npm - @pipecat-ai/moq-transport - Versions diffs - 0.1.0 - Mend

@pipecat-ai/moq-transport 0.1.0

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package/dist/index.module.js.map ADDED Viewed

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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.module.js.map"}

package/package.json ADDED Viewed

@@ -0,0 +1,46 @@
+{
+  "name": "@pipecat-ai/moq-transport",
+  "version": "0.1.0",
+  "license": "BSD-2-Clause",
+  "main": "dist/index.js",
+  "module": "dist/index.module.js",
+  "types": "dist/index.d.ts",
+  "source": "src/index.ts",
+  "repository": {
+    "type": "git",
+    "url": "git+https://github.com/pipecat-ai/pipecat-client-web-transports.git"
+  },
+  "files": [
+    "dist",
+    "package.json",
+    "README.md"
+  ],
+  "scripts": {
+    "build": "parcel build --no-cache",
+    "dev": "parcel watch",
+    "lint": "eslint . --ext ts --report-unused-disable-directives --max-warnings 0"
+  },
+  "devDependencies": {
+    "@pipecat-ai/client-js": "^1.13.0",
+    "@types/node": "^22.9.0",
+    "eslint": "9.39.1",
+    "eslint-config-prettier": "^9.1.0",
+    "eslint-plugin-simple-import-sort": "^12.1.1"
+  },
+  "peerDependencies": {
+    "@pipecat-ai/client-js": "~1.13.0"
+  },
+  "dependencies": {
+    "@moq/hang": "^0.2.12",
+    "@moq/json": "^0.1.2",
+    "@moq/net": "^0.1.6",
+    "@moq/publish": "^0.2.16",
+    "@moq/signals": "^0.1.10",
+    "@moq/watch": "^0.2.18"
+  },
+  "description": "Pipecat Media-over-QUIC (MoQ) Transport Package",
+  "author": "Daily.co",
+  "bugs": {
+    "url": "https://github.com/pipecat-ai/pipecat-client-web-transports/issues"
+  }
+}