack ?? $549dd95e7ad3e81f$var$DEFAULT_TRANSCRIPT_TRACK,
+        audioLatencyMs: opts.audioLatencyMs ?? $549dd95e7ad3e81f$var$DEFAULT_AUDIO_LATENCY_MS,
+        audioBufferMaxMs: opts.audioBufferMaxMs ?? $549dd95e7ad3e81f$var$DEFAULT_AUDIO_BUFFER_MAX_MS,
+        audioSampleRate: opts.audioSampleRate ?? $549dd95e7ad3e81f$var$DEFAULT_AUDIO_SAMPLE_RATE
+    };
+}
+function $549dd95e7ad3e81f$var$isMoqStartResponse(value) {
+    return "moq" in value;
+}
+/** Decode the bot's base64 SHA-256 cert hash into the ArrayBuffer shape
+ *  ``WebTransport.serverCertificateHashes`` expects. */ function $549dd95e7ad3e81f$var$decodeCertHash(b64) {
+    const bin = atob(b64);
+    const bytes = new Uint8Array(bin.length);
+    for(let i = 0; i < bin.length; i++)bytes[i] = bin.charCodeAt(i);
+    return bytes.buffer;
+}
+class $549dd95e7ad3e81f$export$739cde4b077543d1 extends (0, $3Y32T$Transport) {
+    constructor(options){
+        super();
+        // Connection + reactive root.
+        this._reload = null;
+        this._signals = null;
+        // Publish side (mic → bot).
+        this._publishBroadcast = null;
+        this._microphone = null;
+        // RTVI JSON side-channel from client → bot, symmetric with the bot's
+        // outbound `transcript` track. Published as a lossless `@moq/json`
+        // append-stream (compression on); `sendMessage` appends each RTVI
+        // message and the bot's `_forward_peer_transcript` on the Python side
+        // drains it. The stream Producer binds to a single track with no
+        // built-in fan-out, and `publishTrack` serves each subscriber its own
+        // track, so we hold one Producer per subscription (`_transcriptOut`)
+        // and replay the message log (`_transcriptLog`) into each — so a bot
+        // that subscribes late still gets every message, in order.
+        this._transcriptOut = null;
+        this._transcriptLog = null;
+        this._micEnabled = new (0, $3Y32T$Signal)(true);
+        this._micConstraints = new (0, $3Y32T$Signal)(undefined);
+        this._preferredMicId = new (0, $3Y32T$Signal)(undefined);
+        // Encoder sample rate is wired into the catalog the bot reads. We
+        // expose this as a Signal so `_connect` can update it from the resolved
+        // options at connect time.
+        this._audioSampleRate = new (0, $3Y32T$Signal)(undefined);
+        // Watch side (bot → playback). Buffered playback via @moq/watch:
+        // Broadcast -> Audio.Source -> Decoder -> Emitter, with `Sync.latencyMax`
+        // letting faster-than-real-time TTS build up instead of the player
+        // skipping ahead, and `reset()` flushing the buffer on interruption.
+        this._watchBroadcast = null;
+        this._sync = null;
+        this._audioSource = null;
+        this._audioDecoder = null;
+        this._audioEmitter = null;
+        this._moqOptions = $549dd95e7ad3e81f$var$applyDefaults(options);
+        this._state = "disconnected";
+    }
+    // --------------------------------------------------------------------
+    // Lifecycle
+    // --------------------------------------------------------------------
+    initialize(options, messageHandler) {
+        this._options = options;
+        this._callbacks = options.callbacks ?? {};
+        this._onMessage = messageHandler;
+        this.state = "initialized";
+    }
+    async initDevices() {
+        // Spin up the Microphone with permission requested so labels are
+        // available for the picker before the user clicks Connect. Real
+        // capture constraints (incl. sample rate) get applied in `_connect`
+        // once we have the resolved options.
+        if (!this._microphone) this._microphone = new $3Y32T$Source.Microphone({
+            enabled: this._micEnabled,
+            constraints: this._micConstraints,
+            device: {
+                preferred: this._preferredMicId
+            }
+        });
+        this._microphone.device.requestPermission();
+    }
+    _validateConnectionParams(connectParams) {
+        if (connectParams === undefined || connectParams === null) return this._optionsAsInput();
+        if (typeof connectParams !== "object") throw new (0, $3Y32T$RTVIError)("MoqTransport connect params must be an object");
+        // The bot's `/start` response is `{ moq: {...} }`, not already-shaped
+        // `MoqTransportOptions` — unwrap and decode it here so apps can pass
+        // the raw response straight through with no transport-specific glue.
+        if ($549dd95e7ad3e81f$var$isMoqStartResponse(connectParams)) {
+            const { moq: moq } = connectParams;
+            if (!moq) throw new (0, $3Y32T$RTVIError)("MoqTransport connect params did not include `moq` config (server may not be running with `-t moq`)");
+            return {
+                ...this._optionsAsInput(),
+                relayUrl: moq.relayUrl,
+                clientId: moq.clientId,
+                botId: moq.botId,
+                namespace: moq.namespace,
+                transcriptTrack: moq.transcriptTrack,
+                serverCertificateHashes: moq.certHash ? [
+                    {
+                        algorithm: "sha-256",
+                        value: $549dd95e7ad3e81f$var$decodeCertHash(moq.certHash)
+                    }
+                ] : undefined
+            };
+        }
+        return {
+            ...this._optionsAsInput(),
+            ...connectParams
+        };
+    }
+    /** Return the resolved options re-shaped to the public input type
+     *  (used when merging with connect-time params). */ _optionsAsInput() {
+        return {
+            ...this._moqOptions
+        };
+    }
+    async _connect(connectParams) {
+        const merged = connectParams ? {
+            ...this._moqOptions,
+            ...$549dd95e7ad3e81f$var$applyDefaults({
+                ...this._moqOptions,
+                ...connectParams
+            })
+        } : this._moqOptions;
+        if (!merged.relayUrl) throw new (0, $3Y32T$RTVIError)("MoqTransport requires `relayUrl`");
+        this._moqOptions = merged;
+        this.state = "connecting";
+        let url;
+        try {
+            url = new URL(merged.relayUrl);
+        } catch (err) {
+            this.state = "error";
+            throw new (0, $3Y32T$RTVIError)(`MoqTransport invalid relayUrl ${JSON.stringify(merged.relayUrl)}: ${err instanceof Error ? err.message : String(err)}`);
+        }
+        // @moq/net narrows `algorithm` to the literal `"sha-256"`, while
+        // the browser's `WebTransportHash` types it as `string`. Cast at
+        // the boundary; SHA-256 is the only algorithm WebTransport accepts.
+        const webtransport = merged.serverCertificateHashes ? {
+            serverCertificateHashes: merged.serverCertificateHashes.flatMap((h)=>h.value ? [
+                    {
+                        algorithm: "sha-256",
+                        value: h.value
+                    }
+                ] : [])
+        } : undefined;
+        // Reload auto-reconnects on disconnect; Publish.Broadcast and
+        // Watch.Broadcast both react to its `established` signal.
+        this._reload = new $3Y32T$Connection.Reload({
+            enabled: new (0, $3Y32T$Signal)(true),
+            url: new (0, $3Y32T$Signal)(url),
+            webtransport: webtransport
+        });
+        // One reactive root for status mirroring. Connect/Watch are
+        // self-driving via their own internal effects.
+        this._signals = new (0, $3Y32T$Effect)();
+        this._signals.run((eff)=>{
+            const status = eff.get(this._reload.status);
+            if (status === "connected") {
+                if (this._state === "connecting") this.state = "connected";
+            } else if (status === "connecting") this.state = "connecting";
+            else if (status === "disconnected") {
+                if (this._state !== "disconnecting" && this._state !== "disconnected") this.state = "disconnected";
+            }
+        });
+        const ourPath = $3Y32T$Path.from(merged.namespace, merged.clientId);
+        const botPath = $3Y32T$Path.from(merged.namespace, merged.botId);
+        // ----------------------------------------------------------------
+        // Publish — Microphone owns getUserMedia, device selection, and
+        // produces a reactive `source` signal that Publish.Broadcast
+        // consumes. Constraints (channelCount:1, sampleRate) and preferred
+        // deviceId flow through the signals we hold a reference to, so
+        // updateMic() re-routes audio without re-creating the broadcast.
+        // The encoder's sampleRate Signal pins what the catalog advertises.
+        // ----------------------------------------------------------------
+        this._micConstraints.set({
+            channelCount: {
+                exact: 1
+            },
+            sampleRate: {
+                ideal: merged.audioSampleRate
+            },
+            echoCancellation: true,
+            noiseSuppression: true
+        });
+        this._audioSampleRate.set(merged.audioSampleRate);
+        if (!this._microphone) this._microphone = new $3Y32T$Source.Microphone({
+            enabled: this._micEnabled,
+            constraints: this._micConstraints,
+            device: {
+                preferred: this._preferredMicId
+            }
+        });
+        this._publishBroadcast = new $3Y32T$Broadcast({
+            connection: this._reload.established,
+            enabled: new (0, $3Y32T$Signal)(true),
+            name: new (0, $3Y32T$Signal)(ourPath),
+            audio: {
+                source: this._microphone.source,
+                enabled: this._micEnabled,
+                sampleRate: this._audioSampleRate
+            }
+        });
+        // Client-side transcript: `sendMessage` appends each RTVI message to
+        // a lossless JSON append-stream. `publishTrack` hands us a fresh track
+        // per subscription (only one, in the normal single-bot flow), so we
+        // spin up a `Json.Stream.Producer` per subscriber and replay the
+        // message log into it — a bot that subscribes after we've already sent
+        // messages still gets the full log, in order. The bot subscribes to
+        // this track by its name — same convention as the bot's own transcript
+        // track — so no catalog entry is needed.
+        this._transcriptLog = [];
+        this._transcriptOut = new Set();
+        this._publishBroadcast.publishTrack(merged.transcriptTrack, (track, effect)=>{
+            const producer = new $3Y32T$Stream.Producer(track, {
+                compression: true
+            });
+            for (const msg of this._transcriptLog ?? [])producer.append(msg);
+            this._transcriptOut?.add(producer);
+            effect.cleanup(()=>{
+                this._transcriptOut?.delete(producer);
+                producer.finish();
+            });
+        });
+        // Log the mic settings the browser actually granted, so we can see
+        // when a UA ignores the constraint (e.g. macOS often pins 48k
+        // regardless of `sampleRate.ideal`).
+        this._signals.run((eff)=>{
+            const src = eff.get(this._microphone.source);
+            if (!src) return;
+            const track = "track" in src ? src.track : src;
+            const s = track.getSettings();
+            console.log(`[MoqTransport] publish: requested=${merged.audioSampleRate}Hz, ` + `mic granted=${s.sampleRate}Hz, channels=${s.channelCount}, ` + `deviceId=${s.deviceId}`);
+        });
+        // ----------------------------------------------------------------
+        // Watch — Broadcast handles catalog discovery and rendition
+        // tracking; Audio.Source picks the active audio rendition;
+        // Audio.Decoder runs the WebCodecs decode loop and feeds an
+        // AudioWorklet ring buffer; Audio.Emitter routes that to the
+        // speakers. `Sync.latencyMax` lets faster-than-real-time TTS build
+        // up a buffer instead of the player skipping ahead; `reset()`
+        // (invoked on `user-started-speaking`) flushes it on interruption.
+        // We also tap Decoder.root → MediaStreamAudioDestinationNode so
+        // tracks().bot.audio returns a MediaStreamTrack.
+        //
+        // `catalogFormat: "hang"` is pinned because the pipecat bot publishes
+        // a hang-format catalog (camelCase `sampleRate`). The auto-detector
+        // would also land on "hang" here (no suffix on the broadcast name,
+        // and hang is the DEFAULT_FORMAT) but pinning removes ambiguity if
+        // a future publisher adds a suffix or a different default ships.
+        // ----------------------------------------------------------------
+        this._watchBroadcast = new $3Y32T$Broadcast1({
+            connection: this._reload.established,
+            enabled: new (0, $3Y32T$Signal)(true),
+            name: new (0, $3Y32T$Signal)(botPath),
+            catalogFormat: new (0, $3Y32T$Signal)("hang")
+        });
+        // Latency range: floor = interactive jitter buffer (audioLatencyMs),
+        // ceiling = how much faster-than-real-time TTS the player will hold
+        // before dropping (audioBufferMaxMs). A number-typed max opens the
+        // buffer; "real-time" collapses to the floor (skip-ahead behavior).
+        const sync = new $3Y32T$Sync({
+            connection: this._reload.established,
+            latency: new (0, $3Y32T$Signal)({
+                min: merged.audioLatencyMs,
+                max: merged.audioBufferMaxMs === "real-time" ? "real-time" : merged.audioBufferMaxMs
+            })
+        });
+        this._sync = sync;
+        this._audioSource = new $3Y32T$Audio.Source(sync, {
+            broadcast: this._watchBroadcast
+        });
+        this._audioDecoder = new $3Y32T$Audio.Decoder(this._audioSource, {
+            enabled: new (0, $3Y32T$Signal)(true)
+        });
+        this._audioEmitter = new $3Y32T$Audio.Emitter(this._audioDecoder);
+        // Bridge Decoder.root (AudioNode) → MediaStreamTrack for tracks().bot.audio.
+        this._signals.run((eff)=>{
+            const ctx = eff.get(this._audioDecoder.context);
+            const root = eff.get(this._audioDecoder.root);
+            if (!ctx || !root) return;
+            const dest = ctx.createMediaStreamDestination();
+            root.connect(dest);
+            this._botAudioTrack = dest.stream.getAudioTracks()[0];
+            eff.cleanup(()=>{
+                try {
+                    root.disconnect(dest);
+                } catch  {
+                // best-effort.
+                }
+                this._botAudioTrack?.stop();
+                this._botAudioTrack = undefined;
+            });
+        });
+        // Log the bot's announced audio config + the AudioContext rate we
+        // ended up at. If these disagree the voice will play at the wrong
+        // pitch — the catalog drives the AudioContext, so a mismatch means
+        // either a parser bug or the bot is advertising a rate that doesn't
+        // match its actual Opus stream.
+        this._signals.run((eff)=>{
+            const config = eff.get(this._audioSource.config);
+            const ctx = eff.get(this._audioDecoder.context);
+            if (!config && !ctx) return;
+            console.log(`[MoqTransport] consume: catalog codec=${config?.codec}, ` + `catalog sampleRate=${config?.sampleRate}Hz, ` + `catalog channels=${config?.numberOfChannels}, ` + `AudioContext rate=${ctx?.sampleRate}Hz`);
+        });
+        // Transcript — a lossless JSON append-stream over a single track. We
+        // re-subscribe on each (re)connect; @moq/json's stream Consumer yields
+        // every appended record in order, losslessly.
+        this._signals.run((eff)=>{
+            const conn = eff.get(this._reload.established);
+            if (!conn) return;
+            const botBroadcast = conn.consume(botPath);
+            const track = botBroadcast.subscribe(merged.transcriptTrack, 0);
+            const consumer = new $3Y32T$Stream.Consumer(track, {
+                compression: true
+            });
+            const ac = new AbortController();
+            this._drainTranscript(consumer, ac.signal).catch((e)=>{
+                if (!ac.signal.aborted) console.warn("MoqTransport bot-transcript loop:", e);
+            });
+            eff.cleanup(()=>{
+                ac.abort();
+                try {
+                    track.close();
+                } catch  {
+                // best-effort.
+                }
+            });
+        });
+    }
+    async _disconnect() {
+        if (this._state === "disconnected") return;
+        this.state = "disconnecting";
+        try {
+            this._audioEmitter?.close();
+            this._audioDecoder?.close();
+            this._audioSource?.close();
+            this._sync?.close();
+            this._watchBroadcast?.close();
+            // Each producer's own effect.cleanup (registered in the `publishTrack`
+            // callback above) already calls producer.finish() when this cascades
+            // through the broadcast's internal effect scope — no need to also
+            // finish them explicitly here.
+            this._publishBroadcast?.close();
+            this._microphone?.close();
+            this._signals?.close();
+            this._reload?.close();
+        } finally{
+            this._audioEmitter = null;
+            this._audioDecoder = null;
+            this._audioSource = null;
+            this._sync = null;
+            this._watchBroadcast = null;
+            this._transcriptOut = null;
+            this._transcriptLog = null;
+            this._publishBroadcast = null;
+            this._microphone = null;
+            this._signals = null;
+            this._reload = null;
+            this.state = "disconnected";
+            // Fire the SDK's disconnected callback. `state = "disconnected"`
+            // above already triggers `onTransportStateChanged`, but the SDK
+            // wires UI teardown (voice-ui-kit's connection-state hook, the
+            // `Disconnected` event, message-dispatcher cleanup) through
+            // `onDisconnected` — without this call the transport goes idle
+            // but the UI stays stuck in "connected" and never restores its
+            // "Connect" button. Matches the pattern used by
+            // small-webrtc-transport, openai-realtime-webrtc-transport, and
+            // gemini-live-websocket-transport.
+            this._callbacks?.onDisconnected?.();
+        }
+    }
+    sendReadyMessage() {
+        // Flip transport state to `ready` and dispatch the RTVI `client-ready`
+        // message on the client→bot transcript track so the bot's RTVIProcessor
+        // can negotiate protocol version. Mirrors the pattern used by
+        // small-webrtc-transport and daily-transport.
+        this.state = "ready";
+        this.sendMessage((0, $3Y32T$RTVIMessage).clientReady());
+    }
+    // --------------------------------------------------------------------
+    // State
+    // --------------------------------------------------------------------
+    get state() {
+        return this._state;
+    }
+    set state(next) {
+        if (this._state === next) return;
+        this._state = next;
+        this._callbacks?.onTransportStateChanged?.(next);
+    }
+    // --------------------------------------------------------------------
+    // Devices
+    // --------------------------------------------------------------------
+    async getAllMics() {
+        return this._microphone?.device.available.peek() ?? [];
+    }
+    async getAllCams() {
+        return [];
+    }
+    async getAllSpeakers() {
+        return [];
+    }
+    updateMic(micId) {
+        this._preferredMicId.set(micId);
+    }
+    updateCam(_camId) {}
+    updateSpeaker(_speakerId) {}
+    get selectedMic() {
+        const id = this._microphone?.device.active.peek();
+        if (!id) return {};
+        return this._microphone?.device.available.peek()?.find((d)=>d.deviceId === id) ?? {};
+    }
+    get selectedCam() {
+        return {};
+    }
+    get selectedSpeaker() {
+        return {};
+    }
+    enableMic(enable) {
+        this._micEnabled.set(enable);
+    }
+    enableCam(_enable) {}
+    enableScreenShare(_enable) {}
+    get isCamEnabled() {
+        return false;
+    }
+    get isMicEnabled() {
+        return this._micEnabled.get();
+    }
+    get isSharingScreen() {
+        return false;
+    }
+    // --------------------------------------------------------------------
+    // Messaging + tracks
+    // --------------------------------------------------------------------
+    sendMessage(message) {
+        if (!this._transcriptOut || !this._transcriptLog) {
+            console.warn("[MoqTransport] sendMessage called before connect; dropping", message);
+            return;
+        }
+        // Append to the lossless stream. Keep a log too, so a bot that
+        // subscribes after this point is replayed every message in order
+        // (see the `publishTrack` serve callback above).
+        this._transcriptLog.push(message);
+        for (const producer of this._transcriptOut)producer.append(message);
+    }
+    tracks() {
+        const localSource = this._microphone?.source.peek();
+        const localAudio = localSource && "track" in localSource ? localSource.track : localSource;
+        return {
+            local: localAudio ? {
+                audio: localAudio
+            } : {},
+            bot: this._botAudioTrack ? {
+                audio: this._botAudioTrack
+            } : {}
+        };
+    }
+    // --------------------------------------------------------------------
+    // Internals — transcript
+    // --------------------------------------------------------------------
+    /** Pull RTVI messages off the @moq/json stream consumer and hand each
+     *  one to `PipecatClient` via `_onMessage`. The handler resolves the
+     *  SDK's connect promise when bot-ready arrives. */ async _drainTranscript(consumer, signal) {
+        for(;;){
+            const message = await consumer.next();
+            if (!message || signal.aborted) break;
+            if (typeof message !== "object") continue;
+            // Intra-transport shutdown signal from the bot: bot is about to
+            // close its MoQ session. Disable auto-reconnect, tear down the
+            // audio decoder + subscribers on our side while WebTransport is
+            // still healthy — without this the connection just vanishes
+            // underneath Chrome's WebCodecs decoder and can crash the
+            // renderer. Not an RTVI event; don't forward it to `_onMessage`.
+            const label = message.label;
+            const type = message.type;
+            if (label === "moq-transport" && type === "session-ending") {
+                console.log("[MoqTransport] bot signaled session-ending \u2014 closing cleanly");
+                // Fire-and-forget; we're returning from the drain loop right
+                // after, and _disconnect() aborts the consumer that owns us.
+                this._disconnect();
+                return;
+            }
+            // Interruption: the user started talking, so flush the
+            // already-buffered TTS for the bot's previous utterance instead
+            // of letting it drain. With the bot writing faster than
+            // real-time, that buffer can be seconds long.
+            if (type === "user-started-speaking") this._resetBotAudio();
+            this._onMessage?.(message);
+        }
+    }
+    /** Flush buffered bot audio and re-anchor playback at an utterance
+     *  boundary. Called on interruption (`user-started-speaking`) so the
+     *  already-buffered TTS for the previous utterance stops immediately
+     *  instead of draining: re-anchor the sync reference and flush the
+     *  decoder's ring buffer. */ _resetBotAudio() {
+        this._sync?.reset();
+        this._audioDecoder?.reset();
+    }
+}
+$549dd95e7ad3e81f$export$739cde4b077543d1.SERVICE_NAME = "moq-transport";
+
+
+
+
+export {$549dd95e7ad3e81f$export$739cde4b077543d1 as MoqTransport};
+//# sourceMappingURL=index.module.js.map