gemi 0.132.0 → 0.134.0

package/dist/ai/voice/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","names":[],"sources":["../../../ai/voice/state.ts","../../../ai/voice/voiceCall.ts","../../../ai/voice/useVoiceSession.ts"],"sourcesContent":["import type { VoiceCaption, VoiceStatus } from \"./wire\";\n\n/**\n * What `useVoiceSession` shows, as a pure reducer over the call's data-channel\n * events (#921). Separate from the hook so it is tested without a DOM, a\n * microphone or a peer connection.\n *\n * The events are the vendor's, as the data channel delivers them. Which ones\n * arrive there — in particular the `function_call_output` items the server's\n * sideband adds, which end `working` — follows the documented protocol and has\n * not been measured against Azure.\n */\nexport type VoiceState = {\n  phase: \"idle\" | \"connecting\" | \"live\" | \"ended\";\n  /** The user is speaking (VAD) or holding push-to-talk. */\n  userSpeaking: boolean;\n  /** The user stopped and no answer has started playing. */\n  thinking: boolean;\n  /** The voice's audio is playing. */\n  speaking: boolean;\n  /** Delegate calls with no output yet. */\n  working: string[];\n  captions: VoiceCaption[];\n};\n\nexport const MAX_CAPTIONS = 20;\n\nexport function initialVoiceState(): VoiceState {\n  return {\n    phase: \"idle\",\n    userSpeaking: false,\n    thinking: false,\n    speaking: false,\n    working: [],\n    captions: [],\n  };\n}\n\nexport function voiceStatus(state: VoiceState): VoiceStatus {\n  if (state.phase === \"idle\") return \"idle\";\n  if (state.phase === \"connecting\") return \"connecting\";\n  if (state.phase === \"ended\") return \"ended\";\n  if (state.speaking) return \"speaking\";\n  if (state.userSpeaking) return \"listening\";\n  if (state.thinking) return \"thinking\";\n  if (state.working.length) return \"working\";\n  return \"listening\";\n}\n\ntype Event = { type: string; [key: string]: any };\n\n/** One data-channel event onto the state. `delegate` is the delegate tool's name. */\nexport function applyVoiceEvent(state: VoiceState, event: Event, delegate: string): VoiceState {\n  switch (event.type) {\n    case \"input_audio_buffer.speech_started\":\n      return { ...state, userSpeaking: true, thinking: false };\n    case \"input_audio_buffer.speech_stopped\":\n    case \"input_audio_buffer.committed\":\n      return { ...state, userSpeaking: false, thinking: true };\n    case \"output_audio_buffer.started\":\n      return { ...state, speaking: true, thinking: false };\n    case \"output_audio_buffer.stopped\":\n    case \"output_audio_buffer.cleared\":\n      return { ...state, speaking: false };\n    case \"response.done\": {\n      let next = { ...state, thinking: false };\n      for (const item of event.response?.output ?? []) {\n        if (item?.type === \"function_call\" && item.name === delegate && item.call_id) {\n          next = addWorking(next, item.call_id);\n        }\n      }\n      return next;\n    }\n    case \"response.function_call_arguments.done\":\n      return event.name === delegate && event.call_id ? addWorking(state, event.call_id) : state;\n    case \"conversation.item.added\":\n    case \"conversation.item.created\":\n    case \"conversation.item.done\": {\n      const item = event.item;\n      if (item?.type === \"function_call_output\" && state.working.includes(item.call_id)) {\n        return { ...state, working: state.working.filter((id) => id !== item.call_id) };\n      }\n      return state;\n    }\n    case \"conversation.item.input_audio_transcription.delta\":\n      return caption(state, event.item_id, \"user\", event.delta ?? \"\", false);\n    case \"conversation.item.input_audio_transcription.completed\":\n      return caption(state, event.item_id, \"user\", event.transcript ?? \"\", true);\n    case \"response.output_audio_transcript.delta\":\n    case \"response.audio_transcript.delta\":\n      return caption(state, event.item_id, \"assistant\", event.delta ?? \"\", false);\n    case \"response.output_audio_transcript.done\":\n    case \"response.audio_transcript.done\":\n      return caption(state, event.item_id, \"assistant\", event.transcript ?? \"\", true);\n    default:\n      return state;\n  }\n}\n\nfunction addWorking(state: VoiceState, callId: string): VoiceState {\n  return state.working.includes(callId) ? state : { ...state, working: [...state.working, callId] };\n}\n\n/** A delta appends to its line; a final text replaces it. */\nfunction caption(\n  state: VoiceState,\n  id: string | undefined,\n  role: VoiceCaption[\"role\"],\n  text: string,\n  final: boolean,\n): VoiceState {\n  if (!id) return state;\n  const index = state.captions.findIndex((line) => line.id === id);\n  const captions = [...state.captions];\n  if (index === -1) {\n    if (!text) return state;\n    captions.push({ id, role, text, final });\n  } else {\n    const line = captions[index]!;\n    captions[index] = { ...line, text: final ? text || line.text : line.text + text, final };\n  }\n  return { ...state, captions: captions.slice(-MAX_CAPTIONS) };\n}\n","import type { VoiceStartResponse } from \"./wire\";\n\n/**\n * The browser's half of a voice call (#921): the microphone, the peer\n * connection, the vendor's data channel and the app's three routes.\n *\n * Only `gemi/ai/voice` reaches it, so it is in an app's bundle only where the\n * app uses that entry. The packaging test checks it stays out of the entry\n * chunk.\n *\n * No vendor SDK: a raw `RTCPeerConnection`, with the audio going straight to\n * the vendor and the SDP offer going to the app's route, which creates the\n * call with the server's key. The browser never holds a key or a secret.\n */\n\nexport type VoiceCallOptions = {\n  /** `/api/<route>`. */\n  url: string;\n  threadId?: string;\n  body?: unknown;\n  pushToTalk?: boolean;\n  headers?: Record<string, string>;\n  /** Every vendor event from the data channel. */\n  onEvent: (event: { type: string; [key: string]: unknown }) => void;\n  /** The connection closed or failed. */\n  onClose: (error?: Error) => void;\n  /** What push-to-talk turns detection back on to. Default semantic VAD. */\n  turnDetection?: Record<string, unknown>;\n};\n\nexport type VoiceCall = {\n  sessionId: string;\n  threadId: string;\n  delegate: string;\n  endsAt?: number;\n  mute(muted: boolean): void;\n  setPushToTalk(on: boolean): void;\n  /** Push-to-talk: start speaking. */\n  press(): void;\n  /** Push-to-talk: done speaking; the turn is committed and answered. */\n  release(): void;\n  setBody(body: unknown): Promise<void>;\n  /** Hangs up through the app's route and closes everything. */\n  end(): Promise<void>;\n  /** Closes everything without telling the server (it learns from the vendor). */\n  close(): void;\n};\n\nexport class VoiceCallError extends Error {\n  readonly status?: number;\n  readonly code?: string;\n\n  constructor(message: string, options: { status?: number; code?: string; cause?: unknown } = {}) {\n    super(message, { cause: options.cause });\n    this.name = \"VoiceCallError\";\n    this.status = options.status;\n    this.code = options.code;\n  }\n}\n\nexport async function startVoiceCall(options: VoiceCallOptions): Promise<VoiceCall> {\n  if (typeof RTCPeerConnection === \"undefined\" || !navigator.mediaDevices?.getUserMedia) {\n    throw new VoiceCallError(\"This browser can't make voice calls.\", { code: \"unsupported\" });\n  }\n\n  let stream: MediaStream;\n  try {\n    stream = await navigator.mediaDevices.getUserMedia({\n      audio: { echoCancellation: true, noiseSuppression: true, autoGainControl: true },\n    });\n  } catch (error) {\n    throw new VoiceCallError(\"The microphone is not available.\", {\n      code: \"microphone_denied\",\n      cause: error,\n    });\n  }\n  const track = stream.getAudioTracks()[0]!;\n  let pushToTalk = options.pushToTalk === true;\n  let muted = false;\n  track.enabled = !pushToTalk;\n\n  const pc = new RTCPeerConnection();\n  const audio = document.createElement(\"audio\");\n  audio.autoplay = true;\n  pc.ontrack = (event) => {\n    audio.srcObject = event.streams[0] ?? new MediaStream([event.track]);\n  };\n  pc.addTrack(track, stream);\n  const channel = pc.createDataChannel(\"oai-events\");\n\n  let closed = false;\n  const close = (error?: Error) => {\n    if (closed) return;\n    closed = true;\n    try {\n      channel.close();\n    } catch {\n      // already closed\n    }\n    pc.close();\n    for (const t of stream.getTracks()) t.stop();\n    audio.srcObject = null;\n    options.onClose(error);\n  };\n\n  channel.addEventListener(\"message\", (message) => {\n    let event: unknown;\n    try {\n      event = JSON.parse(String(message.data));\n    } catch {\n      return;\n    }\n    if (\n      event &&\n      typeof event === \"object\" &&\n      typeof (event as { type?: unknown }).type === \"string\"\n    ) {\n      options.onEvent(event as { type: string });\n    }\n  });\n  channel.addEventListener(\"close\", () => close());\n  pc.addEventListener(\"connectionstatechange\", () => {\n    if (pc.connectionState === \"failed\") close(new VoiceCallError(\"The voice connection failed.\"));\n    if (pc.connectionState === \"closed\") close();\n  });\n\n  // Events sent before the data channel opens (push-to-talk pressed at once)\n  // wait for it rather than being dropped.\n  const outbox: string[] = [];\n  channel.addEventListener(\"open\", () => {\n    for (const data of outbox.splice(0)) channel.send(data);\n  });\n  const send = (event: Record<string, unknown>) => {\n    const data = JSON.stringify(event);\n    if (channel.readyState === \"open\") channel.send(data);\n    else if (channel.readyState === \"connecting\") outbox.push(data);\n  };\n  const request = (method: string, url: string, payload?: unknown, init: RequestInit = {}) =>\n    fetch(url, {\n      method,\n      headers: { \"Content-Type\": \"application/json\", ...options.headers },\n      ...(payload === undefined ? {} : { body: JSON.stringify(payload) }),\n      ...init,\n    });\n\n  let started: VoiceStartResponse;\n  try {\n    const offer = await pc.createOffer();\n    await pc.setLocalDescription(offer);\n    const response = await request(\"POST\", options.url, {\n      ...(options.body && typeof options.body === \"object\" ? options.body : {}),\n      sdp: offer.sdp,\n      ...(options.threadId ? { threadId: options.threadId } : {}),\n      ...(pushToTalk ? { pushToTalk: true } : {}),\n    });\n    if (!response.ok) {\n      const error = await response.json().catch(() => null);\n      throw new VoiceCallError(\n        error?.error?.message ?? `The voice call could not be started (${response.status}).`,\n        { status: response.status, code: error?.error?.code },\n      );\n    }\n    started = await response.json();\n    await pc.setRemoteDescription({ type: \"answer\", sdp: started.sdp });\n  } catch (error) {\n    close();\n    throw error instanceof VoiceCallError\n      ? error\n      : new VoiceCallError(\"The voice call could not be started.\", { cause: error });\n  }\n\n  const sessionUrl = `${options.url}/${encodeURIComponent(started.sessionId)}`;\n  const applyTrack = () => {\n    track.enabled = !muted && !pushToTalk;\n  };\n\n  return {\n    sessionId: started.sessionId,\n    threadId: started.threadId,\n    delegate: started.delegate,\n    endsAt: started.endsAt,\n    mute(on) {\n      muted = on;\n      applyTrack();\n    },\n    setPushToTalk(on) {\n      pushToTalk = on;\n      applyTrack();\n      send({\n        type: \"session.update\",\n        session: {\n          type: \"realtime\",\n          audio: {\n            input: {\n              turn_detection: on ? null : (options.turnDetection ?? { type: \"semantic_vad\" }),\n            },\n          },\n        },\n      });\n    },\n    press() {\n      if (!pushToTalk || muted) return;\n      send({ type: \"input_audio_buffer.clear\" });\n      track.enabled = true;\n    },\n    release() {\n      if (!pushToTalk) return;\n      track.enabled = false;\n      send({ type: \"input_audio_buffer.commit\" });\n      send({ type: \"response.create\" });\n    },\n    async setBody(body) {\n      const response = await request(\"PATCH\", sessionUrl, body ?? {});\n      if (!response.ok && response.status !== 409) {\n        throw new VoiceCallError(`The voice session could not be updated (${response.status}).`, {\n          status: response.status,\n        });\n      }\n    },\n    async end() {\n      try {\n        // `keepalive`, so a hang-up sent as the tab closes still reaches the\n        // route: minutes are billed until it does.\n        await request(\"DELETE\", sessionUrl, undefined, { keepalive: true });\n      } finally {\n        close();\n      }\n    },\n    close: () => close(),\n  };\n}\n","import { useCallback, useEffect, useMemo, useRef, useState } from \"react\";\nimport type { RPC } from \"../../client/rpc\";\nimport { applyVoiceEvent, initialVoiceState, voiceStatus, type VoiceState } from \"./state\";\nimport { startVoiceCall, type VoiceCall } from \"./voiceCall\";\nimport type { VoiceCaption, VoiceStatus } from \"./wire\";\n\n/**\n * The voice routes, picked out of `RPC` the way `useChat` picks its agent\n * routes: `this.voice(Controller)` mounted at `\"/chat/voice\"` is the key\n * `\"/chat/voice\"`, carrying the controller's `Body`.\n */\ntype VoiceRoutes = {\n  [K in keyof RPC as RPC[K] extends { __voice: true } ? K : never]: RPC[K];\n};\n\ntype BodyOf<P extends keyof VoiceRoutes> = VoiceRoutes[P] extends { body: infer B }\n  ? B\n  : Record<string, unknown>;\n\nexport type UseVoiceSessionParams<P extends keyof VoiceRoutes> = {\n  /** The chat thread delegated turns are stored on, as `useChat` names it. */\n  threadId?: string;\n  /** The controller's `Body`, sent when the call starts. Change it during the\n   *  call with `setBody`. */\n  body?: BodyOf<P>;\n  /** Start with push-to-talk instead of voice activity detection. */\n  pushToTalk?: boolean;\n  /** Values for the route path's `:params`, e.g. `{ siteId }` for\n   *  `\"/sites/:siteId/voice\"`. */\n  params?: Record<string, string>;\n  headers?: Record<string, string>;\n  /** The call could not start, or failed. */\n  onError?: (error: Error) => void;\n  /** The call is over, however it ended. */\n  onEnd?: () => void;\n};\n\nexport type UseVoiceSessionResult<P extends keyof VoiceRoutes> = {\n  status: VoiceStatus;\n  /** The last lines said, the user's and the voice's, oldest first. */\n  captions: VoiceCaption[];\n  /** Why the call could not start or ended early. `code` is\n   *  `microphone_denied`, `unsupported`, or the route's error code. */\n  error: (Error & { code?: string; status?: number }) | null;\n  muted: boolean;\n  pushToTalk: boolean;\n  /** When the server will end the call, in epoch ms. */\n  endsAt?: number;\n  sessionId?: string;\n  /** Asks for the microphone and starts the call. Call it from a user gesture\n   *  (a click): browsers allow the microphone and audio playback only then. */\n  start(): Promise<void>;\n  /** Hangs up. */\n  end(): Promise<void>;\n  mute(muted?: boolean): void;\n  setPushToTalk(on: boolean): void;\n  /** Push-to-talk: hold to speak (pointer down / key down). */\n  press(): void;\n  /** Push-to-talk: release to send (pointer up / key up). */\n  release(): void;\n  /** Sends a new `body` (the selection, the open page) for the next\n   *  delegated request. */\n  setBody(body: BodyOf<P>): Promise<void>;\n};\n\n/**\n * A voice call to an agent, for a route mounted with `this.voice(Controller)`\n * (#921).\n *\n * ```tsx\n * const voice = useVoiceSession(\"/page-builder/voice\", { threadId, body: { pageId } });\n * <button onClick={() => (voice.status === \"idle\" ? voice.start() : voice.end())}>\n *   {voice.status}\n * </button>\n * ```\n *\n * Import it from `gemi/ai/voice`, an entry of its own, so an app that never\n * uses voice never bundles it, and use it from a view or a dynamically imported\n * component, so its code is not in the entry chunk.\n *\n * @experimental Not yet verified against a live call.\n */\nexport function useVoiceSession<P extends keyof VoiceRoutes>(\n  path: P,\n  params: UseVoiceSessionParams<P> = {},\n): UseVoiceSessionResult<P> {\n  // Not `useParams()`, which `useChat` uses: it is in a chunk `gemi/client`\n  // shares, and reaching it from here made the app's bundler move it out of\n  // the entry chunk into one of its own that every route then imports. A path\n  // with parameters names them in `params`.\n  const routeParams = params.params;\n  const url = useMemo(() => {\n    let out = String(path).replace(/^[A-Z]+:/, \"\");\n    for (const [key, value] of Object.entries(routeParams ?? {})) {\n      out = out.replace(`:${key}?`, value).replace(`:${key}`, value);\n    }\n    return `/api${out}`;\n  }, [path, routeParams && JSON.stringify(routeParams)]);\n\n  const [state, setState] = useState<VoiceState>(initialVoiceState);\n  const [error, setError] = useState<UseVoiceSessionResult<P>[\"error\"]>(null);\n  const [muted, setMuted] = useState(false);\n  const [pushToTalk, setPushToTalkState] = useState(params.pushToTalk === true);\n  const [call, setCall] = useState<VoiceCall | null>(null);\n  const callRef = useRef<VoiceCall | null>(null);\n  /** A `start()` in flight: a second click must not start a second call. */\n  const startingRef = useRef(false);\n  const latest = useRef(params);\n  latest.current = params;\n  const delegateRef = useRef(\"ask_agent\");\n  const mounted = useRef(true);\n\n  useEffect(() => {\n    mounted.current = true;\n    return () => {\n      mounted.current = false;\n      // Leaving the page ends the call: minutes are billed while it is up.\n      void callRef.current?.end().catch(() => undefined);\n      callRef.current = null;\n    };\n  }, []);\n\n  const start = useCallback(async () => {\n    if (callRef.current || startingRef.current) return;\n    startingRef.current = true;\n    setError(null);\n    setState({ ...initialVoiceState(), phase: \"connecting\" });\n    try {\n      const { threadId, body, headers } = latest.current;\n      const started = await startVoiceCall({\n        url,\n        threadId,\n        body,\n        headers,\n        pushToTalk,\n        onEvent: (event) => {\n          if (!mounted.current) return;\n          setState((current) =>\n            current.phase === \"connecting\"\n              ? applyVoiceEvent({ ...current, phase: \"live\" }, event, delegateRef.current)\n              : applyVoiceEvent(current, event, delegateRef.current),\n          );\n        },\n        onClose: (closeError) => {\n          callRef.current = null;\n          if (!mounted.current) return;\n          setCall(null);\n          setState((current) => ({ ...current, phase: \"ended\", speaking: false, thinking: false }));\n          if (closeError) {\n            setError(closeError);\n            latest.current.onError?.(closeError);\n          }\n          latest.current.onEnd?.();\n        },\n      });\n      delegateRef.current = started.delegate;\n      if (!mounted.current) {\n        await started.end().catch(() => undefined);\n        return;\n      }\n      callRef.current = started;\n      setCall(started);\n      setState((current) =>\n        current.phase === \"connecting\" ? { ...current, phase: \"live\" } : current,\n      );\n    } catch (failure) {\n      const reported = failure instanceof Error ? failure : new Error(String(failure));\n      if (!mounted.current) return;\n      setState((current) => ({ ...current, phase: \"ended\" }));\n      setError(reported);\n      latest.current.onError?.(reported);\n    } finally {\n      startingRef.current = false;\n    }\n  }, [url, pushToTalk]);\n\n  const end = useCallback(async () => {\n    const current = callRef.current;\n    callRef.current = null;\n    await current?.end();\n  }, []);\n\n  const mute = useCallback((on?: boolean) => {\n    setMuted((was) => {\n      const next = on ?? !was;\n      callRef.current?.mute(next);\n      return next;\n    });\n  }, []);\n\n  const setPushToTalk = useCallback((on: boolean) => {\n    setPushToTalkState(on);\n    callRef.current?.setPushToTalk(on);\n  }, []);\n\n  const press = useCallback(() => {\n    callRef.current?.press();\n    setState((current) => ({ ...current, userSpeaking: true }));\n  }, []);\n\n  const release = useCallback(() => {\n    callRef.current?.release();\n    setState((current) => ({ ...current, userSpeaking: false, thinking: true }));\n  }, []);\n\n  const setBody = useCallback(async (body: BodyOf<P>) => {\n    latest.current = { ...latest.current, body };\n    await callRef.current?.setBody(body);\n  }, []);\n\n  return {\n    status: voiceStatus(state),\n    captions: state.captions,\n    error,\n    muted,\n    pushToTalk,\n    endsAt: call?.endsAt,\n    sessionId: call?.sessionId,\n    start,\n    end,\n    mute,\n    setPushToTalk,\n    press,\n    release,\n    setBody,\n  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+{"version":3,"file":"index.js","names":[],"sources":["../../../ai/voice/state.ts","../../../ai/voice/voiceCall.ts","../../../ai/voice/useVoiceSession.ts","../../../ai/voice/energyVad.ts","../../../ai/voice/liveTranscription.ts","../../../ai/voice/transcriptionState.ts","../../../ai/voice/useLiveTranscription.ts"],"sourcesContent":["import type { VoiceCaption, VoiceStatus } from \"./wire\";\n\n/**\n * What `useVoiceSession` shows, as a pure reducer over the call's data-channel\n * events (#921). Separate from the hook so it is tested without a DOM, a\n * microphone or a peer connection.\n *\n * The events are the vendor's, as the data channel delivers them. Which ones\n * arrive there — in particular the `function_call_output` items the server's\n * sideband adds, which end `working` — follows the documented protocol and has\n * not been measured against Azure.\n */\nexport type VoiceState = {\n  phase: \"idle\" | \"connecting\" | \"live\" | \"ended\";\n  /** The user is speaking (VAD) or holding push-to-talk. */\n  userSpeaking: boolean;\n  /** The user stopped and no answer has started playing. */\n  thinking: boolean;\n  /** The voice's audio is playing. */\n  speaking: boolean;\n  /** Delegate calls with no output yet. */\n  working: string[];\n  captions: VoiceCaption[];\n};\n\nexport const MAX_CAPTIONS = 20;\n\nexport function initialVoiceState(): VoiceState {\n  return {\n    phase: \"idle\",\n    userSpeaking: false,\n    thinking: false,\n    speaking: false,\n    working: [],\n    captions: [],\n  };\n}\n\nexport function voiceStatus(state: VoiceState): VoiceStatus {\n  if (state.phase === \"idle\") return \"idle\";\n  if (state.phase === \"connecting\") return \"connecting\";\n  if (state.phase === \"ended\") return \"ended\";\n  if (state.speaking) return \"speaking\";\n  if (state.userSpeaking) return \"listening\";\n  if (state.thinking) return \"thinking\";\n  if (state.working.length) return \"working\";\n  return \"listening\";\n}\n\ntype Event = { type: string; [key: string]: any };\n\n/** One data-channel event onto the state. `delegate` is the delegate tool's name. */\nexport function applyVoiceEvent(state: VoiceState, event: Event, delegate: string): VoiceState {\n  switch (event.type) {\n    case \"input_audio_buffer.speech_started\":\n      return { ...state, userSpeaking: true, thinking: false };\n    case \"input_audio_buffer.speech_stopped\":\n    case \"input_audio_buffer.committed\":\n      return { ...state, userSpeaking: false, thinking: true };\n    case \"output_audio_buffer.started\":\n      return { ...state, speaking: true, thinking: false };\n    case \"output_audio_buffer.stopped\":\n    case \"output_audio_buffer.cleared\":\n      return { ...state, speaking: false };\n    case \"response.done\": {\n      let next = { ...state, thinking: false };\n      for (const item of event.response?.output ?? []) {\n        if (item?.type === \"function_call\" && item.name === delegate && item.call_id) {\n          next = addWorking(next, item.call_id);\n        }\n      }\n      return next;\n    }\n    case \"response.function_call_arguments.done\":\n      return event.name === delegate && event.call_id ? addWorking(state, event.call_id) : state;\n    case \"conversation.item.added\":\n    case \"conversation.item.created\":\n    case \"conversation.item.done\": {\n      const item = event.item;\n      if (item?.type === \"function_call_output\" && state.working.includes(item.call_id)) {\n        return { ...state, working: state.working.filter((id) => id !== item.call_id) };\n      }\n      return state;\n    }\n    case \"conversation.item.input_audio_transcription.delta\":\n      return caption(state, event.item_id, \"user\", event.delta ?? \"\", false);\n    case \"conversation.item.input_audio_transcription.completed\":\n      return caption(state, event.item_id, \"user\", event.transcript ?? \"\", true);\n    case \"response.output_audio_transcript.delta\":\n    case \"response.audio_transcript.delta\":\n      return caption(state, event.item_id, \"assistant\", event.delta ?? \"\", false);\n    case \"response.output_audio_transcript.done\":\n    case \"response.audio_transcript.done\":\n      return caption(state, event.item_id, \"assistant\", event.transcript ?? \"\", true);\n    default:\n      return state;\n  }\n}\n\nfunction addWorking(state: VoiceState, callId: string): VoiceState {\n  return state.working.includes(callId) ? state : { ...state, working: [...state.working, callId] };\n}\n\n/** A delta appends to its line; a final text replaces it. */\nfunction caption(\n  state: VoiceState,\n  id: string | undefined,\n  role: VoiceCaption[\"role\"],\n  text: string,\n  final: boolean,\n): VoiceState {\n  if (!id) return state;\n  const index = state.captions.findIndex((line) => line.id === id);\n  const captions = [...state.captions];\n  if (index === -1) {\n    if (!text) return state;\n    captions.push({ id, role, text, final });\n  } else {\n    const line = captions[index]!;\n    captions[index] = { ...line, text: final ? text || line.text : line.text + text, final };\n  }\n  return { ...state, captions: captions.slice(-MAX_CAPTIONS) };\n}\n","import type { VoiceStartResponse } from \"./wire\";\n\n/**\n * The browser's half of a voice call (#921): the microphone, the peer\n * connection, the vendor's data channel and the app's three routes.\n *\n * Only `gemi/ai/voice` reaches it, so it is in an app's bundle only where the\n * app uses that entry. The packaging test checks it stays out of the entry\n * chunk.\n *\n * No vendor SDK: a raw `RTCPeerConnection`, with the audio going straight to\n * the vendor and the SDP offer going to the app's route, which creates the\n * call with the server's key. The browser never holds a key or a secret.\n */\n\nexport type VoiceCallOptions = {\n  /** `/api/<route>`. */\n  url: string;\n  threadId?: string;\n  body?: unknown;\n  pushToTalk?: boolean;\n  headers?: Record<string, string>;\n  /** Every vendor event from the data channel. */\n  onEvent: (event: { type: string; [key: string]: unknown }) => void;\n  /** The connection closed or failed. */\n  onClose: (error?: Error) => void;\n  /** What push-to-talk turns detection back on to. Default semantic VAD. */\n  turnDetection?: Record<string, unknown>;\n};\n\nexport type VoiceCall = {\n  sessionId: string;\n  threadId: string;\n  delegate: string;\n  endsAt?: number;\n  mute(muted: boolean): void;\n  setPushToTalk(on: boolean): void;\n  /** Push-to-talk: start speaking. */\n  press(): void;\n  /** Push-to-talk: done speaking; the turn is committed and answered. */\n  release(): void;\n  setBody(body: unknown): Promise<void>;\n  /** Hangs up through the app's route and closes everything. */\n  end(): Promise<void>;\n  /** Closes everything without telling the server (it learns from the vendor). */\n  close(): void;\n};\n\nexport class VoiceCallError extends Error {\n  readonly status?: number;\n  readonly code?: string;\n\n  constructor(message: string, options: { status?: number; code?: string; cause?: unknown } = {}) {\n    super(message, { cause: options.cause });\n    this.name = \"VoiceCallError\";\n    this.status = options.status;\n    this.code = options.code;\n  }\n}\n\nexport async function startVoiceCall(options: VoiceCallOptions): Promise<VoiceCall> {\n  if (typeof RTCPeerConnection === \"undefined\" || !navigator.mediaDevices?.getUserMedia) {\n    throw new VoiceCallError(\"This browser can't make voice calls.\", { code: \"unsupported\" });\n  }\n\n  let stream: MediaStream;\n  try {\n    stream = await navigator.mediaDevices.getUserMedia({\n      audio: { echoCancellation: true, noiseSuppression: true, autoGainControl: true },\n    });\n  } catch (error) {\n    throw new VoiceCallError(\"The microphone is not available.\", {\n      code: \"microphone_denied\",\n      cause: error,\n    });\n  }\n  const track = stream.getAudioTracks()[0]!;\n  let pushToTalk = options.pushToTalk === true;\n  let muted = false;\n  track.enabled = !pushToTalk;\n\n  const pc = new RTCPeerConnection();\n  const audio = document.createElement(\"audio\");\n  audio.autoplay = true;\n  pc.ontrack = (event) => {\n    audio.srcObject = event.streams[0] ?? new MediaStream([event.track]);\n  };\n  pc.addTrack(track, stream);\n  const channel = pc.createDataChannel(\"oai-events\");\n\n  let closed = false;\n  const close = (error?: Error) => {\n    if (closed) return;\n    closed = true;\n    try {\n      channel.close();\n    } catch {\n      // already closed\n    }\n    pc.close();\n    for (const t of stream.getTracks()) t.stop();\n    audio.srcObject = null;\n    options.onClose(error);\n  };\n\n  channel.addEventListener(\"message\", (message) => {\n    let event: unknown;\n    try {\n      event = JSON.parse(String(message.data));\n    } catch {\n      return;\n    }\n    if (\n      event &&\n      typeof event === \"object\" &&\n      typeof (event as { type?: unknown }).type === \"string\"\n    ) {\n      options.onEvent(event as { type: string });\n    }\n  });\n  channel.addEventListener(\"close\", () => close());\n  pc.addEventListener(\"connectionstatechange\", () => {\n    if (pc.connectionState === \"failed\") close(new VoiceCallError(\"The voice connection failed.\"));\n    if (pc.connectionState === \"closed\") close();\n  });\n\n  // Events sent before the data channel opens (push-to-talk pressed at once)\n  // wait for it rather than being dropped.\n  const outbox: string[] = [];\n  channel.addEventListener(\"open\", () => {\n    for (const data of outbox.splice(0)) channel.send(data);\n  });\n  const send = (event: Record<string, unknown>) => {\n    const data = JSON.stringify(event);\n    if (channel.readyState === \"open\") channel.send(data);\n    else if (channel.readyState === \"connecting\") outbox.push(data);\n  };\n  const request = (method: string, url: string, payload?: unknown, init: RequestInit = {}) =>\n    fetch(url, {\n      method,\n      headers: { \"Content-Type\": \"application/json\", ...options.headers },\n      ...(payload === undefined ? {} : { body: JSON.stringify(payload) }),\n      ...init,\n    });\n\n  let started: VoiceStartResponse;\n  try {\n    const offer = await pc.createOffer();\n    await pc.setLocalDescription(offer);\n    const response = await request(\"POST\", options.url, {\n      ...(options.body && typeof options.body === \"object\" ? options.body : {}),\n      sdp: offer.sdp,\n      ...(options.threadId ? { threadId: options.threadId } : {}),\n      ...(pushToTalk ? { pushToTalk: true } : {}),\n    });\n    if (!response.ok) {\n      const error = await response.json().catch(() => null);\n      throw new VoiceCallError(\n        error?.error?.message ?? `The voice call could not be started (${response.status}).`,\n        { status: response.status, code: error?.error?.code },\n      );\n    }\n    started = await response.json();\n    await pc.setRemoteDescription({ type: \"answer\", sdp: started.sdp });\n  } catch (error) {\n    close();\n    throw error instanceof VoiceCallError\n      ? error\n      : new VoiceCallError(\"The voice call could not be started.\", { cause: error });\n  }\n\n  const sessionUrl = `${options.url}/${encodeURIComponent(started.sessionId)}`;\n  const applyTrack = () => {\n    track.enabled = !muted && !pushToTalk;\n  };\n\n  return {\n    sessionId: started.sessionId,\n    threadId: started.threadId,\n    delegate: started.delegate,\n    endsAt: started.endsAt,\n    mute(on) {\n      muted = on;\n      applyTrack();\n    },\n    setPushToTalk(on) {\n      pushToTalk = on;\n      applyTrack();\n      send({\n        type: \"session.update\",\n        session: {\n          type: \"realtime\",\n          audio: {\n            input: {\n              turn_detection: on ? null : (options.turnDetection ?? { type: \"semantic_vad\" }),\n            },\n          },\n        },\n      });\n    },\n    press() {\n      if (!pushToTalk || muted) return;\n      send({ type: \"input_audio_buffer.clear\" });\n      track.enabled = true;\n    },\n    release() {\n      if (!pushToTalk) return;\n      track.enabled = false;\n      send({ type: \"input_audio_buffer.commit\" });\n      send({ type: \"response.create\" });\n    },\n    async setBody(body) {\n      const response = await request(\"PATCH\", sessionUrl, body ?? {});\n      if (!response.ok && response.status !== 409) {\n        throw new VoiceCallError(`The voice session could not be updated (${response.status}).`, {\n          status: response.status,\n        });\n      }\n    },\n    async end() {\n      try {\n        // `keepalive`, so a hang-up sent as the tab closes still reaches the\n        // route: minutes are billed until it does.\n        await request(\"DELETE\", sessionUrl, undefined, { keepalive: true });\n      } finally {\n        close();\n      }\n    },\n    close: () => close(),\n  };\n}\n","import { useCallback, useEffect, useMemo, useRef, useState } from \"react\";\nimport type { RPC } from \"../../client/rpc\";\nimport { applyVoiceEvent, initialVoiceState, voiceStatus, type VoiceState } from \"./state\";\nimport { startVoiceCall, type VoiceCall } from \"./voiceCall\";\nimport type { VoiceCaption, VoiceStatus } from \"./wire\";\n\n/**\n * The voice routes, picked out of `RPC` the way `useChat` picks its agent\n * routes: `this.voice(Controller)` mounted at `\"/chat/voice\"` is the key\n * `\"/chat/voice\"`, carrying the controller's `Body`.\n */\ntype VoiceRoutes = {\n  [K in keyof RPC as RPC[K] extends { __voice: true } ? K : never]: RPC[K];\n};\n\ntype BodyOf<P extends keyof VoiceRoutes> = VoiceRoutes[P] extends { body: infer B }\n  ? B\n  : Record<string, unknown>;\n\nexport type UseVoiceSessionParams<P extends keyof VoiceRoutes> = {\n  /** The chat thread delegated turns are stored on, as `useChat` names it. */\n  threadId?: string;\n  /** The controller's `Body`, sent when the call starts. Change it during the\n   *  call with `setBody`. */\n  body?: BodyOf<P>;\n  /** Start with push-to-talk instead of voice activity detection. */\n  pushToTalk?: boolean;\n  /** Values for the route path's `:params`, e.g. `{ siteId }` for\n   *  `\"/sites/:siteId/voice\"`. */\n  params?: Record<string, string>;\n  headers?: Record<string, string>;\n  /** The call could not start, or failed. */\n  onError?: (error: Error) => void;\n  /** The call is over, however it ended. */\n  onEnd?: () => void;\n};\n\nexport type UseVoiceSessionResult<P extends keyof VoiceRoutes> = {\n  status: VoiceStatus;\n  /** The last lines said, the user's and the voice's, oldest first. */\n  captions: VoiceCaption[];\n  /** Why the call could not start or ended early. `code` is\n   *  `microphone_denied`, `unsupported`, or the route's error code. */\n  error: (Error & { code?: string; status?: number }) | null;\n  muted: boolean;\n  pushToTalk: boolean;\n  /** When the server will end the call, in epoch ms. */\n  endsAt?: number;\n  sessionId?: string;\n  /** Asks for the microphone and starts the call. Call it from a user gesture\n   *  (a click): browsers allow the microphone and audio playback only then. */\n  start(): Promise<void>;\n  /** Hangs up. */\n  end(): Promise<void>;\n  mute(muted?: boolean): void;\n  setPushToTalk(on: boolean): void;\n  /** Push-to-talk: hold to speak (pointer down / key down). */\n  press(): void;\n  /** Push-to-talk: release to send (pointer up / key up). */\n  release(): void;\n  /** Sends a new `body` (the selection, the open page) for the next\n   *  delegated request. */\n  setBody(body: BodyOf<P>): Promise<void>;\n};\n\n/**\n * A voice call to an agent, for a route mounted with `this.voice(Controller)`\n * (#921).\n *\n * ```tsx\n * const voice = useVoiceSession(\"/page-builder/voice\", { threadId, body: { pageId } });\n * <button onClick={() => (voice.status === \"idle\" ? voice.start() : voice.end())}>\n *   {voice.status}\n * </button>\n * ```\n *\n * Import it from `gemi/ai/voice`, an entry of its own, so an app that never\n * uses voice never bundles it, and use it from a view or a dynamically imported\n * component, so its code is not in the entry chunk.\n *\n * @experimental Not yet verified against a live call.\n */\nexport function useVoiceSession<P extends keyof VoiceRoutes>(\n  path: P,\n  params: UseVoiceSessionParams<P> = {},\n): UseVoiceSessionResult<P> {\n  // Not `useParams()`, which `useChat` uses: it is in a chunk `gemi/client`\n  // shares, and reaching it from here made the app's bundler move it out of\n  // the entry chunk into one of its own that every route then imports. A path\n  // with parameters names them in `params`.\n  const routeParams = params.params;\n  const url = useMemo(() => {\n    let out = String(path).replace(/^[A-Z]+:/, \"\");\n    for (const [key, value] of Object.entries(routeParams ?? {})) {\n      out = out.replace(`:${key}?`, value).replace(`:${key}`, value);\n    }\n    return `/api${out}`;\n  }, [path, routeParams && JSON.stringify(routeParams)]);\n\n  const [state, setState] = useState<VoiceState>(initialVoiceState);\n  const [error, setError] = useState<UseVoiceSessionResult<P>[\"error\"]>(null);\n  const [muted, setMuted] = useState(false);\n  const [pushToTalk, setPushToTalkState] = useState(params.pushToTalk === true);\n  const [call, setCall] = useState<VoiceCall | null>(null);\n  const callRef = useRef<VoiceCall | null>(null);\n  /** A `start()` in flight: a second click must not start a second call. */\n  const startingRef = useRef(false);\n  const latest = useRef(params);\n  latest.current = params;\n  const delegateRef = useRef(\"ask_agent\");\n  const mounted = useRef(true);\n\n  useEffect(() => {\n    mounted.current = true;\n    return () => {\n      mounted.current = false;\n      // Leaving the page ends the call: minutes are billed while it is up.\n      void callRef.current?.end().catch(() => undefined);\n      callRef.current = null;\n    };\n  }, []);\n\n  const start = useCallback(async () => {\n    if (callRef.current || startingRef.current) return;\n    startingRef.current = true;\n    setError(null);\n    setState({ ...initialVoiceState(), phase: \"connecting\" });\n    try {\n      const { threadId, body, headers } = latest.current;\n      const started = await startVoiceCall({\n        url,\n        threadId,\n        body,\n        headers,\n        pushToTalk,\n        onEvent: (event) => {\n          if (!mounted.current) return;\n          setState((current) =>\n            current.phase === \"connecting\"\n              ? applyVoiceEvent({ ...current, phase: \"live\" }, event, delegateRef.current)\n              : applyVoiceEvent(current, event, delegateRef.current),\n          );\n        },\n        onClose: (closeError) => {\n          callRef.current = null;\n          if (!mounted.current) return;\n          setCall(null);\n          setState((current) => ({ ...current, phase: \"ended\", speaking: false, thinking: false }));\n          if (closeError) {\n            setError(closeError);\n            latest.current.onError?.(closeError);\n          }\n          latest.current.onEnd?.();\n        },\n      });\n      delegateRef.current = started.delegate;\n      if (!mounted.current) {\n        await started.end().catch(() => undefined);\n        return;\n      }\n      callRef.current = started;\n      setCall(started);\n      setState((current) =>\n        current.phase === \"connecting\" ? { ...current, phase: \"live\" } : current,\n      );\n    } catch (failure) {\n      const reported = failure instanceof Error ? failure : new Error(String(failure));\n      if (!mounted.current) return;\n      setState((current) => ({ ...current, phase: \"ended\" }));\n      setError(reported);\n      latest.current.onError?.(reported);\n    } finally {\n      startingRef.current = false;\n    }\n  }, [url, pushToTalk]);\n\n  const end = useCallback(async () => {\n    const current = callRef.current;\n    callRef.current = null;\n    await current?.end();\n  }, []);\n\n  const mute = useCallback((on?: boolean) => {\n    setMuted((was) => {\n      const next = on ?? !was;\n      callRef.current?.mute(next);\n      return next;\n    });\n  }, []);\n\n  const setPushToTalk = useCallback((on: boolean) => {\n    setPushToTalkState(on);\n    callRef.current?.setPushToTalk(on);\n  }, []);\n\n  const press = useCallback(() => {\n    callRef.current?.press();\n    setState((current) => ({ ...current, userSpeaking: true }));\n  }, []);\n\n  const release = useCallback(() => {\n    callRef.current?.release();\n    setState((current) => ({ ...current, userSpeaking: false, thinking: true }));\n  }, []);\n\n  const setBody = useCallback(async (body: BodyOf<P>) => {\n    latest.current = { ...latest.current, body };\n    await callRef.current?.setBody(body);\n  }, []);\n\n  return {\n    status: voiceStatus(state),\n    captions: state.captions,\n    error,\n    muted,\n    pushToTalk,\n    endsAt: call?.endsAt,\n    sessionId: call?.sessionId,\n    start,\n    end,\n    mute,\n    setPushToTalk,\n    press,\n    release,\n    setBody,\n  };\n}\n","/**\n * The built-in end-of-speech detector for `useLiveTranscription` (#929): a\n * small energy (RMS) detector with an adaptive noise floor and hysteresis.\n *\n * WHY ENERGY AND NOT A MODEL. `gpt-live-transcribe` has no server-side turn\n * detection, so the browser decides when the user stopped and commits the\n * turn. A neural VAD (Silero through `@ricky0123/vad-web`) is better in\n * noise, but it ships an ONNX runtime and a model, megabytes against the few\n * hundred bytes here, and needs its wasm/model files served from the app's\n * origin. What this detector has to do is narrow: the user is close to the\n * mic, the browser's own noise suppression and echo cancellation are on, and\n * the question is only \"did the speech stop for `silenceMs`\". An adaptive\n * floor handles a fan or a room; a false end in a long pause costs one early\n * commit, after which the rest of the sentence is the next turn. Apps that\n * need more plug a model in through `vad` (see `VoiceActivityDetector`).\n *\n * Pure: frames in (RMS and a timestamp), `\"start\"` / `\"end\"` out, so it is\n * tested without audio.\n */\n\nexport type EnergyVadOptions = {\n  /** Silence after speech before the turn ends. Default 350 ms. */\n  silenceMs?: number;\n  /** Speech must last this long to count, so a click or a cough doesn't open a\n   *  turn. Default 120 ms. */\n  minSpeechMs?: number;\n  /** The lowest RMS (0–1 float samples) that can count as speech, whatever\n   *  the noise floor. Default 0.015. */\n  threshold?: number;\n  /** Speech is RMS above the noise floor times this. Default 3. */\n  floorRatio?: number;\n};\n\nexport const DEFAULT_SILENCE_MS = 350;\n\nexport type EnergyVad = {\n  /** One analysis frame: its RMS and when it was taken (ms). */\n  push(rms: number, at: number): \"start\" | \"end\" | null;\n  readonly speaking: boolean;\n  reset(): void;\n};\n\nexport function createEnergyVad(options: EnergyVadOptions = {}): EnergyVad {\n  const silenceMs = options.silenceMs ?? DEFAULT_SILENCE_MS;\n  const minSpeechMs = options.minSpeechMs ?? 120;\n  const minThreshold = options.threshold ?? 0.015;\n  const floorRatio = options.floorRatio ?? 3;\n\n  let floor = minThreshold / floorRatio;\n  let speaking = false;\n  /** When the current run of loud frames began, before it counts as speech. */\n  let loudSince: number | null = null;\n  /** When the current run of quiet frames began, while speaking. */\n  let quietSince: number | null = null;\n\n  return {\n    get speaking() {\n      return speaking;\n    },\n    reset() {\n      speaking = false;\n      loudSince = null;\n      quietSince = null;\n    },\n    push(rms, at) {\n      const threshold = Math.max(minThreshold, floor * floorRatio);\n      if (!speaking) {\n        if (rms > threshold) {\n          loudSince ??= at;\n          if (at - loudSince >= minSpeechMs) {\n            speaking = true;\n            quietSince = null;\n            return \"start\";\n          }\n        } else {\n          loudSince = null;\n          // The floor follows the room only while nobody speaks: fast down,\n          // slow up, so a rising voice is not learned as noise.\n          floor = rms < floor ? floor * 0.9 + rms * 0.1 : floor * 0.995 + rms * 0.005;\n        }\n        return null;\n      }\n      // Hysteresis: speech ends below a lower bar than it starts above, so a\n      // soft syllable doesn't end the turn.\n      if (rms > threshold * 0.7) {\n        quietSince = null;\n        return null;\n      }\n      quietSince ??= at;\n      if (at - quietSince >= silenceMs) {\n        speaking = false;\n        loudSince = null;\n        quietSince = null;\n        return \"end\";\n      }\n      return null;\n    },\n  };\n}\n\n/** RMS of float time-domain samples. */\nexport function rmsOf(samples: Float32Array): number {\n  let sum = 0;\n  for (let i = 0; i < samples.length; i++) sum += samples[i]! * samples[i]!;\n  return samples.length ? Math.sqrt(sum / samples.length) : 0;\n}\n","import { createEnergyVad, rmsOf, type EnergyVadOptions } from \"./energyVad\";\n\n/**\n * The browser's half of a live transcription session (#929): the microphone,\n * a raw `RTCPeerConnection` to the vendor, the `oai-events` data channel, a\n * level meter and end-of-speech detection that commits each turn.\n *\n * No vendor SDK, and nothing from `gemi/ai`'s server half: the secret (or the\n * SDP answer) comes from the app's own route, which holds the key.\n *\n * @experimental Follows OpenAI's realtime transcription and WebRTC guides; not\n * yet run against a live session.\n */\n\n/** OpenAI's WebRTC endpoint, for a secret whose route doesn't say otherwise.\n *  `clientSecret()` on the server returns `callsUrl` (Azure's host included). */\nexport const OPENAI_REALTIME_CALLS_URL = \"https://api.openai.com/v1/realtime/calls\";\n\n/** What the app's token route answers: `realtime.clientSecret(...)`'s result,\n *  or at least its `value`. */\nexport type LiveTranscriptionToken = {\n  value: string;\n  /** Where to post the SDP offer. Default OpenAI's. */\n  callsUrl?: string;\n};\n\n/** A pluggable end-of-speech detector, such as `@ricky0123/vad-web`'s\n *  `MicVAD` on the same stream. Call `speechStart` / `speechEnd` as speech\n *  begins and ends; `speechEnd` commits the turn. */\nexport type VoiceActivityDetector = (\n  stream: MediaStream,\n  events: { speechStart(): void; speechEnd(): void },\n) => { stop(): void } | Promise<{ stop(): void }>;\n\nexport type LiveTranscriptionError = Error & { code?: string; status?: number };\n\nexport class LiveTranscriptionFailure extends Error {\n  readonly code?: string;\n  readonly status?: number;\n\n  constructor(message: string, options: { code?: string; status?: number; cause?: unknown } = {}) {\n    super(message, { cause: options.cause });\n    this.name = \"LiveTranscriptionError\";\n    this.code = options.code;\n    this.status = options.status;\n  }\n}\n\nexport type LiveTranscriptionConnectOptions = {\n  /**\n   * `\"token\"` (default): `source` gives a client secret and the browser posts\n   * its SDP to the vendor itself. `\"sdp\"`: the browser posts `{ sdp, ...body }`\n   * to `source` and the route answers `{ sdp }` — the route then creates the\n   * call (`realtime.createCall`) and can meter it on the sideband.\n   */\n  connect?: \"token\" | \"sdp\";\n  /** The app's route (POSTed `body` as JSON), or a function giving the token. */\n  source: string | (() => Promise<LiveTranscriptionToken>);\n  body?: Record<string, unknown>;\n  headers?: Record<string, string>;\n  /** `\"energy\"` (default): the built-in detector. A function: your own. `false`:\n   *  no detection; turns end only on `commit()`. */\n  vad?: \"energy\" | VoiceActivityDetector | false;\n  /** The built-in detector's options; `silenceMs` defaults to 350. */\n  energy?: EnergyVadOptions;\n  /** The vendor's events from the data channel. */\n  onEvent(event: { type: string; [key: string]: unknown }): void;\n  /** The channel opened: the session is live. */\n  onOpen(): void;\n  onSpeechStart(): void;\n  /** The detector heard the user stop. `committed`: the turn was committed\n   *  (there was speech in it and the channel was open). */\n  onSpeechEnd(committed: boolean): void;\n  /** Microphone level, 0–1, about every 50 ms. */\n  onLevel(level: number): void;\n  /** Called once when the connection closed or failed. */\n  onClose(error?: LiveTranscriptionError): void;\n};\n\nexport type LiveTranscriptionConnection = {\n  /** Sends `input_audio_buffer.commit`. False when the channel isn't open. */\n  commit(): boolean;\n  /** Commits only if speech was heard since the last commit (always, with no\n   *  detector). For a graceful stop. */\n  flush(): boolean;\n  /** Any client event, e.g. a `session.update` with new keywords. */\n  send(event: Record<string, unknown>): boolean;\n  /** Pauses the microphone track (nothing is sent while muted). */\n  mute(on: boolean): void;\n  close(): void;\n};\n\n/** How often the level meter (and the built-in detector) reads the mic. */\nconst FRAME_MS = 30;\n\nfunction microphoneError(error: unknown): LiveTranscriptionFailure {\n  const name = (error as { name?: string } | null)?.name;\n  return name === \"NotAllowedError\" || name === \"SecurityError\"\n    ? new LiveTranscriptionFailure(\"Microphone access was denied.\", {\n        code: \"microphone_denied\",\n        cause: error,\n      })\n    : new LiveTranscriptionFailure(\"No microphone is available.\", {\n        code: \"microphone_unavailable\",\n        cause: error,\n      });\n}\n\nasync function routeError(response: Response, what: string): Promise<LiveTranscriptionFailure> {\n  const body: any = await response.json().catch(() => null);\n  return new LiveTranscriptionFailure(body?.error?.message ?? `${what} (${response.status}).`, {\n    status: response.status,\n    code: body?.error?.code,\n  });\n}\n\n/**\n * Opens a live transcription session. Resolves once the SDP answer is applied;\n * `onOpen` follows when the data channel opens. `signal` aborts a start in\n * flight (unmount, `stop()` while connecting): everything opened so far is\n * closed and the promise rejects with an `AbortError`.\n */\nexport async function startLiveTranscription(\n  options: LiveTranscriptionConnectOptions,\n  signal?: AbortSignal,\n): Promise<LiveTranscriptionConnection> {\n  if (typeof RTCPeerConnection === \"undefined\" || !navigator.mediaDevices?.getUserMedia) {\n    throw new LiveTranscriptionFailure(\"This browser can't stream audio.\", { code: \"unsupported\" });\n  }\n  const connect = options.connect ?? \"token\";\n  const post = (url: string, payload: unknown) =>\n    fetch(url, {\n      method: \"POST\",\n      headers: { \"Content-Type\": \"application/json\", ...options.headers },\n      body: JSON.stringify(payload),\n      signal,\n    });\n\n  // The token and the microphone in parallel: both are on the critical path.\n  const tokenPromise: Promise<LiveTranscriptionToken | null> =\n    connect === \"sdp\"\n      ? Promise.resolve(null)\n      : typeof options.source === \"function\"\n        ? options.source()\n        : post(options.source, options.body ?? {}).then(async (response) => {\n            if (!response.ok) throw await routeError(response, \"Live transcription is unavailable\");\n            return response.json();\n          });\n  tokenPromise.catch(() => undefined);\n\n  let stream: MediaStream;\n  try {\n    stream = await navigator.mediaDevices.getUserMedia({\n      audio: { echoCancellation: true, noiseSuppression: true, autoGainControl: true },\n    });\n  } catch (error) {\n    throw microphoneError(error);\n  }\n\n  const track = stream.getAudioTracks()[0];\n  let pc: RTCPeerConnection | undefined;\n  let channel: RTCDataChannel | undefined;\n  let audioContext: AudioContext | undefined;\n  let meter: ReturnType<typeof setInterval> | undefined;\n  let detector: { stop(): void } | undefined;\n  let closed = false;\n  /** `onClose` is for a session that started; a failed start throws instead. */\n  let started = false;\n  /** Speech heard since the last commit: a commit with none is pointless. */\n  let heardSpeech = false;\n\n  const close = (error?: LiveTranscriptionFailure) => {\n    if (closed) return;\n    closed = true;\n    if (meter) clearInterval(meter);\n    try {\n      detector?.stop();\n    } catch {\n      // a third-party detector failing to stop must not keep the mic open\n    }\n    try {\n      channel?.close();\n    } catch {\n      // already closed\n    }\n    pc?.close();\n    for (const t of stream.getTracks()) t.stop();\n    void audioContext?.close().catch(() => undefined);\n    if (started) options.onClose(error);\n  };\n\n  const send = (event: Record<string, unknown>) => {\n    if (closed || channel?.readyState !== \"open\") return false;\n    channel.send(JSON.stringify(event));\n    return true;\n  };\n  const commit = () => {\n    const sent = send({ type: \"input_audio_buffer.commit\" });\n    if (sent) heardSpeech = false;\n    return sent;\n  };\n  const speech = {\n    speechStart() {\n      if (closed) return;\n      heardSpeech = true;\n      options.onSpeechStart();\n    },\n    speechEnd() {\n      if (closed) return;\n      // Only a turn with speech in it is committed: the detector's \"end\" after\n      // a muted stretch or a reset would commit silence.\n      options.onSpeechEnd(heardSpeech ? commit() : false);\n    },\n  };\n\n  try {\n    if (!track)\n      throw new LiveTranscriptionFailure(\"No microphone is available.\", {\n        code: \"microphone_unavailable\",\n      });\n    pc = new RTCPeerConnection();\n    pc.addTrack(track, stream);\n    channel = pc.createDataChannel(\"oai-events\");\n    channel.addEventListener(\"open\", () => {\n      if (!closed) options.onOpen();\n    });\n    channel.addEventListener(\"message\", (message) => {\n      let event: unknown;\n      try {\n        event = JSON.parse(String(message.data));\n      } catch {\n        return;\n      }\n      if (\n        event &&\n        typeof event === \"object\" &&\n        typeof (event as { type?: unknown }).type === \"string\"\n      ) {\n        options.onEvent(event as { type: string });\n      }\n    });\n    channel.addEventListener(\"close\", () => close());\n    const peer = pc;\n    peer.addEventListener(\"connectionstatechange\", () => {\n      if (peer.connectionState === \"failed\") {\n        close(\n          new LiveTranscriptionFailure(\"The transcription connection failed.\", {\n            code: \"connection_failed\",\n          }),\n        );\n      }\n      if (peer.connectionState === \"closed\") close();\n    });\n\n    const offer = await pc.createOffer();\n    await pc.setLocalDescription(offer);\n    let answer: string;\n    if (connect === \"sdp\") {\n      if (typeof options.source !== \"string\") {\n        throw new TypeError('connect: \"sdp\" needs the route\\'s URL as the source.');\n      }\n      const response = await post(options.source, { ...options.body, sdp: offer.sdp });\n      if (!response.ok) throw await routeError(response, \"Live transcription is unavailable\");\n      answer = String(((await response.json()) as { sdp?: unknown }).sdp ?? \"\");\n    } else {\n      const token = await tokenPromise;\n      if (!token || typeof token.value !== \"string\") {\n        throw new LiveTranscriptionFailure(\"The token route answered no secret.\", {\n          code: \"no_token\",\n        });\n      }\n      const response = await fetch(token.callsUrl ?? OPENAI_REALTIME_CALLS_URL, {\n        method: \"POST\",\n        headers: { Authorization: `Bearer ${token.value}`, \"Content-Type\": \"application/sdp\" },\n        body: offer.sdp,\n        signal,\n      });\n      if (!response.ok)\n        throw await routeError(response, \"The transcription session could not start\");\n      answer = await response.text();\n    }\n    if (!answer) throw new LiveTranscriptionFailure(\"No SDP answer.\", { code: \"provider_error\" });\n    if (signal?.aborted) throw signal.reason ?? new DOMException(\"Aborted\", \"AbortError\");\n    await pc.setRemoteDescription({ type: \"answer\", sdp: answer });\n\n    // The level meter, which also feeds the built-in detector.\n    const energy =\n      options.vad === undefined || options.vad === \"energy\"\n        ? createEnergyVad(options.energy)\n        : null;\n    if (typeof AudioContext !== \"undefined\") {\n      audioContext = new AudioContext();\n      void audioContext.resume?.().catch(() => undefined);\n      const analyser = audioContext.createAnalyser();\n      analyser.fftSize = 1024;\n      audioContext.createMediaStreamSource(stream).connect(analyser);\n      const samples = new Float32Array(analyser.fftSize);\n      let lastLevel = -1;\n      meter = setInterval(() => {\n        analyser.getFloatTimeDomainData(samples);\n        // A muted track reads silence, so it ends a turn in progress.\n        const rms = track.enabled ? rmsOf(samples) : 0;\n        const level = Math.min(1, rms / 0.2);\n        if (Math.abs(level - lastLevel) >= 0.02) {\n          lastLevel = level;\n          options.onLevel(level);\n        }\n        const change = energy?.push(rms, Date.now());\n        if (change === \"start\") speech.speechStart();\n        else if (change === \"end\") speech.speechEnd();\n      }, FRAME_MS);\n    } else if (energy) {\n      throw new LiveTranscriptionFailure(\"This browser can't analyse audio.\", {\n        code: \"unsupported\",\n      });\n    }\n    if (typeof options.vad === \"function\") detector = await options.vad(stream, speech);\n  } catch (error) {\n    close();\n    if (signal?.aborted || (error as { name?: string })?.name === \"AbortError\") throw error;\n    throw error instanceof LiveTranscriptionFailure\n      ? error\n      : new LiveTranscriptionFailure(\"Live transcription could not start.\", { cause: error });\n  }\n\n  if (closed) {\n    throw new LiveTranscriptionFailure(\"The transcription connection closed while starting.\", {\n      code: \"connection_failed\",\n    });\n  }\n  started = true;\n  return {\n    commit,\n    flush: () => (heardSpeech || options.vad === false ? commit() : false),\n    send,\n    mute(on) {\n      if (track) track.enabled = !on;\n    },\n    close: () => close(),\n  };\n}\n","/**\n * What `useLiveTranscription` shows, as a pure reducer over a transcription\n * session's data-channel events (#929). Tested without a DOM or a peer\n * connection, like `state.ts` for `useVoiceSession`.\n *\n * The events are OpenAI's (realtime transcription guide):\n * `conversation.item.input_audio_transcription.delta` (`item_id`, `delta`)\n * while the user speaks, and `.completed` (`item_id`, `transcript`, `usage`)\n * once the turn is committed. Completions of different turns may arrive out of\n * order, so turns are keyed and ordered by `item_id`, by when each item was\n * first seen.\n */\n\n/** One finished turn: what the user said between two commits. */\nexport type TranscriptTurn = {\n  /** The vendor's item id. */\n  id: string;\n  text: string;\n  /** Audio seconds the vendor billed for it, when it said (duration-billed\n   *  models such as `gpt-live-transcribe`). */\n  seconds?: number;\n};\n\nexport type TranscriptionState = {\n  /** Item ids in the order they were first seen. */\n  order: string[];\n  /** Text so far of items not yet completed. */\n  partials: Record<string, string>;\n  /** Completed turns, oldest first, at most `MAX_TURNS`. */\n  turns: TranscriptTurn[];\n  /** Commits sent whose transcript hasn't arrived. */\n  pendingCommits: number;\n};\n\nexport const MAX_TURNS = 50;\n\nexport function initialTranscriptionState(): TranscriptionState {\n  return { order: [], partials: {}, turns: [], pendingCommits: 0 };\n}\n\n/** The live text: every unfinished item's text, in order. */\nexport function partialText(state: TranscriptionState): string {\n  return state.order\n    .map((id) => state.partials[id])\n    .filter((text): text is string => !!text)\n    .join(\" \")\n    .trim();\n}\n\nexport type TranscriptionEvent = { type: string; [key: string]: unknown };\n\nexport type TranscriptionEffect =\n  | { type: \"final\"; turn: TranscriptTurn }\n  | { type: \"error\"; code?: string; message: string };\n\n/** Vendor error codes that are part of normal operation, not failures: a\n *  commit with nothing in the buffer (a manual `commit()` in silence). */\nconst BENIGN_ERRORS = new Set([\"input_audio_buffer_commit_empty\"]);\n\nexport function applyTranscriptionEvent(\n  state: TranscriptionState,\n  event: TranscriptionEvent,\n): { state: TranscriptionState; effects: TranscriptionEffect[] } {\n  const itemId = typeof event.item_id === \"string\" ? event.item_id : undefined;\n  const seen = (id: string) => (state.order.includes(id) ? state.order : [...state.order, id]);\n\n  switch (event.type) {\n    case \"conversation.item.input_audio_transcription.delta\": {\n      if (!itemId || typeof event.delta !== \"string\") break;\n      if (state.turns.some((turn) => turn.id === itemId)) break;\n      return {\n        state: {\n          ...state,\n          order: seen(itemId),\n          partials: { ...state.partials, [itemId]: (state.partials[itemId] ?? \"\") + event.delta },\n        },\n        effects: [],\n      };\n    }\n    case \"input_audio_buffer.committed\": {\n      if (!itemId) break;\n      return { state: { ...state, order: seen(itemId) }, effects: [] };\n    }\n    case \"conversation.item.input_audio_transcription.completed\": {\n      if (!itemId || state.turns.some((turn) => turn.id === itemId)) break;\n      const usage = event.usage as { type?: string; seconds?: unknown } | undefined;\n      const seconds = usage?.type === \"duration\" ? Number(usage.seconds) : NaN;\n      const turn: TranscriptTurn = {\n        id: itemId,\n        text: typeof event.transcript === \"string\" ? event.transcript.trim() : \"\",\n        ...(Number.isFinite(seconds) ? { seconds } : {}),\n      };\n      const order = seen(itemId);\n      const { [itemId]: _done, ...partials } = state.partials;\n      const all = [...state.turns, turn].sort((a, b) => order.indexOf(a.id) - order.indexOf(b.id));\n      const turns = all.slice(-MAX_TURNS);\n      // Turns that fell off the end leave `order` too, so a long session\n      // doesn't grow it without bound. Items still pending stay.\n      const dropped = new Set(all.slice(0, -MAX_TURNS).map((t) => t.id));\n      return {\n        state: {\n          ...state,\n          order: dropped.size ? order.filter((id) => !dropped.has(id)) : order,\n          partials,\n          turns,\n          pendingCommits: Math.max(0, state.pendingCommits - 1),\n        },\n        effects: [{ type: \"final\", turn }],\n      };\n    }\n    case \"conversation.item.input_audio_transcription.failed\": {\n      const { [itemId ?? \"\"]: _dropped, ...partials } = state.partials;\n      const error = (event.error ?? {}) as { code?: string; message?: string };\n      return {\n        state: { ...state, partials, pendingCommits: Math.max(0, state.pendingCommits - 1) },\n        effects: [\n          { type: \"error\", code: error.code, message: error.message ?? \"Transcription failed.\" },\n        ],\n      };\n    }\n    case \"error\": {\n      const error = (event.error ?? {}) as { code?: string; message?: string };\n      if (error.code && BENIGN_ERRORS.has(error.code)) {\n        return {\n          state: { ...state, pendingCommits: Math.max(0, state.pendingCommits - 1) },\n          effects: [],\n        };\n      }\n      return {\n        state,\n        effects: [{ type: \"error\", code: error.code, message: error.message ?? \"Vendor error.\" }],\n      };\n    }\n  }\n  return { state, effects: [] };\n}\n","import { useCallback, useEffect, useRef, useState } from \"react\";\nimport type { EnergyVadOptions } from \"./energyVad\";\nimport {\n  startLiveTranscription,\n  type LiveTranscriptionConnection,\n  type LiveTranscriptionError,\n  type LiveTranscriptionToken,\n  type VoiceActivityDetector,\n} from \"./liveTranscription\";\nimport {\n  applyTranscriptionEvent,\n  initialTranscriptionState,\n  partialText,\n  type TranscriptionState,\n  type TranscriptTurn,\n} from \"./transcriptionState\";\n\nexport type LiveTranscriptionStatus = \"idle\" | \"connecting\" | \"listening\";\n\nexport type UseLiveTranscriptionOptions = {\n  /** `\"token\"` (default) or `\"sdp\"`. See the docs: `\"sdp\"` lets the route\n   *  create the call and meter it on the sideband. */\n  connect?: \"token\" | \"sdp\";\n  /** Sent to the route as JSON with each start. */\n  body?: Record<string, unknown>;\n  headers?: Record<string, string>;\n  /** Silence after speech that ends a turn, for the built-in detector.\n   *  Default 350 ms. */\n  silenceMs?: number;\n  /** More options for the built-in detector. */\n  energy?: Omit<EnergyVadOptions, \"silenceMs\">;\n  /** `\"energy\"` (default), your own detector (e.g. `@ricky0123/vad-web`), or\n   *  `false` to end turns only with `commit()`. */\n  vad?: \"energy\" | VoiceActivityDetector | false;\n  /** The live text changed (a delta arrived, or a turn completed). */\n  onPartial?: (text: string) => void;\n  /** A turn's final transcript. */\n  onFinal?: (turn: TranscriptTurn) => void;\n  onSpeechStart?: () => void;\n  /** The user stopped; the turn was committed and its final is on the way. */\n  onSpeechEnd?: () => void;\n  onError?: (error: LiveTranscriptionError) => void;\n};\n\nexport type UseLiveTranscriptionResult = {\n  status: LiveTranscriptionStatus;\n  /** What the user is saying now, before the turn is committed. */\n  partial: string;\n  /** Finished turns, oldest first (the last 50). */\n  turns: TranscriptTurn[];\n  /** The user is speaking, by the end-of-speech detector. */\n  speaking: boolean;\n  /** Microphone level, 0–1. */\n  level: number;\n  muted: boolean;\n  /** The last failure. `code` is `microphone_denied`,\n   *  `microphone_unavailable`, `unsupported`, `connection_failed`, the\n   *  route's error code, or the vendor's. */\n  error: LiveTranscriptionError | null;\n  /** Asks for the microphone and opens the session. Call it from a user\n   *  gesture. Does nothing while a session is starting or live. */\n  start(): Promise<void>;\n  /** Closes the session and releases the microphone. `{ flush: true }`\n   *  commits speech in progress first and waits (up to `timeoutMs`, default\n   *  1500) for its final transcript. */\n  stop(options?: { flush?: boolean; timeoutMs?: number }): Promise<void>;\n  /** Ends the turn now (`input_audio_buffer.commit`), e.g. for push-to-talk or\n   *  a \"send\" button. */\n  commit(): void;\n  mute(on?: boolean): void;\n  /** Clears `turns` and `partial` (not the session). */\n  reset(): void;\n};\n\n/**\n * Live speech-to-text (#929): the microphone streams to a transcription\n * session (`gpt-live-transcribe` by default) over WebRTC. `partial` grows as\n * the user speaks; when they stop for `silenceMs`, the turn is committed and\n * its final transcript arrives in `turns` and `onFinal`. Nothing is spoken\n * back.\n *\n * `source` is the app's route that answers a client secret (from\n * `realtime.clientSecret({ session: { type: \"transcription\", … } })`), or a\n * function returning one. With `connect: \"sdp\"`, it is the route that takes\n * the SDP offer and answers `{ sdp }`.\n *\n * ```tsx\n * const live = useLiveTranscription(\"/api/voice/token\", {\n *   onFinal: (turn) => route(turn.text),\n * });\n * <button onClick={() => (live.status === \"idle\" ? live.start() : live.stop())}>Voice</button>\n * <p>{live.partial}</p>\n * ```\n *\n * From `gemi/ai/voice`, so only the views that use it load it.\n *\n * @experimental Tested against faked browser APIs only.\n */\nexport function useLiveTranscription(\n  source: string | (() => Promise<LiveTranscriptionToken>),\n  options: UseLiveTranscriptionOptions = {},\n): UseLiveTranscriptionResult {\n  const [status, setStatus] = useState<LiveTranscriptionStatus>(\"idle\");\n  const [transcript, setTranscript] = useState<TranscriptionState>(initialTranscriptionState);\n  const [speaking, setSpeaking] = useState(false);\n  const [level, setLevel] = useState(0);\n  const [muted, setMuted] = useState(false);\n  const [error, setError] = useState<LiveTranscriptionError | null>(null);\n\n  const latest = useRef({ source, options });\n  latest.current = { source, options };\n  const stateRef = useRef(transcript);\n  const connectionRef = useRef<LiveTranscriptionConnection | null>(null);\n  const abortRef = useRef<AbortController | null>(null);\n  /** Resolves `stop({ flush })`'s wait when the last pending final arrives. */\n  const drainedRef = useRef<(() => void) | null>(null);\n  const mounted = useRef(true);\n\n  const update = useCallback((next: TranscriptionState) => {\n    const before = partialText(stateRef.current);\n    stateRef.current = next;\n    if (!mounted.current) return;\n    setTranscript(next);\n    const after = partialText(next);\n    if (after !== before) latest.current.options.onPartial?.(after);\n  }, []);\n\n  const fail = useCallback((failure: LiveTranscriptionError) => {\n    if (!mounted.current) return;\n    setError(failure);\n    latest.current.options.onError?.(failure);\n  }, []);\n\n  const teardown = useCallback(() => {\n    abortRef.current?.abort();\n    abortRef.current = null;\n    const connection = connectionRef.current;\n    connectionRef.current = null;\n    connection?.close();\n  }, []);\n\n  useEffect(() => {\n    mounted.current = true;\n    return () => {\n      mounted.current = false;\n      // Leaving the page releases the mic and ends the session (it is billed\n      // while open).\n      teardown();\n    };\n  }, [teardown]);\n\n  const start = useCallback(async () => {\n    if (connectionRef.current || abortRef.current) return;\n    const abort = new AbortController();\n    abortRef.current = abort;\n    setError(null);\n    setStatus(\"connecting\");\n    const { source: src, options: opts } = latest.current;\n    let opened = false;\n    try {\n      const connection = await startLiveTranscription(\n        {\n          connect: opts.connect,\n          source: src,\n          body: opts.body,\n          headers: opts.headers,\n          vad: opts.vad,\n          energy: { ...opts.energy, silenceMs: opts.silenceMs },\n          onOpen: () => {\n            opened = true;\n            if (mounted.current && connectionRef.current) setStatus(\"listening\");\n          },\n          onEvent: (event) => {\n            const { state, effects } = applyTranscriptionEvent(stateRef.current, event);\n            if (state !== stateRef.current) update(state);\n            for (const effect of effects) {\n              if (effect.type === \"final\") {\n                if (mounted.current) latest.current.options.onFinal?.(effect.turn);\n              } else {\n                fail(Object.assign(new Error(effect.message), { code: effect.code }));\n              }\n            }\n            if (stateRef.current.pendingCommits === 0) drainedRef.current?.();\n          },\n          onSpeechStart: () => {\n            if (!mounted.current) return;\n            setSpeaking(true);\n            latest.current.options.onSpeechStart?.();\n          },\n          onSpeechEnd: (committed) => {\n            if (committed) {\n              update({ ...stateRef.current, pendingCommits: stateRef.current.pendingCommits + 1 });\n            }\n            if (!mounted.current) return;\n            setSpeaking(false);\n            latest.current.options.onSpeechEnd?.();\n          },\n          onLevel: (value) => {\n            if (mounted.current) setLevel(value);\n          },\n          onClose: (closeError) => {\n            connectionRef.current = null;\n            drainedRef.current?.();\n            if (!mounted.current) return;\n            setStatus(\"idle\");\n            setSpeaking(false);\n            setLevel(0);\n            if (closeError) fail(closeError);\n          },\n        },\n        abort.signal,\n      );\n      if (abort.signal.aborted) {\n        connection.close();\n        return;\n      }\n      connectionRef.current = connection;\n      connection.mute(muted);\n      if (opened && mounted.current) setStatus(\"listening\");\n    } catch (failure) {\n      if (abort.signal.aborted) return;\n      if (mounted.current) setStatus(\"idle\");\n      fail(failure instanceof Error ? failure : new Error(String(failure)));\n    } finally {\n      if (abortRef.current === abort) abortRef.current = null;\n    }\n  }, [fail, update, muted]);\n\n  const commit = useCallback(() => {\n    if (connectionRef.current?.commit()) {\n      update({ ...stateRef.current, pendingCommits: stateRef.current.pendingCommits + 1 });\n    }\n  }, [update]);\n\n  const stop = useCallback(\n    async ({ flush = false, timeoutMs = 1500 }: { flush?: boolean; timeoutMs?: number } = {}) => {\n      const connection = connectionRef.current;\n      if (flush && connection) {\n        if (connection.flush()) {\n          update({ ...stateRef.current, pendingCommits: stateRef.current.pendingCommits + 1 });\n        }\n        if 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package/dist/ai/voice/liveTranscription.d.ts

@@ -0,0 +1,99 @@
+import { type EnergyVadOptions } from "./energyVad";
+/**
+ * The browser's half of a live transcription session (#929): the microphone,
+ * a raw `RTCPeerConnection` to the vendor, the `oai-events` data channel, a
+ * level meter and end-of-speech detection that commits each turn.
+ *
+ * No vendor SDK, and nothing from `gemi/ai`'s server half: the secret (or the
+ * SDP answer) comes from the app's own route, which holds the key.
+ *
+ * @experimental Follows OpenAI's realtime transcription and WebRTC guides; not
+ * yet run against a live session.
+ */
+/** OpenAI's WebRTC endpoint, for a secret whose route doesn't say otherwise.
+ *  `clientSecret()` on the server returns `callsUrl` (Azure's host included). */
+export declare const OPENAI_REALTIME_CALLS_URL = "https://api.openai.com/v1/realtime/calls";
+/** What the app's token route answers: `realtime.clientSecret(...)`'s result,
+ *  or at least its `value`. */
+export type LiveTranscriptionToken = {
+    value: string;
+    /** Where to post the SDP offer. Default OpenAI's. */
+    callsUrl?: string;
+};
+/** A pluggable end-of-speech detector, such as `@ricky0123/vad-web`'s
+ *  `MicVAD` on the same stream. Call `speechStart` / `speechEnd` as speech
+ *  begins and ends; `speechEnd` commits the turn. */
+export type VoiceActivityDetector = (stream: MediaStream, events: {
+    speechStart(): void;
+    speechEnd(): void;
+}) => {
+    stop(): void;
+} | Promise<{
+    stop(): void;
+}>;
+export type LiveTranscriptionError = Error & {
+    code?: string;
+    status?: number;
+};
+export declare class LiveTranscriptionFailure extends Error {
+    readonly code?: string;
+    readonly status?: number;
+    constructor(message: string, options?: {
+        code?: string;
+        status?: number;
+        cause?: unknown;
+    });
+}
+export type LiveTranscriptionConnectOptions = {
+    /**
+     * `"token"` (default): `source` gives a client secret and the browser posts
+     * its SDP to the vendor itself. `"sdp"`: the browser posts `{ sdp, ...body }`
+     * to `source` and the route answers `{ sdp }` — the route then creates the
+     * call (`realtime.createCall`) and can meter it on the sideband.
+     */
+    connect?: "token" | "sdp";
+    /** The app's route (POSTed `body` as JSON), or a function giving the token. */
+    source: string | (() => Promise<LiveTranscriptionToken>);
+    body?: Record<string, unknown>;
+    headers?: Record<string, string>;
+    /** `"energy"` (default): the built-in detector. A function: your own. `false`:
+     *  no detection; turns end only on `commit()`. */
+    vad?: "energy" | VoiceActivityDetector | false;
+    /** The built-in detector's options; `silenceMs` defaults to 350. */
+    energy?: EnergyVadOptions;
+    /** The vendor's events from the data channel. */
+    onEvent(event: {
+        type: string;
+        [key: string]: unknown;
+    }): void;
+    /** The channel opened: the session is live. */
+    onOpen(): void;
+    onSpeechStart(): void;
+    /** The detector heard the user stop. `committed`: the turn was committed
+     *  (there was speech in it and the channel was open). */
+    onSpeechEnd(committed: boolean): void;
+    /** Microphone level, 0–1, about every 50 ms. */
+    onLevel(level: number): void;
+    /** Called once when the connection closed or failed. */
+    onClose(error?: LiveTranscriptionError): void;
+};
+export type LiveTranscriptionConnection = {
+    /** Sends `input_audio_buffer.commit`. False when the channel isn't open. */
+    commit(): boolean;
+    /** Commits only if speech was heard since the last commit (always, with no
+     *  detector). For a graceful stop. */
+    flush(): boolean;
+    /** Any client event, e.g. a `session.update` with new keywords. */
+    send(event: Record<string, unknown>): boolean;
+    /** Pauses the microphone track (nothing is sent while muted). */
+    mute(on: boolean): void;
+    close(): void;
+};
+/**
+ * Opens a live transcription session. Resolves once the SDP answer is applied;
+ * `onOpen` follows when the data channel opens. `signal` aborts a start in
+ * flight (unmount, `stop()` while connecting): everything opened so far is
+ * closed and the promise rejects with an `AbortError`.
+ */
+export declare function startLiveTranscription(options: LiveTranscriptionConnectOptions, signal?: AbortSignal): Promise<LiveTranscriptionConnection>;
+//# sourceMappingURL=liveTranscription.d.ts.map

package/dist/ai/voice/liveTranscription.d.ts.map

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package/dist/ai/voice/transcriptionState.d.ts

@@ -0,0 +1,52 @@
+/**
+ * What `useLiveTranscription` shows, as a pure reducer over a transcription
+ * session's data-channel events (#929). Tested without a DOM or a peer
+ * connection, like `state.ts` for `useVoiceSession`.
+ *
+ * The events are OpenAI's (realtime transcription guide):
+ * `conversation.item.input_audio_transcription.delta` (`item_id`, `delta`)
+ * while the user speaks, and `.completed` (`item_id`, `transcript`, `usage`)
+ * once the turn is committed. Completions of different turns may arrive out of
+ * order, so turns are keyed and ordered by `item_id`, by when each item was
+ * first seen.
+ */
+/** One finished turn: what the user said between two commits. */
+export type TranscriptTurn = {
+    /** The vendor's item id. */
+    id: string;
+    text: string;
+    /** Audio seconds the vendor billed for it, when it said (duration-billed
+     *  models such as `gpt-live-transcribe`). */
+    seconds?: number;
+};
+export type TranscriptionState = {
+    /** Item ids in the order they were first seen. */
+    order: string[];
+    /** Text so far of items not yet completed. */
+    partials: Record<string, string>;
+    /** Completed turns, oldest first, at most `MAX_TURNS`. */
+    turns: TranscriptTurn[];
+    /** Commits sent whose transcript hasn't arrived. */
+    pendingCommits: number;
+};
+export declare const MAX_TURNS = 50;
+export declare function initialTranscriptionState(): TranscriptionState;
+/** The live text: every unfinished item's text, in order. */
+export declare function partialText(state: TranscriptionState): string;
+export type TranscriptionEvent = {
+    type: string;
+    [key: string]: unknown;
+};
+export type TranscriptionEffect = {
+    type: "final";
+    turn: TranscriptTurn;
+} | {
+    type: "error";
+    code?: string;
+    message: string;
+};
+export declare function applyTranscriptionEvent(state: TranscriptionState, event: TranscriptionEvent): {
+    state: TranscriptionState;
+    effects: TranscriptionEffect[];
+};
+//# sourceMappingURL=transcriptionState.d.ts.map

package/dist/ai/voice/transcriptionState.d.ts.map

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package/dist/ai/voice/useLiveTranscription.d.ts

@@ -0,0 +1,86 @@
+import type { EnergyVadOptions } from "./energyVad";
+import { type LiveTranscriptionError, type LiveTranscriptionToken, type VoiceActivityDetector } from "./liveTranscription";
+import { type TranscriptTurn } from "./transcriptionState";
+export type LiveTranscriptionStatus = "idle" | "connecting" | "listening";
+export type UseLiveTranscriptionOptions = {
+    /** `"token"` (default) or `"sdp"`. See the docs: `"sdp"` lets the route
+     *  create the call and meter it on the sideband. */
+    connect?: "token" | "sdp";
+    /** Sent to the route as JSON with each start. */
+    body?: Record<string, unknown>;
+    headers?: Record<string, string>;
+    /** Silence after speech that ends a turn, for the built-in detector.
+     *  Default 350 ms. */
+    silenceMs?: number;
+    /** More options for the built-in detector. */
+    energy?: Omit<EnergyVadOptions, "silenceMs">;
+    /** `"energy"` (default), your own detector (e.g. `@ricky0123/vad-web`), or
+     *  `false` to end turns only with `commit()`. */
+    vad?: "energy" | VoiceActivityDetector | false;
+    /** The live text changed (a delta arrived, or a turn completed). */
+    onPartial?: (text: string) => void;
+    /** A turn's final transcript. */
+    onFinal?: (turn: TranscriptTurn) => void;
+    onSpeechStart?: () => void;
+    /** The user stopped; the turn was committed and its final is on the way. */
+    onSpeechEnd?: () => void;
+    onError?: (error: LiveTranscriptionError) => void;
+};
+export type UseLiveTranscriptionResult = {
+    status: LiveTranscriptionStatus;
+    /** What the user is saying now, before the turn is committed. */
+    partial: string;
+    /** Finished turns, oldest first (the last 50). */
+    turns: TranscriptTurn[];
+    /** The user is speaking, by the end-of-speech detector. */
+    speaking: boolean;
+    /** Microphone level, 0–1. */
+    level: number;
+    muted: boolean;
+    /** The last failure. `code` is `microphone_denied`,
+     *  `microphone_unavailable`, `unsupported`, `connection_failed`, the
+     *  route's error code, or the vendor's. */
+    error: LiveTranscriptionError | null;
+    /** Asks for the microphone and opens the session. Call it from a user
+     *  gesture. Does nothing while a session is starting or live. */
+    start(): Promise<void>;
+    /** Closes the session and releases the microphone. `{ flush: true }`
+     *  commits speech in progress first and waits (up to `timeoutMs`, default
+     *  1500) for its final transcript. */
+    stop(options?: {
+        flush?: boolean;
+        timeoutMs?: number;
+    }): Promise<void>;
+    /** Ends the turn now (`input_audio_buffer.commit`), e.g. for push-to-talk or
+     *  a "send" button. */
+    commit(): void;
+    mute(on?: boolean): void;
+    /** Clears `turns` and `partial` (not the session). */
+    reset(): void;
+};
+/**
+ * Live speech-to-text (#929): the microphone streams to a transcription
+ * session (`gpt-live-transcribe` by default) over WebRTC. `partial` grows as
+ * the user speaks; when they stop for `silenceMs`, the turn is committed and
+ * its final transcript arrives in `turns` and `onFinal`. Nothing is spoken
+ * back.
+ *
+ * `source` is the app's route that answers a client secret (from
+ * `realtime.clientSecret({ session: { type: "transcription", … } })`), or a
+ * function returning one. With `connect: "sdp"`, it is the route that takes
+ * the SDP offer and answers `{ sdp }`.
+ *
+ * ```tsx
+ * const live = useLiveTranscription("/api/voice/token", {
+ *   onFinal: (turn) => route(turn.text),
+ * });
+ * <button onClick={() => (live.status === "idle" ? live.start() : live.stop())}>Voice</button>
+ * <p>{live.partial}</p>
+ * ```
+ *
+ * From `gemi/ai/voice`, so only the views that use it load it.
+ *
+ * @experimental Tested against faked browser APIs only.
+ */
+export declare function useLiveTranscription(source: string | (() => Promise<LiveTranscriptionToken>), options?: UseLiveTranscriptionOptions): UseLiveTranscriptionResult;
+//# sourceMappingURL=useLiveTranscription.d.ts.map

package/dist/ai/voice/useLiveTranscription.d.ts.map

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package/dist/{chunk-2vteyrjj.js → chunk-3xm7ysef.js}

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+import{R as S,T as h,U as u}from"./chunk-c7g1jb3f.js";import{ma as q,na as w}from"./chunk-mpqvmge4.js";import"./chunk-a22h5cby.js";import"./chunk-epvpmakp.js";import"./chunk-r3v2cnrj.js";import"./chunk-4278e429.js";import"./chunk-kk18xqgc.js";import"./chunk-sa6cxt3g.js";import{ti as N}from"./chunk-zv5qyrqg.js";import{Lk as y}from"./chunk-x9bk5s21.js";import{_k as C}from"./chunk-99a4e6zf.js";import"./chunk-rnp5k8m3.js";import{join as $J}from"path";import{createServer as zJ}from"vite";import{join as l}from"path";import{IncomingMessage as d,ServerResponse as p}from"http";import{Socket as i}from"net";function c(J){return J.replace(/\x1b\[[0-9;]*m/g,"").replace(/\[[0-9;]*m/g,"").replace(/\n\s*\n/g,`
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package/dist/{chunk-t79n15f8.js → chunk-778kv5wf.js}

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