@happyvertical/speech 0.101.2 → 0.102.0

package/dist/index.js.map

@@ -1 +1 @@
-{"version":3,"file":"index.js","names":[],"sources":["../src/shared/retry.ts","../src/shared/http.ts","../src/shared/realtime-auth.ts","../src/shared/pcm.ts","../src/shared/turn-limit.ts","../src/shared/realtime-session.ts","../src/adapters/openai-realtime.ts","../src/adapters/voxtral-realtime.ts","../src/adapters/openai-compatible.ts","../src/adapters/openai-compatible-transcriber.ts","../src/adapters/qwen3.ts","../src/adapters/studio-server.ts","../src/shared/streaming-factory.ts","../src/shared/factory.ts","../src/shared/streaming-client-secret.ts"],"sourcesContent":["/**\n * Retry with exponential backoff for speech provider calls. Retries only\n * rate-limit (429) and server (5xx) responses, honours `Retry-After`, and\n * stops immediately when the caller aborts.\n */\n\nimport { SpeechConfigurationError, SpeechProviderError } from './errors.js';\n\nexport interface SpeechRetryOptions {\n  /** Retries after the first attempt. Default `2`. `0` disables retries. */\n  maxRetries?: number;\n  /** First backoff delay in milliseconds. Default `500`. */\n  initialDelayMs?: number;\n  /**\n   * Upper bound for any single wait in milliseconds. Default `30000`. A\n   * `Retry-After` longer than this ends retrying rather than retrying early.\n   */\n  maxDelayMs?: number;\n}\n\nexport const DEFAULT_SPEECH_RETRY: Required<SpeechRetryOptions> = {\n  maxRetries: 2,\n  initialDelayMs: 500,\n  maxDelayMs: 30_000,\n};\n\nexport function resolveRetryOptions(\n  retry: SpeechRetryOptions | false | undefined,\n  defaults: Required<SpeechRetryOptions> | false = DEFAULT_SPEECH_RETRY,\n): Required<SpeechRetryOptions> | false {\n  if (retry === false) {\n    return false;\n  }\n\n  if (retry === undefined) {\n    return defaults;\n  }\n\n  const base = defaults || DEFAULT_SPEECH_RETRY;\n  return {\n    maxRetries: Math.floor(\n      finiteRetryValue('maxRetries', retry.maxRetries, base.maxRetries),\n    ),\n    initialDelayMs: finiteRetryValue(\n      'initialDelayMs',\n      retry.initialDelayMs,\n      base.initialDelayMs,\n    ),\n    maxDelayMs: finiteRetryValue(\n      'maxDelayMs',\n      retry.maxDelayMs,\n      base.maxDelayMs,\n    ),\n  };\n}\n\n/**\n * Rejects non-finite values (`NaN`, `Infinity`): they would make the retry\n * budget or delay cap unbounded. Negative values clamp to `0`.\n */\nfunction finiteRetryValue(\n  name: keyof SpeechRetryOptions,\n  value: number | undefined,\n  fallback: number,\n): number {\n  if (value === undefined) {\n    return Math.max(0, fallback);\n  }\n\n  if (typeof value !== 'number' || !Number.isFinite(value)) {\n    throw new SpeechConfigurationError(\n      `Speech retry option ${name} must be a finite number`,\n    );\n  }\n\n  return Math.max(0, value);\n}\n\nexport function isRetryableStatus(status: number | undefined): boolean {\n  return status === 429 || (status !== undefined && status >= 500);\n}\n\n/**\n * Parses a `Retry-After` header (delta-seconds or HTTP-date) into\n * milliseconds. Returns `undefined` when absent or unparseable.\n */\nexport function parseRetryAfter(\n  value: string | null | undefined,\n  now: number = Date.now(),\n): number | undefined {\n  if (!value?.trim()) {\n    return undefined;\n  }\n\n  const trimmed = value.trim();\n  if (/^\\d+(\\.\\d+)?$/.test(trimmed)) {\n    return Math.round(Number(trimmed) * 1000);\n  }\n\n  const date = Date.parse(trimmed);\n  return Number.isNaN(date) ? undefined : Math.max(0, date - now);\n}\n\n/**\n * Runs `attempt` until it succeeds, throws a non-retryable error, or the\n * retry budget is exhausted. Retryable errors are {@link SpeechProviderError}\n * instances whose `status` is 429 or 5xx.\n */\nexport async function withSpeechRetry<T>(\n  attempt: (attemptNumber: number) => Promise<T>,\n  retry: Required<SpeechRetryOptions> | false,\n  signal?: AbortSignal,\n): Promise<T> {\n  const maxRetries = retry ? retry.maxRetries : 0;\n\n  for (let attemptNumber = 0; ; attemptNumber += 1) {\n    try {\n      return await attempt(attemptNumber);\n    } catch (error) {\n      if (\n        !retry ||\n        attemptNumber >= maxRetries ||\n        signal?.aborted ||\n        !(error instanceof SpeechProviderError) ||\n        !isRetryableStatus(error.status)\n      ) {\n        throw error;\n      }\n\n      const backoff = Math.min(\n        retry.maxDelayMs,\n        retry.initialDelayMs * 2 ** attemptNumber,\n      );\n      const retryAfterMs = error.retryAfterMs;\n      if (retryAfterMs !== undefined && retryAfterMs > retry.maxDelayMs) {\n        throw error;\n      }\n\n      await sleep(Math.max(backoff, retryAfterMs ?? 0), signal);\n    }\n  }\n}\n\nfunction sleep(ms: number, signal?: AbortSignal): Promise<void> {\n  return new Promise((resolve, reject) => {\n    if (signal?.aborted) {\n      reject(signal.reason);\n      return;\n    }\n\n    const onAbort = () => {\n      clearTimeout(timer);\n      reject(signal?.reason);\n    };\n    const timer = setTimeout(() => {\n      signal?.removeEventListener('abort', onAbort);\n      resolve();\n    }, ms);\n    signal?.addEventListener('abort', onAbort, { once: true });\n  });\n}\n","import { SpeechConfigurationError, SpeechProviderError } from './errors.js';\nimport {\n  parseRetryAfter,\n  type SpeechRetryOptions,\n  withSpeechRetry,\n} from './retry.js';\nimport type {\n  HttpSpeechOptions,\n  SpeechFetch,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  TranscriptResult,\n  TranscriptSegment,\n  WordTiming,\n} from './types.js';\n\n/**\n * Parses a base URL, dropping any fragment and keeping any query string.\n * Throws {@link SpeechConfigurationError} for a blank or unparseable value.\n * The value is not echoed in the error because it may carry credentials.\n */\nfunction parseBaseUrl(baseUrl: string): URL {\n  const trimmed = baseUrl.trim();\n  if (!trimmed) {\n    throw new SpeechConfigurationError('Speech adapter baseUrl is required');\n  }\n\n  let url: URL;\n  try {\n    url = new URL(trimmed);\n  } catch {\n    throw new SpeechConfigurationError(\n      'Speech adapter baseUrl is not a valid absolute URL',\n    );\n  }\n\n  url.hash = '';\n  return url;\n}\n\n/**\n * Returns `baseUrl` with a trailing `/` on its path so relative paths resolve\n * beneath it. The query string is preserved and any fragment is dropped.\n */\nexport function normalizeBaseUrl(baseUrl: string): string {\n  const url = parseBaseUrl(baseUrl);\n  if (!url.pathname.endsWith('/')) {\n    url.pathname = `${url.pathname}/`;\n  }\n\n  return url.toString();\n}\n\n/**\n * Resolves `path` against `baseUrl`. A relative path without its own query\n * inherits the base URL's query (for example a gateway `api-version`); an\n * absolute URL is returned unchanged.\n */\nexport function resolveSpeechUrl(baseUrl: string, path: string): string {\n  const base = new URL(normalizeBaseUrl(baseUrl));\n  const isAbsolute = /^[a-z][a-z\\d+.-]*:/i.test(path);\n  const resolved = new URL(isAbsolute ? path : path.replace(/^\\//, ''), base);\n  if (!isAbsolute && !resolved.search && base.search) {\n    resolved.search = base.search;\n  }\n\n  return resolved.toString();\n}\n\n/**\n * Resolves an OpenAI-compatible endpoint from a base URL. Accepts a server\n * root (`https://api.openai.com` → `/v1/<resource>`), an API root ending in a\n * version segment (`http://gateway/stt/v1` → `/<resource>`), or the full\n * endpoint URL (used as-is). Only the path is rewritten: the query string is\n * preserved and any fragment is dropped.\n */\nexport function resolveOpenAICompatibleUrl(\n  baseUrl: string,\n  resource: string,\n): string {\n  const trimmedResource = resource.replace(/^\\/+|\\/+$/g, '');\n  const url = parseBaseUrl(baseUrl);\n  const pathname = url.pathname.replace(/\\/+$/, '');\n\n  if (pathname.endsWith(`/${trimmedResource}`)) {\n    url.pathname = pathname;\n  } else if (/\\/v\\d+$/.test(pathname)) {\n    url.pathname = `${pathname}/${trimmedResource}`;\n  } else {\n    url.pathname = `${pathname}/v1/${trimmedResource}`;\n  }\n\n  return url.toString();\n}\n\nexport function resolveFetch(fetchOverride?: SpeechFetch): SpeechFetch {\n  if (fetchOverride) {\n    return fetchOverride;\n  }\n\n  const globalFetch = globalThis.fetch;\n  if (!globalFetch) {\n    throw new SpeechConfigurationError(\n      'No fetch implementation available for speech adapter',\n    );\n  }\n\n  return globalFetch.bind(globalThis);\n}\n\nexport function mergeHeaders(\n  options: Pick<HttpSpeechOptions, 'apiKey' | 'headers'>,\n  extra?: HeadersInit,\n): Headers {\n  const headers = new Headers(options.headers);\n\n  if (options.apiKey && !headers.has('authorization')) {\n    headers.set('authorization', `Bearer ${options.apiKey}`);\n  }\n\n  if (extra) {\n    new Headers(extra).forEach((value, key) => {\n      headers.set(key, value);\n    });\n  }\n\n  return headers;\n}\n\nexport abstract class HttpSpeechAdapter {\n  protected readonly baseUrl: string;\n  protected readonly fetchImpl: SpeechFetch;\n  protected readonly apiKey?: string;\n  protected readonly headers?: HeadersInit;\n  protected readonly timeoutMs?: number;\n\n  constructor(options: HttpSpeechOptions) {\n    this.baseUrl = options.baseUrl;\n    this.fetchImpl = resolveFetch(options.fetch);\n    this.apiKey = options.apiKey;\n    this.headers = options.headers;\n    this.timeoutMs = options.timeoutMs;\n  }\n\n  /**\n   * POSTs to `path` (relative to `baseUrl`, or an absolute URL). The optional\n   * `retry` policy retries 429/5xx responses; `timeoutMs` applies per attempt.\n   * The request body must be re-sendable (string, Blob, FormData) when\n   * retries are enabled.\n   */\n  protected async post<T>(\n    adapterName: string,\n    path: string,\n    init: Omit<RequestInit, 'method'>,\n    readResponse: (response: Response) => Promise<T>,\n    retry: Required<SpeechRetryOptions> | false = false,\n  ): Promise<T> {\n    return withSpeechRetry(\n      () => this.postOnce(adapterName, path, init, readResponse),\n      retry,\n      init.signal ?? undefined,\n    );\n  }\n\n  private async postOnce<T>(\n    adapterName: string,\n    path: string,\n    init: Omit<RequestInit, 'method'>,\n    readResponse: (response: Response) => Promise<T>,\n  ): Promise<T> {\n    const timeoutController =\n      this.timeoutMs && this.timeoutMs > 0 ? new AbortController() : undefined;\n    const timeout =\n      timeoutController && this.timeoutMs\n        ? setTimeout(() => timeoutController.abort(), this.timeoutMs)\n        : undefined;\n    const requestSignal = composeAbortSignals(\n      init.signal,\n      timeoutController?.signal,\n    );\n\n    try {\n      // Fail fast when the caller aborted before the request was sent.\n      requestSignal.signal?.throwIfAborted();\n      const response = await this.fetchImpl(\n        resolveSpeechUrl(this.baseUrl, path),\n        {\n          ...init,\n          method: 'POST',\n          headers: mergeHeaders(\n            { apiKey: this.apiKey, headers: this.headers },\n            init.headers,\n          ),\n          signal: requestSignal.signal,\n        },\n      );\n      await assertOk(response, adapterName, this.apiKey);\n      return await readResponse(response);\n    } finally {\n      if (timeout) {\n        clearTimeout(timeout);\n      }\n      requestSignal.cleanup();\n    }\n  }\n}\n\nfunction composeAbortSignals(\n  ...signals: Array<AbortSignal | undefined | null>\n): { signal?: AbortSignal; cleanup: () => void } {\n  const activeSignals = signals.filter((signal): signal is AbortSignal =>\n    Boolean(signal),\n  );\n\n  if (activeSignals.length === 0) {\n    return { cleanup: () => undefined };\n  }\n\n  if (activeSignals.length === 1) {\n    return { signal: activeSignals[0], cleanup: () => undefined };\n  }\n\n  const controller = new AbortController();\n  let cleanup = () => undefined;\n\n  const abortFrom = (signal: AbortSignal) => {\n    if (!controller.signal.aborted) {\n      controller.abort(signal.reason);\n    }\n    cleanup();\n  };\n\n  const onAbort = (event: Event) => {\n    abortFrom(event.target as AbortSignal);\n  };\n\n  cleanup = () => {\n    for (const signal of activeSignals) {\n      signal.removeEventListener('abort', onAbort);\n    }\n  };\n\n  for (const signal of activeSignals) {\n    if (signal.aborted) {\n      abortFrom(signal);\n      return { signal: controller.signal, cleanup: () => undefined };\n    }\n  }\n\n  for (const signal of activeSignals) {\n    signal.addEventListener('abort', onAbort, { once: true });\n  }\n\n  return { signal: controller.signal, cleanup };\n}\n\nasync function assertOk(\n  response: Response,\n  adapterName: string,\n  apiKey?: string,\n): Promise<void> {\n  if (response.ok) {\n    return;\n  }\n\n  const responseBody = await response.text().catch(() => undefined);\n  throw new SpeechProviderError(\n    adapterName,\n    `${adapterName} speech request failed with HTTP ${response.status}`,\n    {\n      status: response.status,\n      responseBody: redactSecret(responseBody, apiKey),\n      retryAfterMs: parseRetryAfter(response.headers.get('retry-after')),\n    },\n  );\n}\n\n/** Removes an API key from provider-supplied text before it is surfaced. */\nexport function redactSecret(\n  text: string | undefined,\n  secret: string | undefined,\n): string | undefined {\n  if (!text || !secret) {\n    return text;\n  }\n\n  return text.split(secret).join('[REDACTED]');\n}\n\nexport function appendOptionalFormValue(\n  form: FormData,\n  name: string,\n  value: unknown,\n): void {\n  if (value === undefined || value === null) {\n    return;\n  }\n\n  form.append(name, String(value));\n}\n\nexport function createHappyVerticalSynthesisForm(\n  request: SynthesisRequest,\n  defaultVoice?: string,\n): FormData {\n  const form = new FormData();\n  const voice =\n    request.voice && typeof request.voice !== 'string'\n      ? request.voice\n      : undefined;\n  form.set('text', request.text);\n  appendOptionalFormValue(\n    form,\n    'language',\n    request.language ?? voice?.language,\n  );\n  appendOptionalFormValue(\n    form,\n    'speaker',\n    voiceToString(request.voice, defaultVoice),\n  );\n  appendOptionalFormValue(form, 'speed', request.speed);\n\n  appendOptionalFormValue(form, 'voice_prompt', voice?.prompt);\n\n  return form;\n}\n\nexport function voiceToString(\n  voice:\n    | string\n    | { id?: string; name?: string; speakerId?: string }\n    | undefined,\n  fallback?: string,\n): string | undefined {\n  if (!voice) {\n    return fallback;\n  }\n\n  if (typeof voice === 'string') {\n    return voice;\n  }\n\n  return voice.id ?? voice.name ?? voice.speakerId ?? fallback;\n}\n\nexport function compactJson(\n  value: Record<string, unknown>,\n): Record<string, unknown> {\n  return Object.fromEntries(\n    Object.entries(value).filter(([, entryValue]) => entryValue !== undefined),\n  );\n}\n\nexport function contentTypeToFormat(contentType: string): string | undefined {\n  const match = /^audio\\/([^;]+)/.exec(contentType);\n  return match?.[1];\n}\n\nexport function arrayBufferFromBytes(bytes: Uint8Array): ArrayBuffer {\n  const copy = new Uint8Array(bytes.byteLength);\n  copy.set(bytes);\n  return copy.buffer;\n}\n\nexport function decodeBase64(base64: string): ArrayBuffer {\n  if (typeof Buffer !== 'undefined') {\n    return arrayBufferFromBytes(Buffer.from(base64, 'base64'));\n  }\n\n  const binary = atob(base64);\n  const bytes = new Uint8Array(binary.length);\n  for (let index = 0; index < binary.length; index += 1) {\n    bytes[index] = binary.charCodeAt(index);\n  }\n  return bytes.buffer;\n}\n\nfunction getNumber(\n  value: Record<string, unknown>,\n  ...keys: string[]\n): number | undefined {\n  for (const key of keys) {\n    const candidate = value[key];\n    if (typeof candidate === 'number') {\n      return candidate;\n    }\n  }\n\n  return undefined;\n}\n\nfunction getString(\n  value: Record<string, unknown>,\n  ...keys: string[]\n): string | undefined {\n  for (const key of keys) {\n    const candidate = value[key];\n    if (typeof candidate === 'string') {\n      return candidate;\n    }\n  }\n\n  return undefined;\n}\n\nexport function normalizeWordTimings(value: unknown): WordTiming[] | undefined {\n  if (!Array.isArray(value)) {\n    return undefined;\n  }\n\n  return value.flatMap((entry): WordTiming[] => {\n    if (!entry || typeof entry !== 'object') {\n      return [];\n    }\n\n    const record = entry as Record<string, unknown>;\n    const word = getString(record, 'word', 'text', 'token');\n    const startSeconds = getNumber(\n      record,\n      'startSeconds',\n      'start',\n      'start_time',\n    );\n    const endSeconds = getNumber(record, 'endSeconds', 'end', 'end_time');\n\n    if (!word || startSeconds === undefined || endSeconds === undefined) {\n      return [];\n    }\n\n    return [\n      {\n        word,\n        startSeconds,\n        endSeconds,\n        confidence: getNumber(record, 'confidence'),\n        speakerId: getString(record, 'speakerId', 'speaker', 'speaker_id'),\n      },\n    ];\n  });\n}\n\nexport function normalizeTranscriptSegments(\n  value: unknown,\n): TranscriptSegment[] | undefined {\n  if (!Array.isArray(value)) {\n    return undefined;\n  }\n\n  return value.flatMap((entry): TranscriptSegment[] => {\n    if (!entry || typeof entry !== 'object') {\n      return [];\n    }\n\n    const record = entry as Record<string, unknown>;\n    const text = getString(record, 'text', 'transcript');\n    if (!text) {\n      return [];\n    }\n\n    return [\n      {\n        text,\n        startSeconds: getNumber(record, 'startSeconds', 'start', 'start_time'),\n        endSeconds: getNumber(record, 'endSeconds', 'end', 'end_time'),\n        confidence: getNumber(record, 'confidence'),\n        speakerId: getString(record, 'speakerId', 'speaker', 'speaker_id'),\n      },\n    ];\n  });\n}\n\nexport async function readTranscriptResponse(\n  response: Response,\n  adapterName: string,\n): Promise<TranscriptResult> {\n  const contentType = response.headers.get('content-type') ?? '';\n\n  if (!contentType.includes('application/json')) {\n    const text = await response.text();\n    return {\n      text,\n      provider: adapterName,\n    };\n  }\n\n  const json = (await response.json()) as Record<string, unknown>;\n  const text = getString(json, 'text', 'transcript', 'transcription') ?? '';\n\n  return {\n    text,\n    language: getString(json, 'language', 'lang'),\n    durationSeconds: getNumber(json, 'durationSeconds', 'duration'),\n    words: normalizeWordTimings(\n      json.words ?? json.wordTimings ?? json.word_timings,\n    ),\n    segments: normalizeTranscriptSegments(json.segments),\n    provider: getString(json, 'provider') ?? adapterName,\n    model: getString(json, 'model'),\n    raw: json,\n  };\n}\n\nexport async function readSynthesizedSpeechResponse(\n  response: Response,\n  adapterName: string,\n): Promise<SynthesizedSpeech> {\n  const responseContentType =\n    response.headers.get('content-type') ?? 'application/octet-stream';\n\n  if (!responseContentType.includes('application/json')) {\n    const audio = await response.arrayBuffer();\n    const sampleRate = parseHeaderNumber(response, 'x-sample-rate');\n    return {\n      audio,\n      contentType: responseContentType,\n      format: contentTypeToFormat(responseContentType),\n      sampleRate,\n      provider: adapterName,\n    };\n  }\n\n  const json = (await response.json()) as Record<string, unknown>;\n  const contentType =\n    getString(json, 'contentType', 'content_type', 'mimeType', 'mime_type') ??\n    'application/octet-stream';\n  const encodedAudio = getString(\n    json,\n    'audio',\n    'audioContent',\n    'audio_content',\n  );\n\n  if (!encodedAudio) {\n    throw new SpeechProviderError(\n      adapterName,\n      `${adapterName} JSON speech response did not include base64 audio`,\n    );\n  }\n\n  return {\n    audio: decodeBase64(encodedAudio),\n    contentType,\n    format:\n      getString(json, 'format', 'response_format') ??\n      contentTypeToFormat(contentType),\n    sampleRate: getNumber(json, 'sampleRate', 'sample_rate'),\n    channels: getNumber(json, 'channels'),\n    durationSeconds: getNumber(json, 'durationSeconds', 'duration'),\n    words: normalizeWordTimings(\n      json.words ?? json.wordTimings ?? json.word_timings,\n    ),\n    provider: getString(json, 'provider') ?? adapterName,\n    model: getString(json, 'model'),\n    raw: json,\n  };\n}\n\nfunction parseHeaderNumber(\n  response: Response,\n  headerName: string,\n): number | undefined {\n  const value = response.headers.get(headerName);\n  if (!value) {\n    return undefined;\n  }\n\n  const parsed = Number(value);\n  return Number.isFinite(parsed) ? parsed : undefined;\n}\n","/**\n * Credential handling shared by realtime (WebSocket) adapters.\n *\n * Node callers authenticate with a long-lived `apiKey` in the\n * `Authorization` header. Browsers cannot set WebSocket headers and must never\n * hold a long-lived key, so they pass a short-lived `clientSecret`, which is\n * sent as a WebSocket subprotocol.\n */\n\nimport { SpeechConfigurationError } from './errors.js';\nimport type { RealtimeConnection } from './realtime-session.js';\nimport type { StreamingClientSecret } from './streaming-types.js';\nimport { isBrowserRuntime } from './websocket.js';\n\n/** Subprotocol prefix that carries a client secret (OpenAI's convention). */\nexport const CLIENT_SECRET_SUBPROTOCOL_PREFIX = 'openai-insecure-api-key.';\n\n/**\n * Long-lived OpenAI-style keys (`sk-…`, `sk-proj-…`, `sk-svcacct-…`). A\n * browser `clientSecret` that looks like one is refused: browsers accept only\n * short-lived tokens such as OpenAI's `ek_…` client secrets.\n */\nconst LONG_LIVED_KEY = /^sk-/;\n\n/** Throws when an adapter is given a long-lived `apiKey` in a browser. */\nexport function assertNoBrowserApiKey(\n  adapter: string,\n  apiKey: string | undefined,\n): void {\n  if (apiKey && isBrowserRuntime()) {\n    throw new SpeechConfigurationError(\n      `${adapter} refuses apiKey in a browser; mint a short-lived clientSecret on your server instead`,\n      adapter,\n    );\n  }\n}\n\n/** Resolves a static or per-session client secret and validates its shape. */\nexport async function resolveClientSecret(\n  adapter: string,\n  secret: StreamingClientSecret | undefined,\n): Promise<string | undefined> {\n  const value = typeof secret === 'function' ? await secret() : secret;\n  const trimmed = value?.trim();\n  if (trimmed === undefined) {\n    return undefined;\n  }\n  if (!/^[\\x21-\\x7e]+$/.test(trimmed)) {\n    throw new SpeechConfigurationError(\n      'clientSecret must be a non-empty token without spaces',\n      adapter,\n    );\n  }\n  if (isBrowserRuntime() && LONG_LIVED_KEY.test(trimmed)) {\n    throw new SpeechConfigurationError(\n      'clientSecret looks like a long-lived API key; browsers accept only short-lived tokens minted by your server (see createStreamingClientSecret)',\n      adapter,\n    );\n  }\n  return trimmed;\n}\n\nexport interface RealtimeConnectorOptions {\n  adapter: string;\n  url: string;\n  apiKey?: string;\n  headers: Record<string, string>;\n  clientSecret?: StreamingClientSecret;\n  /** Subprotocols to offer, given the resolved client secret (if any). */\n  protocols: (secret: string | undefined) => string[];\n}\n\n/**\n * Builds the per-session `connect` callback: refuses handshake headers in a\n * browser, resolves the client secret, and sets `Authorization` from\n * `apiKey` unless a header already supplies it.\n */\nexport function realtimeConnector(\n  options: RealtimeConnectorOptions,\n): () => Promise<RealtimeConnection> {\n  if (isBrowserRuntime() && Object.keys(options.headers).length > 0) {\n    throw new SpeechConfigurationError(\n      'Browsers cannot send WebSocket headers; remove `headers` or connect from a server',\n      options.adapter,\n    );\n  }\n\n  return async () => {\n    const secret = await resolveClientSecret(\n      options.adapter,\n      options.clientSecret,\n    );\n    const headers = { ...options.headers };\n    // Gateway credentials (e.g. `x-bf-vk`, an explicit Authorization) are\n    // redacted from surfaced provider text just like the API key.\n    const secrets: string[] = headerSecrets(headers);\n    if (secret) {\n      secrets.push(secret);\n    }\n    if (options.apiKey) {\n      secrets.push(options.apiKey);\n      if (!hasHeader(headers, 'authorization')) {\n        headers.authorization = `Bearer ${options.apiKey}`;\n      }\n    }\n    return {\n      url: options.url,\n      protocols: options.protocols(secret),\n      headers,\n      secrets,\n    };\n  };\n}\n\n/** Values of other headers shorter than this are not treated as credentials. */\nconst MIN_HEADER_SECRET_LENGTH = 8;\n\n/**\n * Header names that carry credentials (`authorization`, `x-api-key`,\n * `x-bf-vk`, `*-token`, `*-secret`, …). Their values are redacted at any\n * length.\n */\nconst CREDENTIAL_HEADER = /auth|key|token|secret|password|credential|-vk$/i;\n\n/**\n * Values of caller-supplied headers to redact from provider error text.\n * Credential-named headers are redacted whatever their length; for\n * `Authorization`, the credential after the scheme is included too. Other\n * headers may also carry tenant secrets, so their values are redacted when\n * they are at least 8 characters long, which keeps short values such as\n * `1` from mangling messages.\n */\nexport function headerSecrets(headers: HeadersInit | undefined): string[] {\n  const secrets: string[] = [];\n  for (const [key, value] of Object.entries(mergeHeaderRecords(headers))) {\n    const credential = CREDENTIAL_HEADER.test(key);\n    const candidates = [value];\n    if (key.toLowerCase() === 'authorization') {\n      candidates.push(value.replace(/^\\S+\\s+/, ''));\n    }\n    for (const candidate of candidates) {\n      const trimmed = candidate.trim();\n      const long = trimmed.length >= MIN_HEADER_SECRET_LENGTH;\n      if (trimmed && (credential || long) && !secrets.includes(trimmed)) {\n        secrets.push(trimmed);\n      }\n    }\n  }\n  // Longest first, so a full `Bearer <token>` is replaced before its token.\n  return secrets.sort((left, right) => right.length - left.length);\n}\n\nexport function mergeHeaderRecords(\n  ...sets: Array<HeadersInit | undefined>\n): Record<string, string> {\n  const merged: Record<string, string> = {};\n  for (const set of sets) {\n    if (set) {\n      new Headers(set).forEach((value, key) => {\n        merged[key] = value;\n      });\n    }\n  }\n  return merged;\n}\n\nfunction hasHeader(headers: Record<string, string>, name: string): boolean {\n  return Object.keys(headers).some((key) => key.toLowerCase() === name);\n}\n","/**\n * Raw-audio helpers for streaming adapters: byte views, base64 encoding,\n * duration from byte counts, and WAV/`audio/L16` unwrapping for the\n * record-then-send wrapper. Nothing here decodes compressed audio or resamples.\n */\n\nimport { normalizeMimeType } from './audio.js';\nimport { SpeechConfigurationError } from './errors.js';\nimport type {\n  StreamingAudioChunk,\n  StreamingAudioEncoding,\n  StreamingAudioFormat,\n} from './streaming-types.js';\n\nconst ENCODING_BYTES_PER_SAMPLE: Record<StreamingAudioEncoding, number> = {\n  pcm16: 2,\n  g711_ulaw: 1,\n  g711_alaw: 1,\n};\n\n/** Returns a byte view over a chunk without copying. */\nexport function chunkToBytes(\n  chunk: StreamingAudioChunk,\n  adapter?: string,\n): Uint8Array {\n  if (chunk instanceof Uint8Array) {\n    return chunk;\n  }\n  if (chunk instanceof ArrayBuffer) {\n    return new Uint8Array(chunk);\n  }\n  if (ArrayBuffer.isView(chunk)) {\n    return new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength);\n  }\n  throw new SpeechConfigurationError(\n    'Streaming audio chunks must be Uint8Array, ArrayBuffer, or an ArrayBuffer view',\n    adapter,\n  );\n}\n\nexport function encodeBase64(bytes: Uint8Array): string {\n  if (typeof Buffer !== 'undefined') {\n    return Buffer.from(\n      bytes.buffer,\n      bytes.byteOffset,\n      bytes.byteLength,\n    ).toString('base64');\n  }\n\n  let binary = '';\n  const step = 0x8000;\n  for (let index = 0; index < bytes.length; index += step) {\n    binary += String.fromCharCode(...bytes.subarray(index, index + step));\n  }\n  return btoa(binary);\n}\n\n/** Bytes per second of audio in `format`. */\nexport function bytesPerSecond(format: StreamingAudioFormat): number {\n  return (\n    format.sampleRate *\n    format.channels *\n    ENCODING_BYTES_PER_SAMPLE[format.encoding]\n  );\n}\n\n/** Audio duration in seconds for `bytes` of `format`, rounded to milliseconds. */\nexport function audioSecondsForBytes(\n  bytes: number,\n  format: StreamingAudioFormat,\n): number {\n  return Math.round((bytes / bytesPerSecond(format)) * 1000) / 1000;\n}\n\nexport interface RawAudio {\n  bytes: Uint8Array;\n  /** Format fields the input declared (WAV header, MIME parameters, or `AudioInput`). */\n  format: Partial<StreamingAudioFormat>;\n}\n\nconst WAV_TYPES = new Set([\n  'audio/wav',\n  'audio/wave',\n  'audio/x-wav',\n  'audio/vnd.wave',\n]);\nconst RAW_PCM_TYPES = new Set([\n  'audio/pcm',\n  'audio/l16',\n  'audio/raw',\n  'audio/x-raw',\n]);\n/** Untyped bytes: the streaming transcriber's default format applies. */\nconst UNTYPED = 'application/octet-stream';\nconst MULAW_TYPES = new Set(['audio/pcmu', 'audio/basic', 'audio/x-mulaw']);\nconst ALAW_TYPES = new Set(['audio/pcma', 'audio/x-alaw']);\n\n/**\n * Interprets a finished recording as raw audio for a streaming session.\n * Accepts PCM/G.711 WAV files, `audio/pcm`/`audio/L16` (with optional `rate`\n * and `channels` parameters), `audio/pcmu`/`audio/pcma`, and untyped bytes.\n * Compressed containers are rejected with a `SpeechConfigurationError`.\n */\nexport function unwrapRawAudio(\n  bytes: Uint8Array,\n  mimeType: string,\n  adapter: string,\n): RawAudio {\n  const essence = normalizeMimeType(mimeType) ?? 'application/octet-stream';\n  const params = parseMimeParameters(mimeType);\n  const rate = parsePositiveInt(params.rate);\n  const channels = parsePositiveInt(params.channels);\n\n  if (WAV_TYPES.has(essence) || isRiffWave(bytes)) {\n    return parseWav(bytes, adapter);\n  }\n  if (essence === UNTYPED) {\n    // No encoding: `start()` keeps the adapter's `audioFormat` (e.g. G.711).\n    return { bytes, format: compact({ sampleRate: rate, channels }) };\n  }\n  if (RAW_PCM_TYPES.has(essence)) {\n    if (essence === 'audio/l16' && bytes.byteLength % 2 !== 0) {\n      throw new SpeechConfigurationError(\n        `audio/L16 payload has an odd byte length (${bytes.byteLength}); 16-bit samples need an even number of bytes`,\n        adapter,\n      );\n    }\n    return {\n      // RFC 2586: audio/L16 samples are big-endian; pcm16 is little-endian.\n      bytes: essence === 'audio/l16' ? swapBytePairs(bytes) : bytes,\n      format: compact({ encoding: 'pcm16', sampleRate: rate, channels }),\n    };\n  }\n  if (MULAW_TYPES.has(essence)) {\n    return {\n      bytes,\n      format: compact({ encoding: 'g711_ulaw', sampleRate: rate, channels }),\n    };\n  }\n  if (ALAW_TYPES.has(essence)) {\n    return {\n      bytes,\n      format: compact({ encoding: 'g711_alaw', sampleRate: rate, channels }),\n    };\n  }\n\n  throw new SpeechConfigurationError(\n    `${adapter} streams raw audio and cannot decode ${essence}; send PCM WAV or raw PCM, or use an HTTP transcriber for compressed recordings`,\n    adapter,\n  );\n}\n\n/** Returns a copy with each 16-bit sample's bytes swapped. */\nfunction swapBytePairs(bytes: Uint8Array): Uint8Array {\n  const swapped = new Uint8Array(bytes.byteLength);\n  for (let index = 0; index + 1 < bytes.byteLength; index += 2) {\n    swapped[index] = bytes[index + 1] ?? 0;\n    swapped[index + 1] = bytes[index] ?? 0;\n  }\n  return swapped;\n}\n\nfunction isRiffWave(bytes: Uint8Array): boolean {\n  return (\n    bytes.byteLength >= 12 &&\n    ascii(bytes, 0, 4) === 'RIFF' &&\n    ascii(bytes, 8, 4) === 'WAVE'\n  );\n}\n\nfunction parseWav(bytes: Uint8Array, adapter: string): RawAudio {\n  if (!isRiffWave(bytes)) {\n    throw new SpeechConfigurationError('Invalid WAV header', adapter);\n  }\n\n  const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);\n  let offset = 12;\n  let format: Partial<StreamingAudioFormat> | undefined;\n\n  while (offset + 8 <= bytes.byteLength) {\n    const id = ascii(bytes, offset, 4);\n    const size = view.getUint32(offset + 4, true);\n    const body = offset + 8;\n\n    if (id === 'fmt ' && body + 16 <= bytes.byteLength) {\n      const audioFormat = view.getUint16(body, true);\n      const channels = view.getUint16(body + 2, true);\n      const sampleRate = view.getUint32(body + 4, true);\n      const bitsPerSample = view.getUint16(body + 14, true);\n      format = {\n        encoding: wavEncoding(audioFormat, bitsPerSample, adapter),\n        sampleRate,\n        channels,\n      };\n    } else if (id === 'data') {\n      if (!format) {\n        throw new SpeechConfigurationError(\n          'WAV data chunk precedes its fmt chunk',\n          adapter,\n        );\n      }\n      const end = Math.min(bytes.byteLength, body + size);\n      return { bytes: bytes.subarray(body, end), format };\n    }\n\n    // Chunks are padded to an even size.\n    offset = body + size + (size % 2);\n  }\n\n  throw new SpeechConfigurationError('WAV file has no data chunk', adapter);\n}\n\nfunction wavEncoding(\n  audioFormat: number,\n  bitsPerSample: number,\n  adapter: string,\n): StreamingAudioEncoding {\n  // 0xfffe is WAVE_FORMAT_EXTENSIBLE; its common PCM case is treated as PCM.\n  if ((audioFormat === 1 || audioFormat === 0xfffe) && bitsPerSample === 16) {\n    return 'pcm16';\n  }\n  if (audioFormat === 7 && bitsPerSample === 8) {\n    return 'g711_ulaw';\n  }\n  if (audioFormat === 6 && bitsPerSample === 8) {\n    return 'g711_alaw';\n  }\n  throw new SpeechConfigurationError(\n    `Unsupported WAV encoding (format ${audioFormat}, ${bitsPerSample}-bit); expected 16-bit PCM or 8-bit G.711`,\n    adapter,\n  );\n}\n\nfunction ascii(bytes: Uint8Array, offset: number, length: number): string {\n  return String.fromCharCode(...bytes.subarray(offset, offset + length));\n}\n\nfunction parseMimeParameters(mimeType: string): Record<string, string> {\n  const params: Record<string, string> = {};\n  for (const part of mimeType.split(';').slice(1)) {\n    const [key, value] = part.split('=');\n    if (key && value) {\n      params[key.trim().toLowerCase()] = value.trim();\n    }\n  }\n  return params;\n}\n\nfunction parsePositiveInt(value: string | undefined): number | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const parsed = Number.parseInt(value, 10);\n  return Number.isFinite(parsed) && parsed > 0 ? parsed : undefined;\n}\n\nfunction compact(\n  format: Partial<StreamingAudioFormat>,\n): Partial<StreamingAudioFormat> {\n  return Object.fromEntries(\n    Object.entries(format).filter(([, value]) => value !== undefined),\n  ) as Partial<StreamingAudioFormat>;\n}\n","/**\n * Per-turn audio limits for realtime sessions.\n *\n * Some providers cap the audio of one turn rather than of a session: vLLM's\n * realtime endpoint keeps a turn's audio and generated tokens in one model\n * context (`max_model_len`) and clears it after each final commit. A\n * {@link TurnSplitter} measures the audio of the current turn from byte counts\n * and the negotiated format, and picks where to roll over to a new turn: at\n * the first quiet stretch inside a window before the limit, or at the limit\n * itself when no quiet stretch turns up. The silence check is a plain RMS\n * energy test, not a voice-activity model.\n */\n\nimport { SpeechConfigurationError } from './errors.js';\nimport { bytesPerSecond } from './pcm.js';\nimport type {\n  StreamingAudioEncoding,\n  StreamingAudioFormat,\n  StreamingRolloverOptions,\n  StreamingSessionSettings,\n  StreamingTurnLimit,\n} from './streaming-types.js';\n\n/** Default per-turn audio cap for providers that need one (`voxtral-realtime`). */\nexport const DEFAULT_MAX_TURN_SECONDS = 270;\n/** How long before the cap the rollover starts looking for silence. */\nexport const DEFAULT_ROLLOVER_WINDOW_SECONDS = 15;\n/** RMS level (0-1 of full scale, about -40 dBFS) below which audio is quiet. */\nexport const DEFAULT_ROLLOVER_SILENCE_THRESHOLD = 0.01;\n/** Quiet audio needed before the rollover commits. */\nexport const DEFAULT_ROLLOVER_MIN_SILENCE_MS = 300;\n/** Energy is measured over frames of this length. */\nconst ANALYSIS_FRAME_MS = 20;\n\nconst BYTES_PER_SAMPLE: Record<StreamingAudioEncoding, number> = {\n  pcm16: 2,\n  g711_ulaw: 1,\n  g711_alaw: 1,\n};\n\n/** A resolved per-turn limit; `undefined` means turns are unbounded. */\nexport interface TurnLimit {\n  maxTurnSeconds: number;\n  /** `false`: writes past the cap reject with `SPEECH_TURN_TOO_LONG`. */\n  rollover: boolean;\n  windowSeconds: number;\n  silenceThreshold: number;\n  minSilenceMs: number;\n}\n\n/**\n * Resolves the turn limit from settings, highest priority first (session\n * options, then adapter options, which already carry environment values),\n * falling back to `defaultMaxTurnSeconds`. Returns `undefined` when no limit\n * applies or `maxTurnSeconds` is `Infinity`.\n */\nexport function resolveTurnLimit(\n  adapter: string,\n  defaultMaxTurnSeconds: number | undefined,\n  ...settings: Array<\n    Pick<StreamingSessionSettings, 'maxTurnSeconds' | 'rollover'> | undefined\n  >\n): TurnLimit | undefined {\n  const maxTurnSeconds =\n    settings.find((entry) => entry?.maxTurnSeconds !== undefined)\n      ?.maxTurnSeconds ?? defaultMaxTurnSeconds;\n  if (maxTurnSeconds === undefined) {\n    return undefined;\n  }\n  if (\n    typeof maxTurnSeconds !== 'number' ||\n    Number.isNaN(maxTurnSeconds) ||\n    maxTurnSeconds <= 0\n  ) {\n    throw new SpeechConfigurationError(\n      `maxTurnSeconds must be a positive number of seconds or Infinity (got ${String(maxTurnSeconds)})`,\n      adapter,\n    );\n  }\n  if (maxTurnSeconds === Number.POSITIVE_INFINITY) {\n    return undefined;\n  }\n\n  // On/off comes from the first layer that sets `rollover`; each tuning\n  // field from the first layer whose `rollover` object sets that field, so a\n  // `rollover: true` session keeps the adapter's tuning.\n  const rollover = settings.find(\n    (entry) => entry?.rollover !== undefined,\n  )?.rollover;\n  const tuning = <K extends keyof StreamingRolloverOptions>(key: K) =>\n    settings\n      .map((entry) => entry?.rollover)\n      .find(\n        (value): value is StreamingRolloverOptions =>\n          typeof value === 'object' &&\n          value !== null &&\n          value[key] !== undefined,\n      )?.[key];\n  const limit: TurnLimit = {\n    maxTurnSeconds,\n    rollover: rollover !== false,\n    windowSeconds: tuning('windowSeconds') ?? DEFAULT_ROLLOVER_WINDOW_SECONDS,\n    silenceThreshold:\n      tuning('silenceThreshold') ?? DEFAULT_ROLLOVER_SILENCE_THRESHOLD,\n    minSilenceMs: tuning('minSilenceMs') ?? DEFAULT_ROLLOVER_MIN_SILENCE_MS,\n  };\n  assertNonNegative(adapter, 'rollover.windowSeconds', limit.windowSeconds);\n  assertNonNegative(adapter, 'rollover.minSilenceMs', limit.minSilenceMs);\n  assertNonNegative(\n    adapter,\n    'rollover.silenceThreshold',\n    limit.silenceThreshold,\n  );\n  if (limit.silenceThreshold > 1) {\n    throw new SpeechConfigurationError(\n      `rollover.silenceThreshold is an RMS level from 0 to 1 (got ${limit.silenceThreshold})`,\n      adapter,\n    );\n  }\n  return limit;\n}\n\nfunction assertNonNegative(adapter: string, name: string, value: unknown) {\n  if (typeof value !== 'number' || !Number.isFinite(value) || value < 0) {\n    throw new SpeechConfigurationError(\n      `${name} must be a non-negative finite number (got ${String(value)})`,\n      adapter,\n    );\n  }\n}\n\n/** Bytes of one sample frame (one sample per channel). */\nexport function blockAlign(format: StreamingAudioFormat): number {\n  return BYTES_PER_SAMPLE[format.encoding] * format.channels;\n}\n\n/** Whole sample frames in `seconds` of audio, as bytes (rounded down). */\nexport function bytesForSeconds(\n  seconds: number,\n  format: StreamingAudioFormat,\n): number {\n  const align = blockAlign(format);\n  // The epsilon keeps decimal seconds such as 0.3 from losing a frame.\n  return Math.floor((seconds * bytesPerSecond(format)) / align + 1e-9) * align;\n}\n\n/** Digital silence in `format`: zero samples, encoded. */\nexport function silence(\n  bytes: number,\n  format: StreamingAudioFormat,\n): Uint8Array {\n  const fill =\n    format.encoding === 'g711_ulaw'\n      ? 0xff\n      : format.encoding === 'g711_alaw'\n        ? 0xd5\n        : 0;\n  return new Uint8Array(bytes).fill(fill);\n}\n\n/** Bytes of audio one turn may hold: whole sample frames, at least one. */\nexport function turnCapBytes(\n  limit: TurnLimit,\n  format: StreamingAudioFormat,\n): number {\n  return Math.max(\n    blockAlign(format),\n    bytesForSeconds(limit.maxTurnSeconds, format),\n  );\n}\n\n/** The public view of a resolved limit for a session format. */\nexport function describeTurnLimit(\n  limit: TurnLimit | undefined,\n  format: StreamingAudioFormat,\n): StreamingTurnLimit | undefined {\n  return (\n    limit && {\n      maxTurnSeconds: limit.maxTurnSeconds,\n      rollover: limit.rollover,\n      maxTurnBytes: turnCapBytes(limit, format),\n    }\n  );\n}\n\n/**\n * Tracks the audio of the current turn and finds rollover points.\n *\n * Feed every chunk in write order to {@link split} (rollover on) or\n * {@link add} (rollover off), and call {@link reset} whenever the turn ends\n * for another reason (a manual commit).\n */\nexport class TurnSplitter {\n  /** Bytes of the current turn so far. */\n  turnBytes = 0;\n  readonly capBytes: number;\n  private readonly windowStartBytes: number;\n  private readonly frameBytes: number;\n  private readonly minSilenceBytes: number;\n  private readonly threshold: number;\n  private readonly encoding: StreamingAudioEncoding;\n  private readonly sampleBytes: number;\n  // Energy of the analysis frame being filled.\n  private frameFill = 0;\n  private frameSumSquares = 0;\n  private frameSamples = 0;\n  /** Trailing bytes of a sample split across chunks. */\n  private carry: number[] = [];\n  /** Consecutive quiet bytes ending at the last complete frame. */\n  private quietRun = 0;\n\n  constructor(limit: TurnLimit, format: StreamingAudioFormat) {\n    const align = blockAlign(format);\n    this.capBytes = turnCapBytes(limit, format);\n    this.windowStartBytes = Math.max(\n      0,\n      this.capBytes - bytesForSeconds(limit.windowSeconds, format),\n    );\n    this.frameBytes = Math.max(\n      align,\n      bytesForSeconds(ANALYSIS_FRAME_MS / 1000, format),\n    );\n    this.minSilenceBytes = Math.max(\n      this.frameBytes,\n      bytesForSeconds(limit.minSilenceMs / 1000, format),\n    );\n    this.threshold = limit.silenceThreshold;\n    this.encoding = format.encoding;\n    this.sampleBytes = BYTES_PER_SAMPLE[format.encoding];\n  }\n\n  /** Ends the current turn: the next byte starts a new one. */\n  reset(): void {\n    this.turnBytes = 0;\n    this.frameFill = 0;\n    this.frameSumSquares = 0;\n    this.frameSamples = 0;\n    this.carry = [];\n    this.quietRun = 0;\n  }\n\n  /** Whether `bytes` more would push the current turn past the cap. */\n  wouldOverflow(bytes: number): boolean {\n    return this.turnBytes + bytes > this.capBytes;\n  }\n\n  /** Counts `bytes` toward the current turn without analysing them. */\n  add(bytes: number): void {\n    this.turnBytes += bytes;\n  }\n\n  /**\n   * Feeds one chunk and returns the offsets inside it where the turn should\n   * end (each in `1..chunk.byteLength`, ascending). The splitter resets at\n   * each offset, so bytes after it count toward the next turn.\n   */\n  split(chunk: Uint8Array): number[] {\n    const cuts: number[] = [];\n    let offset = 0;\n    while (offset < chunk.byteLength) {\n      if (this.turnBytes < this.windowStartBytes) {\n        // Before the window only the length matters.\n        const skip = Math.min(\n          chunk.byteLength - offset,\n          this.windowStartBytes - this.turnBytes,\n        );\n        offset += skip;\n        this.turnBytes += skip;\n        // With no window (`windowSeconds: 0`) the window starts at the cap.\n        if (this.turnBytes >= this.capBytes) {\n          cuts.push(offset);\n          this.reset();\n        }\n        continue;\n      }\n\n      const take = Math.min(\n        chunk.byteLength - offset,\n        this.frameBytes - this.frameFill,\n        this.capBytes - this.turnBytes,\n      );\n      this.measure(chunk, offset, take);\n      offset += take;\n      this.turnBytes += take;\n      this.frameFill += take;\n\n      let cut = false;\n      if (this.frameFill >= this.frameBytes) {\n        const quiet = this.frameIsQuiet();\n        this.quietRun = quiet ? this.quietRun + this.frameFill : 0;\n        this.frameFill = 0;\n        this.frameSumSquares = 0;\n        this.frameSamples = 0;\n        cut = this.quietRun >= this.minSilenceBytes;\n      }\n      if (cut || this.turnBytes >= this.capBytes) {\n        cuts.push(offset);\n        this.reset();\n      }\n    }\n    return cuts;\n  }\n\n  private frameIsQuiet(): boolean {\n    if (this.frameSamples === 0) {\n      return false;\n    }\n    const rms = Math.sqrt(this.frameSumSquares / this.frameSamples) / 32_768;\n    return rms < this.threshold;\n  }\n\n  private measure(chunk: Uint8Array, start: number, length: number): void {\n    const end = start + length;\n    let index = start;\n    // Complete a sample left over from the previous chunk.\n    while (this.carry.length > 0 && index < end) {\n      this.carry.push(chunk[index] ?? 0);\n      index += 1;\n      if (this.carry.length === this.sampleBytes) {\n        this.addSample(this.carry);\n        this.carry = [];\n      }\n    }\n    for (; index + this.sampleBytes <= end; index += this.sampleBytes) {\n      this.addSampleAt(chunk, index);\n    }\n    for (; index < end; index += 1) {\n      this.carry.push(chunk[index] ?? 0);\n    }\n  }\n\n  private addSample(bytes: number[]): void {\n    this.addLinear(\n      this.encoding === 'pcm16'\n        ? pcm16Sample(bytes[0] ?? 0, bytes[1] ?? 0)\n        : this.decodeG711(bytes[0] ?? 0),\n    );\n  }\n\n  private addSampleAt(chunk: Uint8Array, index: number): void {\n    this.addLinear(\n      this.encoding === 'pcm16'\n        ? pcm16Sample(chunk[index] ?? 0, chunk[index + 1] ?? 0)\n        : this.decodeG711(chunk[index] ?? 0),\n    );\n  }\n\n  private addLinear(sample: number): void {\n    this.frameSumSquares += sample * sample;\n    this.frameSamples += 1;\n  }\n\n  private decodeG711(byte: number): number {\n    return this.encoding === 'g711_ulaw' ? decodeMuLaw(byte) : decodeALaw(byte);\n  }\n}\n\n/** Signed 16-bit little-endian sample. */\nfunction pcm16Sample(low: number, high: number): number {\n  const value = low | (high << 8);\n  return value >= 0x8000 ? value - 0x10000 : value;\n}\n\n/** G.711 mu-law byte to a 16-bit linear sample (ITU-T G.711). */\nexport function decodeMuLaw(byte: number): number {\n  const value = ~byte & 0xff;\n  const exponent = (value >> 4) & 0x07;\n  const mantissa = value & 0x0f;\n  const magnitude = (((mantissa << 3) + 0x84) << exponent) - 0x84;\n  return value & 0x80 ? -magnitude : magnitude;\n}\n\n/** G.711 A-law byte to a 16-bit linear sample (ITU-T G.711). */\nexport function decodeALaw(byte: number): number {\n  const value = byte ^ 0x55;\n  const exponent = (value >> 4) & 0x07;\n  const mantissa = value & 0x0f;\n  const magnitude =\n    exponent === 0\n      ? (mantissa << 4) + 8\n      : ((mantissa << 4) + 0x108) << (exponent - 1);\n  return value & 0x80 ? magnitude : -magnitude;\n}\n","/**\n * Protocol-agnostic realtime transcription session over a WebSocket.\n *\n * The session owns the socket lifecycle, write queue and backpressure,\n * timeouts, turn bookkeeping, events, and the final `TranscriptResult`. A\n * {@link RealtimeProtocol} supplies the wire format, so a sibling adapter\n * (for example a vLLM-served Voxtral realtime endpoint) only has to map its\n * own messages.\n */\n\nimport {\n  SpeechConfigurationError,\n  SpeechError,\n  SpeechProviderError,\n} from './errors.js';\nimport { redactSecret } from './http.js';\nimport {\n  audioSecondsForBytes,\n  bytesPerSecond,\n  chunkToBytes,\n  encodeBase64,\n} from './pcm.js';\nimport type {\n  StreamingAudioChunk,\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionEventName,\n  StreamingSessionEvents,\n  StreamingSessionListener,\n  StreamingSessionState,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport {\n  blockAlign,\n  silence,\n  type TurnLimit,\n  TurnSplitter,\n} from './turn-limit.js';\nimport type { TranscriptResult, TranscriptSegment } from './types.js';\nimport {\n  audioSecondsFromProviderUsage,\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from './usage.js';\nimport {\n  messageDataToText,\n  type SpeechWebSocket,\n  type SpeechWebSocketCloseEvent,\n  type SpeechWebSocketFactory,\n} from './websocket.js';\n\nexport const DEFAULT_STREAMING_TIMEOUT_MS = 30_000;\nexport const DEFAULT_STREAMING_CONNECT_TIMEOUT_MS = 10_000;\nexport const DEFAULT_STREAMING_HIGH_WATER_MARK = 1024 * 1024;\nexport const DEFAULT_STREAMING_MAX_BUFFERED_BYTES = 16 * 1024 * 1024;\nconst DRAIN_POLL_MS = 10;\n\n/** Normalised events a protocol parser returns for one server message. */\nexport type RealtimeProtocolEvent =\n  | { type: 'session'; sessionId?: string }\n  /** The provider committed a turn (from a client commit or server VAD). */\n  | { type: 'committed'; itemId?: string }\n  /** The provider rejected a commit because its buffer was empty. Not fatal. */\n  | { type: 'commit_empty' }\n  | { type: 'partial'; itemId?: string; delta: string }\n  | {\n      type: 'final';\n      itemId?: string;\n      text: string;\n      providerUsage?: Record<string, unknown>;\n      raw?: unknown;\n      /**\n       * `false` when the provider reports that the turn consumed no audio\n       * (vLLM: `usage.prompt_tokens <= 1`). A turn that was sent audio but\n       * consumed none means the provider's turn state is out of sync, and the\n       * session fails instead of silently dropping that audio.\n       */\n      audioConsumed?: boolean;\n    }\n  | { type: 'speech_started'; itemId?: string; audioMs?: number }\n  | { type: 'speech_stopped'; itemId?: string; audioMs?: number }\n  /** A fatal provider error. */\n  | { type: 'error'; message: string; code?: string; raw?: unknown };\n\n/** Resolved per-session settings handed to the protocol. */\nexport interface RealtimeSessionConfig {\n  model: string;\n  format: StreamingAudioFormat;\n  turnDetection: StreamingTurnDetection;\n  language?: string;\n  prompt?: string;\n}\n\n/** Wire format of one realtime provider. */\nexport interface RealtimeProtocol {\n  /** Adapter name used for errors and usage, e.g. `openai-realtime`. */\n  readonly provider: string;\n  /**\n   * How a client commit is acknowledged: by a `committed` event (OpenAI), or\n   * only by the turn's `final` event (servers without a commit ack).\n   */\n  readonly commitAck: 'committed' | 'final';\n  /**\n   * Whether `end()` commits only when uncommitted audio remains\n   * (`if-audio`), or always sends a final commit (`always`, for servers that\n   * need an explicit end-of-stream marker).\n   */\n  readonly endCommit: 'if-audio' | 'always';\n  /**\n   * Default per-turn audio cap in seconds, for providers whose model context\n   * covers one turn (vLLM resets it after each final commit). Adapters pass\n   * it to `resolveTurnLimit()`; a session with a turn limit rolls over to a\n   * new turn before the cap, or rejects writes past it. Unset: no default.\n   */\n  readonly maxTurnSeconds?: number;\n  /**\n   * Shortest audio, in seconds, the provider accepts in one commit (OpenAI:\n   * 100 ms; shorter commits are rejected as empty). A session with a turn\n   * limit pads a shorter, non-empty turn with silence before committing it,\n   * so the tail left after a rollover is transcribed instead of dropped.\n   */\n  readonly minCommitSeconds?: number;\n  /** Messages sent once the socket opens (session configuration). */\n  sessionMessages(config: RealtimeSessionConfig): unknown[];\n  /** One audio chunk, already base64-encoded. */\n  appendMessage(base64Audio: string): unknown;\n  /** Ends the current turn. `final` is true for the commit sent by `end()`. */\n  commitMessage(options: { final: boolean }): unknown;\n  /**\n   * Opens a turn before its first audio, for servers that only start\n   * generating after an explicit start signal (vLLM's non-final\n   * `input_audio_buffer.commit`). When set, the session sends it before the\n   * first append of every turn, and holds audio written after a commit until\n   * every outstanding turn has been acknowledged, so a server that resets its\n   * buffer between turns never drops audio.\n   */\n  turnStartMessage?(): unknown;\n  /** Maps one parsed JSON server message to zero or more events. */\n  parse(message: unknown): RealtimeProtocolEvent[];\n}\n\n/** Where and how to open the socket; resolved per session. */\nexport interface RealtimeConnection {\n  url: string;\n  protocols: string[];\n  headers: Record<string, string>;\n  /** Credentials to redact from surfaced provider text. */\n  secrets: string[];\n}\n\nexport interface RealtimeSessionInit {\n  protocol: RealtimeProtocol;\n  config: RealtimeSessionConfig;\n  /** Resolves the connection (may mint a short-lived token). */\n  connect: () => Promise<RealtimeConnection>;\n  createWebSocket: SpeechWebSocketFactory;\n  timeoutMs: number;\n  connectTimeoutMs: number;\n  highWaterMark: number;\n  maxBufferedBytes: number;\n  signal?: AbortSignal;\n  /** Called in order with the aggregated usage when `end()` succeeds. */\n  onUsage: Array<SpeechUsageCallback | undefined>;\n  /** Per-turn audio limit; unset means turns are unbounded. */\n  turnLimit?: TurnLimit;\n}\n\ntype QueueItem =\n  | {\n      kind: 'audio';\n      base64: string;\n      bytes: number;\n      resolve: () => void;\n      reject: (error: Error) => void;\n    }\n  | { kind: 'commit'; final: boolean };\n\ninterface FinalRecord {\n  itemId?: string;\n  text: string;\n  providerUsage?: Record<string, unknown>;\n  raw?: unknown;\n  arrival: number;\n}\n\nconst noop = () => undefined;\n\nexport class RealtimeTranscriptionSession implements StreamingSession {\n  state: StreamingSessionState = 'connecting';\n  sessionId?: string;\n  readonly ready: Promise<void>;\n\n  private readonly init: RealtimeSessionInit;\n  private readonly provider: string;\n  private resolveReady: () => void = noop;\n  private rejectReady: (error: Error) => void = noop;\n  private socket?: SpeechWebSocket;\n  private opened = false;\n  private secrets: string[] = [];\n  private queue: QueueItem[] = [];\n  private queued = 0;\n  private pumping = false;\n  private totalBytes = 0;\n  private uncommittedBytes = 0;\n  private outstandingCommits = 0;\n  /** Per outstanding commit, in order: whether it carried audio. */\n  private readonly commitHadAudio: boolean[] = [];\n  private turnOpen = false;\n  /** `Date.now()` of the last server message; bounds inactivity waits. */\n  private lastServerMessageAt = 0;\n  private readonly itemOrder: string[] = [];\n  private readonly pendingItems = new Set<string>();\n  private readonly vadStoppedItems = new Set<string>();\n  private readonly partials = new Map<string, string>();\n  private readonly finals: FinalRecord[] = [];\n  private readonly listeners = new Map<\n    StreamingSessionEventName,\n    Set<(event: never) => void>\n  >();\n  private error?: Error;\n  private endPromise?: Promise<TranscriptResult>;\n  private endWaiter?: { resolve: () => void; reject: (error: Error) => void };\n  private connectTimer?: ReturnType<typeof setTimeout>;\n  private endTimer?: ReturnType<typeof setTimeout>;\n  private closeEmitted = false;\n  /** Measures the current turn in write order when a turn limit applies. */\n  private readonly splitter?: TurnSplitter;\n  private readonly onAbort = () => {\n    this.fail(toError(this.init.signal?.reason, this.provider));\n  };\n\n  constructor(init: RealtimeSessionInit) {\n    this.init = init;\n    this.provider = init.protocol.provider;\n    assertTurnLimitFitsProtocol(init.protocol, init.turnLimit);\n    if (init.turnLimit) {\n      this.splitter = new TurnSplitter(init.turnLimit, init.config.format);\n    }\n    this.ready = new Promise<void>((resolve, reject) => {\n      this.resolveReady = resolve;\n      this.rejectReady = reject;\n    });\n    this.ready.catch(noop);\n\n    if (init.signal?.aborted) {\n      this.fail(toError(init.signal.reason, this.provider));\n      return;\n    }\n    init.signal?.addEventListener('abort', this.onAbort, { once: true });\n    void this.connect();\n  }\n\n  get queuedBytes(): number {\n    return this.queued;\n  }\n\n  write(chunk: StreamingAudioChunk): Promise<void> {\n    const promise = this.enqueueAudio(chunk);\n    promise.catch(noop);\n    return promise;\n  }\n\n  commit(): void {\n    if (this.isTerminal() || this.endPromise) {\n      return;\n    }\n    this.queue.push({ kind: 'commit', final: false });\n    this.splitter?.reset();\n    void this.pump();\n  }\n\n  end(): Promise<TranscriptResult> {\n    if (this.endPromise) {\n      return this.endPromise;\n    }\n\n    if (this.error) {\n      this.endPromise = Promise.reject(this.error);\n      return this.endPromise;\n    }\n\n    this.state = 'ending';\n    this.queue.push({ kind: 'commit', final: true });\n    const settled = new Promise<void>((resolve, reject) => {\n      this.endWaiter = { resolve, reject };\n    });\n    this.armEndTimer();\n    this.endPromise = settled.then(() => this.finish());\n    void this.pump();\n    this.checkEnded();\n    return this.endPromise;\n  }\n\n  /**\n   * (Re)starts the end timeout. It measures provider inactivity: every server\n   * message restarts it, so a long recording that is still being transcribed\n   * is not cut off, while a silent provider still fails after `timeoutMs`.\n   */\n  private armEndTimer(): void {\n    clearTimeout(this.endTimer);\n    this.endTimer = setTimeout(() => {\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} timed out after ${this.init.timeoutMs} ms waiting for the final transcript`,\n        ),\n      );\n    }, this.init.timeoutMs);\n  }\n\n  abort(reason?: unknown): void {\n    this.fail(toError(reason, this.provider));\n  }\n\n  on<E extends StreamingSessionEventName>(\n    event: E,\n    listener: StreamingSessionListener<E>,\n  ): () => void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as (event: never) => void);\n    return () => this.off(event, listener);\n  }\n\n  off<E extends StreamingSessionEventName>(\n    event: E,\n    listener: StreamingSessionListener<E>,\n  ): void {\n    this.listeners.get(event)?.delete(listener as (event: never) => void);\n  }\n\n  private enqueueAudio(chunk: StreamingAudioChunk): Promise<void> {\n    if (this.error) {\n      return Promise.reject(this.error);\n    }\n    if (this.endPromise || this.isTerminal()) {\n      return Promise.reject(\n        new SpeechError(\n          'Cannot write to a streaming session after end()',\n          'SPEECH_SESSION_ENDED',\n          this.provider,\n        ),\n      );\n    }\n\n    let bytes: Uint8Array;\n    try {\n      bytes = chunkToBytes(chunk, this.provider);\n    } catch (error) {\n      return Promise.reject(error);\n    }\n    if (bytes.byteLength === 0) {\n      return Promise.resolve();\n    }\n    if (this.queued + bytes.byteLength > this.init.maxBufferedBytes) {\n      return Promise.reject(\n        new SpeechError(\n          `Streaming write would queue more than maxBufferedBytes (${this.init.maxBufferedBytes}); await write() before sending more audio`,\n          'SPEECH_BACKPRESSURE',\n          this.provider,\n        ),\n      );\n    }\n\n    const splitter = this.splitter;\n    let cuts: number[] = [];\n    if (splitter && !this.init.turnLimit?.rollover) {\n      if (splitter.wouldOverflow(bytes.byteLength)) {\n        return Promise.reject(\n          new SpeechError(\n            `Streaming write would push the current turn past maxTurnSeconds (${this.init.turnLimit?.maxTurnSeconds} s); call commit() before writing more audio`,\n            'SPEECH_TURN_TOO_LONG',\n            this.provider,\n          ),\n        );\n      }\n      splitter.add(bytes.byteLength);\n    } else if (splitter) {\n      cuts = splitter.split(bytes);\n    }\n\n    return new Promise<void>((resolve, reject) => {\n      // A rollover splits the chunk: each part before a cut ends with an\n      // automatic commit, and the write settles with its last part.\n      const pushPart = (start: number, end: number) => {\n        const part = bytes.subarray(start, end);\n        this.queue.push({\n          kind: 'audio',\n          // Encode now so callers may reuse their buffer once write() returns.\n          base64: encodeBase64(part),\n          bytes: part.byteLength,\n          resolve: end === bytes.byteLength ? resolve : noop,\n          reject,\n        });\n        this.queued += part.byteLength;\n      };\n      // Cuts are ascending offsets in 1..byteLength, so no part is empty.\n      let start = 0;\n      for (const cut of cuts) {\n        pushPart(start, cut);\n        this.queue.push({ kind: 'commit', final: false });\n        start = cut;\n      }\n      if (start < bytes.byteLength) {\n        pushPart(start, bytes.byteLength);\n      }\n      void this.pump();\n    });\n  }\n\n  private async connect(): Promise<void> {\n    this.connectTimer = setTimeout(() => {\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} connection timed out after ${this.init.connectTimeoutMs} ms`,\n        ),\n      );\n    }, this.init.connectTimeoutMs);\n\n    try {\n      const connection = await this.init.connect();\n      if (this.isTerminal()) {\n        return;\n      }\n      this.secrets = connection.secrets;\n      const socket = this.init.createWebSocket(connection.url, {\n        protocols: connection.protocols,\n        headers: connection.headers,\n      });\n      this.socket = socket;\n      try {\n        socket.binaryType = 'arraybuffer';\n      } catch {\n        // Some implementations expose a read-only binaryType.\n      }\n      socket.addEventListener('open', () => this.onOpen());\n      socket.addEventListener('message', (event) => this.onMessage(event.data));\n      socket.addEventListener('error', () => {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} WebSocket error`,\n          ),\n        );\n      });\n      socket.addEventListener('close', (event) => this.onClose(event));\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n    }\n  }\n\n  private onOpen(): void {\n    clearTimeout(this.connectTimer);\n    if (this.isTerminal()) {\n      this.closeSocket();\n      return;\n    }\n\n    this.opened = true;\n    for (const message of this.init.protocol.sessionMessages(\n      this.init.config,\n    )) {\n      if (!this.send(message)) {\n        return;\n      }\n    }\n    if (this.state === 'connecting') {\n      this.state = 'open';\n    }\n    this.resolveReady();\n    this.emit('open', { sessionId: this.sessionId });\n    void this.pump();\n  }\n\n  private onMessage(data: unknown): void {\n    if (this.isTerminal()) {\n      return;\n    }\n\n    this.lastServerMessageAt = Date.now();\n    if (this.endWaiter) {\n      this.armEndTimer();\n    }\n\n    const text = messageDataToText(data);\n    if (text === undefined) {\n      return;\n    }\n\n    let parsed: unknown;\n    try {\n      parsed = JSON.parse(text);\n    } catch {\n      return;\n    }\n\n    let events: RealtimeProtocolEvent[];\n    try {\n      events = this.init.protocol.parse(parsed);\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n      return;\n    }\n\n    for (const event of events) {\n      if (this.isTerminal()) {\n        return;\n      }\n      this.handle(event);\n    }\n    this.checkEnded();\n  }\n\n  private handle(event: RealtimeProtocolEvent): void {\n    switch (event.type) {\n      case 'session':\n        this.sessionId = event.sessionId ?? this.sessionId;\n        return;\n      case 'committed':\n        if (event.itemId && !this.pendingItems.has(event.itemId)) {\n          this.pendingItems.add(event.itemId);\n          this.itemOrder.push(event.itemId);\n        }\n        // A turn whose speech_stopped we saw was committed by server VAD, so\n        // it does not acknowledge one of our commits.\n        if (\n          this.init.protocol.commitAck === 'committed' &&\n          !(event.itemId && this.vadStoppedItems.has(event.itemId))\n        ) {\n          this.ackCommit();\n        }\n        return;\n      case 'commit_empty':\n        this.ackCommit();\n        return;\n      case 'partial': {\n        const key = event.itemId ?? '';\n        const text = (this.partials.get(key) ?? '') + event.delta;\n        this.partials.set(key, text);\n        this.emit('partial', {\n          delta: event.delta,\n          text,\n          itemId: event.itemId,\n        });\n        return;\n      }\n      case 'final': {\n        let serverEnded = false;\n        if (this.init.protocol.commitAck === 'final') {\n          if (this.outstandingCommits === 0) {\n            // A final we did not ask for: the server ended the open turn by\n            // itself (e.g. a server-side generation limit).\n            serverEnded = true;\n          } else if (this.ackCommit() && event.audioConsumed === false) {\n            // Our turn carried audio but the server consumed none. vLLM does\n            // this when a stale end-of-turn marker (left by a commit that\n            // crossed a server-ended turn) ends the turn at once.\n            this.fail(\n              new SpeechProviderError(\n                this.provider,\n                `${this.provider} finished a turn without transcribing its audio (the server's turn state is out of sync, e.g. after a commit crossed a server-ended turn); start a new session`,\n              ),\n            );\n            return;\n          }\n        }\n        if (event.itemId) {\n          this.pendingItems.delete(event.itemId);\n        }\n        this.partials.delete(event.itemId ?? '');\n        this.finals.push({\n          itemId: event.itemId,\n          text: event.text,\n          providerUsage: event.providerUsage,\n          raw: event.raw,\n          arrival: this.finals.length,\n        });\n        this.emit('final', {\n          text: event.text,\n          itemId: event.itemId,\n          providerUsage: event.providerUsage,\n          raw: event.raw,\n        });\n        if (serverEnded) {\n          // Fail closed: audio sent after the cut-off was never transcribed,\n          // and the server may now be out of step with our turns.\n          this.fail(\n            new SpeechProviderError(\n              this.provider,\n              `${this.provider} ended a turn before it was committed (for example at a server generation limit); audio sent after that point was not transcribed. Call commit() more often and start a new session`,\n            ),\n          );\n        }\n        return;\n      }\n      case 'speech_started':\n        this.emit(event.type, { itemId: event.itemId, audioMs: event.audioMs });\n        return;\n      case 'speech_stopped':\n        if (event.itemId) {\n          this.vadStoppedItems.add(event.itemId);\n        }\n        this.emit(event.type, { itemId: event.itemId, audioMs: event.audioMs });\n        return;\n      case 'error':\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            this.redact(\n              `${this.provider} error${event.code ? ` (${event.code})` : ''}: ${event.message}`,\n            ) ?? `${this.provider} error`,\n            {\n              responseBody:\n                event.raw === undefined\n                  ? undefined\n                  : this.redact(JSON.stringify(event.raw)),\n            },\n          ),\n        );\n        return;\n    }\n  }\n\n  /** Acknowledges the oldest outstanding commit; returns whether it carried audio. */\n  private ackCommit(): boolean {\n    this.outstandingCommits = Math.max(0, this.outstandingCommits - 1);\n    return this.commitHadAudio.shift() ?? false;\n  }\n\n  private async pump(): Promise<void> {\n    if (this.pumping || !this.opened) {\n      return;\n    }\n\n    this.pumping = true;\n    try {\n      while (this.queue.length > 0 && !this.isTerminal()) {\n        await this.waitForDrain();\n        if (!(await this.openTurnIfNeeded())) {\n          break;\n        }\n        const item = this.queue.shift();\n        if (!item || this.isTerminal()) {\n          if (item?.kind === 'audio') {\n            item.reject(this.error ?? new Error('Session closed'));\n          }\n          break;\n        }\n\n        if (item.kind === 'commit') {\n          this.sendCommit(item.final);\n          continue;\n        }\n\n        this.queued -= item.bytes;\n        if (!this.send(this.init.protocol.appendMessage(item.base64))) {\n          item.reject(this.error ?? new Error('Session closed'));\n          break;\n        }\n        this.totalBytes += item.bytes;\n        this.uncommittedBytes += item.bytes;\n        await this.waitForDrain();\n        if (this.error) {\n          item.reject(this.error);\n        } else {\n          item.resolve();\n        }\n      }\n    } finally {\n      this.pumping = false;\n    }\n    this.checkEnded();\n  }\n\n  private sendCommit(final: boolean): void {\n    const needed =\n      this.uncommittedBytes > 0 ||\n      (final && this.init.protocol.endCommit === 'always');\n    if (!needed || !this.padShortTurn()) {\n      return;\n    }\n    if (this.send(this.init.protocol.commitMessage({ final }))) {\n      this.commitHadAudio.push(this.uncommittedBytes > 0);\n      this.uncommittedBytes = 0;\n      this.outstandingCommits += 1;\n      this.turnOpen = false;\n    }\n  }\n\n  /**\n   * In a session with a turn limit, tops up a non-empty turn shorter than the\n   * protocol's `minCommitSeconds` with silence (not counted as streamed\n   * audio). Rollover can leave such a tail, which the provider would\n   * otherwise reject as empty and drop. Returns false when the send failed.\n   */\n  private padShortTurn(): boolean {\n    const minSeconds = this.init.protocol.minCommitSeconds;\n    if (!this.init.turnLimit || !minSeconds || this.uncommittedBytes === 0) {\n      return true;\n    }\n    const format = this.init.config.format;\n    const align = blockAlign(format);\n    // Round up to whole sample frames so the turn reaches the minimum.\n    const minBytes =\n      Math.ceil((minSeconds * bytesPerSecond(format)) / align - 1e-9) * align;\n    const missing = minBytes - this.uncommittedBytes;\n    if (missing <= 0) {\n      return true;\n    }\n    if (\n      !this.send(\n        this.init.protocol.appendMessage(\n          encodeBase64(silence(missing, format)),\n        ),\n      )\n    ) {\n      return false;\n    }\n    this.uncommittedBytes += missing;\n    return true;\n  }\n\n  /**\n   * For protocols with an explicit turn start: before the next audio, waits\n   * for every outstanding turn to be acknowledged, then opens a new turn. Like\n   * the end timeout, the wait is bounded by provider inactivity: it fails only\n   * after `timeoutMs` without any server message. Returns false when the\n   * session has failed.\n   */\n  private async openTurnIfNeeded(): Promise<boolean> {\n    const start = this.init.protocol.turnStartMessage;\n    if (!start || this.turnOpen || this.queue[0]?.kind !== 'audio') {\n      return true;\n    }\n\n    const waitStartedAt = Date.now();\n    while (!this.isTerminal() && this.outstandingCommits > 0) {\n      const lastActivity = Math.max(waitStartedAt, this.lastServerMessageAt);\n      if (Date.now() - lastActivity >= this.init.timeoutMs) {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} timed out after ${this.init.timeoutMs} ms without a server message while waiting for the previous turn to finish`,\n          ),\n        );\n        break;\n      }\n      await new Promise((resolve) => setTimeout(resolve, DRAIN_POLL_MS));\n    }\n    if (this.isTerminal() || !this.send(start.call(this.init.protocol))) {\n      return false;\n    }\n    this.turnOpen = true;\n    return true;\n  }\n\n  /**\n   * Waits until the socket's send buffer is at or below `highWaterMark`. A\n   * socket that stays congested for `timeoutMs` fails the session, so a\n   * stalled connection can never leave `write()` pending forever.\n   */\n  private async waitForDrain(): Promise<void> {\n    const deadline = Date.now() + this.init.timeoutMs;\n    while (\n      !this.isTerminal() &&\n      this.socket &&\n      this.socket.bufferedAmount > this.init.highWaterMark\n    ) {\n      if (Date.now() >= deadline) {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} socket did not drain below highWaterMark within ${this.init.timeoutMs} ms`,\n          ),\n        );\n        return;\n      }\n      await new Promise((resolve) => setTimeout(resolve, DRAIN_POLL_MS));\n    }\n  }\n\n  private send(message: unknown): boolean {\n    try {\n      this.socket?.send(JSON.stringify(message));\n      return true;\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n      return false;\n    }\n  }\n\n  private checkEnded(): void {\n    if (\n      !this.endWaiter ||\n      this.isTerminal() ||\n      !this.opened ||\n      this.pumping ||\n      this.queue.length > 0 ||\n      this.outstandingCommits > 0 ||\n      this.pendingItems.size > 0\n    ) {\n      return;\n    }\n\n    clearTimeout(this.endTimer);\n    const waiter = this.endWaiter;\n    this.endWaiter = undefined;\n    waiter.resolve();\n  }\n\n  private async finish(): Promise<TranscriptResult> {\n    this.state = 'closed';\n    this.cleanup();\n    this.closeSocket();\n\n    const ordered = [...this.finals].sort(\n      (left, right) => this.rank(left) - this.rank(right),\n    );\n    const segments: TranscriptSegment[] = ordered\n      .map((final) => final.text.trim())\n      .filter(Boolean)\n      .map((text) => ({ text }));\n    const providerUsage = sumUsage(ordered.map((final) => final.providerUsage));\n    const durationSeconds = audioSecondsForBytes(\n      this.totalBytes,\n      this.init.config.format,\n    );\n    const usage: SpeechUsage = {\n      operation: 'transcription',\n      provider: this.provider,\n      model: this.init.config.model,\n      audioSeconds:\n        audioSecondsFromProviderUsage(providerUsage) ?? durationSeconds,\n      bytes: this.totalBytes,\n      providerUsage,\n    };\n    const result: TranscriptResult = {\n      text: segments.map((segment) => segment.text).join(' '),\n      language: this.init.config.language,\n      durationSeconds,\n      segments,\n      provider: this.provider,\n      model: this.init.config.model,\n      raw: {\n        sessionId: this.sessionId,\n        finals: ordered.map((final) => final.raw),\n      },\n      usage,\n    };\n\n    await reportSpeechUsage(usage, ...this.init.onUsage);\n    return result;\n  }\n\n  private rank(final: FinalRecord): number {\n    const index = final.itemId ? this.itemOrder.indexOf(final.itemId) : -1;\n    return index >= 0 ? index : this.itemOrder.length + final.arrival;\n  }\n\n  private onClose(event: SpeechWebSocketCloseEvent): void {\n    clearTimeout(this.connectTimer);\n    if (!this.isTerminal()) {\n      const code = event.code ?? 1005;\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} socket closed unexpectedly (code ${code}${event.reason ? `: ${this.redact(event.reason)}` : ''})`,\n        ),\n      );\n    }\n    if (!this.closeEmitted) {\n      this.closeEmitted = true;\n      this.emit('close', {\n        code: event.code ?? 1005,\n        reason: this.redact(event.reason ?? '') ?? '',\n      });\n    }\n  }\n\n  private fail(error: Error): void {\n    if (this.isTerminal()) {\n      return;\n    }\n\n    this.error = error;\n    this.state = 'failed';\n    this.cleanup();\n    const queue = this.queue;\n    this.queue = [];\n    this.queued = 0;\n    for (const item of queue) {\n      if (item.kind === 'audio') {\n        item.reject(error);\n      }\n    }\n    this.rejectReady(error);\n    const waiter = this.endWaiter;\n    this.endWaiter = undefined;\n    waiter?.reject(error);\n    this.emit('error', error);\n    this.closeSocket();\n  }\n\n  private cleanup(): void {\n    clearTimeout(this.connectTimer);\n    clearTimeout(this.endTimer);\n    this.init.signal?.removeEventListener('abort', this.onAbort);\n  }\n\n  private closeSocket(): void {\n    try {\n      this.socket?.close(1000);\n    } catch {\n      // Already closing.\n    }\n  }\n\n  private emit<E extends StreamingSessionEventName>(\n    event: E,\n    payload: StreamingSessionEvents[E],\n  ): void {\n    for (const listener of [...(this.listeners.get(event) ?? [])]) {\n      try {\n        (listener as StreamingSessionListener<E>)(payload);\n      } catch (error) {\n        // A throwing listener fails the session so the bug surfaces via end().\n        if (event !== 'error' && event !== 'close') {\n          this.fail(toError(error, this.provider));\n        }\n      }\n    }\n  }\n\n  private isTerminal(): boolean {\n    return this.state === 'closed' || this.state === 'failed';\n  }\n\n  private redact(text: string | undefined): string | undefined {\n    let result = text;\n    for (const secret of this.secrets) {\n      result = redactSecret(result, secret);\n    }\n    return result;\n  }\n}\n\n/**\n * Throws when a turn cap is below the provider's minimum commit length:\n * padding a short turn up to that minimum would then exceed the cap.\n */\nexport function assertTurnLimitFitsProtocol(\n  protocol: Pick<RealtimeProtocol, 'provider' | 'minCommitSeconds'>,\n  limit: Pick<TurnLimit, 'maxTurnSeconds'> | undefined,\n): void {\n  const minimum = protocol.minCommitSeconds;\n  if (limit && minimum && limit.maxTurnSeconds < minimum) {\n    throw new SpeechConfigurationError(\n      `maxTurnSeconds (${limit.maxTurnSeconds}) is below ${protocol.provider}'s minimum commit of ${minimum} s`,\n      protocol.provider,\n    );\n  }\n}\n\nfunction toError(value: unknown, provider: string): Error {\n  if (value instanceof Error) {\n    return value;\n  }\n  return new SpeechError(\n    value === undefined ? 'Streaming session aborted' : String(value),\n    'SPEECH_ABORTED',\n    provider,\n  );\n}\n\n/**\n * Sums numeric fields of per-turn provider usage blocks (recursively), keeping\n * the first value of non-numeric fields such as `type`.\n */\nexport function sumUsage(\n  blocks: Array<Record<string, unknown> | undefined>,\n): Record<string, unknown> | undefined {\n  const present = blocks.filter(\n    (block): block is Record<string, unknown> =>\n      Boolean(block) && typeof block === 'object',\n  );\n  if (present.length === 0) {\n    return undefined;\n  }\n\n  const total: Record<string, unknown> = {};\n  for (const block of present) {\n    for (const [key, value] of Object.entries(block)) {\n      const current = total[key];\n      if (typeof value === 'number') {\n        total[key] = (typeof current === 'number' ? current : 0) + value;\n      } else if (value && typeof value === 'object' && !Array.isArray(value)) {\n        total[key] = sumUsage([\n          current as Record<string, unknown> | undefined,\n          value as Record<string, unknown>,\n        ]);\n      } else if (current === undefined) {\n        total[key] = value;\n      }\n    }\n  }\n  return total;\n}\n","/**\n * OpenAI Realtime transcription adapter (`openai-realtime`).\n *\n * Targets the GA Realtime API transcription session (no `OpenAI-Beta`\n * header): connect to `wss://api.openai.com/v1/realtime?intent=transcription`,\n * send `session.update` with `session.type: \"transcription\"`, stream\n * `input_audio_buffer.append`, and read\n * `conversation.item.input_audio_transcription.delta` / `.completed`.\n * Reference: https://developers.openai.com/api/docs/guides/realtime-transcription\n * and https://developers.openai.com/api/reference/resources/realtime/client-events\n */\n\nimport { SpeechConfigurationError } from '../shared/errors.js';\nimport { compactJson, resolveOpenAICompatibleUrl } from '../shared/http.js';\nimport {\n  assertNoBrowserApiKey,\n  CLIENT_SECRET_SUBPROTOCOL_PREFIX,\n  mergeHeaderRecords,\n  realtimeConnector,\n} from '../shared/realtime-auth.js';\nimport {\n  assertTurnLimitFitsProtocol,\n  DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n  DEFAULT_STREAMING_HIGH_WATER_MARK,\n  DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n  DEFAULT_STREAMING_TIMEOUT_MS,\n  type RealtimeProtocol,\n  type RealtimeProtocolEvent,\n  type RealtimeSessionConfig,\n  RealtimeTranscriptionSession,\n} from '../shared/realtime-session.js';\nimport type {\n  OpenAIRealtimeTranscriberOptions,\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTurnDetection,\n  StreamingTurnLimit,\n} from '../shared/streaming-types.js';\nimport {\n  describeTurnLimit,\n  resolveTurnLimit,\n  type TurnLimit,\n} from '../shared/turn-limit.js';\nimport {\n  resolveWebSocketFactory,\n  type SpeechWebSocketFactory,\n  toWebSocketUrl,\n} from '../shared/websocket.js';\n\nconst ADAPTER = 'openai-realtime';\nexport const OPENAI_REALTIME_DEFAULT_URL = 'wss://api.openai.com/v1/realtime';\nexport const OPENAI_REALTIME_DEFAULT_MODEL = 'gpt-4o-transcribe';\n\n/** GA realtime input formats: 24 kHz mono PCM16, or 8 kHz mono G.711. */\nconst WIRE_FORMATS: Record<\n  StreamingAudioFormat['encoding'],\n  { type: string; sampleRate: number }\n> = {\n  pcm16: { type: 'audio/pcm', sampleRate: 24_000 },\n  g711_ulaw: { type: 'audio/pcmu', sampleRate: 8_000 },\n  g711_alaw: { type: 'audio/pcma', sampleRate: 8_000 },\n};\n\n/** The OpenAI Realtime transcription wire protocol. */\nexport const openAIRealtimeProtocol = (\n  extras: Pick<\n    OpenAIRealtimeTranscriberOptions,\n    'noiseReduction' | 'include' | 'transcriptionOptions'\n  > = {},\n): RealtimeProtocol => ({\n  provider: ADAPTER,\n  commitAck: 'committed',\n  endCommit: 'if-audio',\n  // No per-turn cap by default: server VAD ends turns. Manual sessions may opt\n  // in with `maxTurnSeconds`.\n  // OpenAI rejects a commit with under 100 ms of audio as empty.\n  minCommitSeconds: 0.1,\n  sessionMessages(config: RealtimeSessionConfig) {\n    return [\n      {\n        type: 'session.update',\n        session: openAIRealtimeTranscriptionSession(config, extras),\n      },\n    ];\n  },\n  appendMessage(base64Audio: string) {\n    return { type: 'input_audio_buffer.append', audio: base64Audio };\n  },\n  commitMessage() {\n    return { type: 'input_audio_buffer.commit' };\n  },\n  parse: parseOpenAIRealtimeEvent,\n});\n\n/**\n * The GA transcription session object (`session.type: \"transcription\"`),\n * shared by `session.update` and the client-secret mint request.\n */\nexport function openAIRealtimeTranscriptionSession(\n  config: RealtimeSessionConfig,\n  extras: Pick<\n    OpenAIRealtimeTranscriberOptions,\n    'noiseReduction' | 'include' | 'transcriptionOptions'\n  > = {},\n): Record<string, unknown> {\n  const input: Record<string, unknown> = {\n    format:\n      config.format.encoding === 'pcm16'\n        ? { type: 'audio/pcm', rate: config.format.sampleRate }\n        : { type: WIRE_FORMATS[config.format.encoding].type },\n    // Extension fields first: the typed settings must win, so the model\n    // sent (or bound to a minted client secret) always matches the model\n    // reported in usage.\n    transcription: {\n      ...compactJson(extras.transcriptionOptions ?? {}),\n      ...compactJson({\n        model: config.model,\n        language: config.language,\n        prompt: config.prompt,\n      }),\n    },\n    turn_detection: turnDetectionToWire(config.turnDetection),\n  };\n  if (extras.noiseReduction) {\n    input.noise_reduction = { type: extras.noiseReduction };\n  }\n\n  return compactJson({\n    type: 'transcription',\n    audio: { input },\n    include: extras.include,\n  });\n}\n\n/** Maps one OpenAI Realtime server event to normalised session events. */\nexport function parseOpenAIRealtimeEvent(\n  message: unknown,\n): RealtimeProtocolEvent[] {\n  if (!message || typeof message !== 'object') {\n    return [];\n  }\n\n  const event = message as Record<string, unknown>;\n  const itemId = stringField(event, 'item_id');\n\n  switch (event.type) {\n    case 'session.created':\n    case 'session.updated':\n    case 'transcription_session.created':\n    case 'transcription_session.updated': {\n      const session = event.session as Record<string, unknown> | undefined;\n      return [\n        { type: 'session', sessionId: session && stringField(session, 'id') },\n      ];\n    }\n    case 'input_audio_buffer.committed':\n      return [{ type: 'committed', itemId }];\n    case 'input_audio_buffer.speech_started':\n      return [\n        {\n          type: 'speech_started',\n          itemId,\n          audioMs: numberField(event, 'audio_start_ms'),\n        },\n      ];\n    case 'input_audio_buffer.speech_stopped':\n      return [\n        {\n          type: 'speech_stopped',\n          itemId,\n          audioMs: numberField(event, 'audio_end_ms'),\n        },\n      ];\n    case 'conversation.item.input_audio_transcription.delta': {\n      const delta = stringField(event, 'delta');\n      return delta ? [{ type: 'partial', itemId, delta }] : [];\n    }\n    case 'conversation.item.input_audio_transcription.completed': {\n      const usage = event.usage;\n      return [\n        {\n          type: 'final',\n          itemId,\n          text: stringField(event, 'transcript') ?? '',\n          providerUsage:\n            usage && typeof usage === 'object' && !Array.isArray(usage)\n              ? (usage as Record<string, unknown>)\n              : undefined,\n          raw: event,\n        },\n      ];\n    }\n    case 'conversation.item.input_audio_transcription.failed': {\n      const error = (event.error ?? {}) as Record<string, unknown>;\n      return [\n        {\n          type: 'error',\n          message: stringField(error, 'message') ?? 'transcription failed',\n          code: stringField(error, 'code'),\n          raw: event,\n        },\n      ];\n    }\n    case 'error': {\n      const error = (event.error ?? {}) as Record<string, unknown>;\n      const code = stringField(error, 'code');\n      if (code === 'input_audio_buffer_commit_empty') {\n        return [{ type: 'commit_empty' }];\n      }\n      return [\n        {\n          type: 'error',\n          message: stringField(error, 'message') ?? 'unknown error',\n          code,\n          raw: event,\n        },\n      ];\n    }\n    default:\n      return [];\n  }\n}\n\n/**\n * Realtime transcriber for OpenAI and OpenAI-Realtime-compatible servers.\n *\n * Auth: `apiKey` is sent as `Authorization: Bearer` (Node only). In browsers,\n * pass `clientSecret` (an ephemeral client secret minted by your server); it\n * is sent as the `openai-insecure-api-key.<secret>` subprotocol. Supplying\n * `apiKey` or `headers` in a browser throws.\n */\nexport class OpenAIRealtimeTranscriber implements StreamingTranscriber {\n  readonly type = ADAPTER;\n  readonly audioFormat: StreamingAudioFormat;\n\n  private readonly options: OpenAIRealtimeTranscriberOptions;\n  private readonly url: string;\n  private readonly createWebSocket: SpeechWebSocketFactory;\n  private readonly protocol: RealtimeProtocol;\n\n  constructor(options: OpenAIRealtimeTranscriberOptions = {}) {\n    assertNoBrowserApiKey(ADAPTER, options.apiKey);\n\n    this.options = options;\n    this.url = resolveRealtimeUrl(\n      options.baseUrl ?? OPENAI_REALTIME_DEFAULT_URL,\n    );\n    this.createWebSocket = resolveWebSocketFactory(ADAPTER, options);\n    this.audioFormat = resolveOpenAIRealtimeFormat(options.format);\n    this.protocol = openAIRealtimeProtocol(options);\n    resolveTurnLimit(ADAPTER, undefined, options);\n  }\n\n  /**\n   * A turn limit applies only to `manual` sessions with `maxTurnSeconds` set;\n   * under server or semantic VAD the provider ends turns itself.\n   */\n  turnLimit(\n    options: StreamingSessionOptions = {},\n  ): StreamingTurnLimit | undefined {\n    return describeTurnLimit(\n      this.resolveTurnLimit(options),\n      this.sessionFormat(options),\n    );\n  }\n\n  private sessionFormat(\n    options: StreamingSessionOptions,\n  ): StreamingAudioFormat {\n    // A different encoding starts from that encoding's defaults.\n    return resolveOpenAIRealtimeFormat(\n      options.format?.encoding &&\n        options.format.encoding !== this.audioFormat.encoding\n        ? options.format\n        : { ...this.audioFormat, ...options.format },\n    );\n  }\n\n  private resolveTurnLimit(\n    options: StreamingSessionOptions,\n  ): TurnLimit | undefined {\n    const limit = resolveTurnLimit(\n      ADAPTER,\n      this.protocol.maxTurnSeconds,\n      options,\n      this.options,\n    );\n    if (this.turnDetection(options).type !== 'manual') {\n      return undefined;\n    }\n    assertTurnLimitFitsProtocol(this.protocol, limit);\n    return limit;\n  }\n\n  private turnDetection(\n    options: StreamingSessionOptions,\n  ): StreamingTurnDetection {\n    return (\n      options.turnDetection ??\n      this.options.turnDetection ?? { type: 'server_vad' }\n    );\n  }\n\n  start(options: StreamingSessionOptions = {}): StreamingSession {\n    const format = this.sessionFormat(options);\n    const connect = realtimeConnector({\n      adapter: ADAPTER,\n      url: this.url,\n      apiKey: this.options.apiKey,\n      headers: mergeHeaderRecords(this.options.headers, options.headers),\n      clientSecret: options.clientSecret ?? this.options.clientSecret,\n      protocols: (secret) =>\n        secret\n          ? ['realtime', `${CLIENT_SECRET_SUBPROTOCOL_PREFIX}${secret}`]\n          : ['realtime'],\n    });\n\n    return new RealtimeTranscriptionSession({\n      protocol: this.protocol,\n      config: {\n        model:\n          options.model?.trim() ||\n          this.options.model?.trim() ||\n          OPENAI_REALTIME_DEFAULT_MODEL,\n        format,\n        turnDetection: this.turnDetection(options),\n        language: options.language ?? this.options.language,\n        prompt: options.prompt ?? this.options.prompt,\n      },\n      connect,\n      createWebSocket: this.createWebSocket,\n      timeoutMs: this.options.timeoutMs ?? DEFAULT_STREAMING_TIMEOUT_MS,\n      connectTimeoutMs:\n        this.options.connectTimeoutMs ?? DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n      highWaterMark:\n        this.options.highWaterMark ?? DEFAULT_STREAMING_HIGH_WATER_MARK,\n      maxBufferedBytes:\n        this.options.maxBufferedBytes ?? DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n      signal: options.signal,\n      onUsage: [this.options.onUsage, options.onUsage],\n      turnLimit: this.resolveTurnLimit(options),\n    });\n  }\n}\n\n/**\n * Resolves a realtime WebSocket URL: `http(s)` becomes `ws(s)`, a server root\n * gets `/v1/realtime`, a versioned API root gets `/realtime`, and\n * `intent=transcription` is added unless an `intent` is already present. No\n * `model` query parameter is added: the model travels in `session.update`.\n */\nexport function resolveRealtimeUrl(baseUrl: string): string {\n  const url = new URL(\n    toWebSocketUrl(resolveOpenAICompatibleUrl(baseUrl, 'realtime'), ADAPTER),\n  );\n  if (!url.searchParams.has('intent')) {\n    url.searchParams.set('intent', 'transcription');\n  }\n  return url.toString();\n}\n\nexport function resolveOpenAIRealtimeFormat(\n  format: Partial<StreamingAudioFormat> | undefined,\n): StreamingAudioFormat {\n  const encoding = format?.encoding ?? 'pcm16';\n  const wire = WIRE_FORMATS[encoding];\n  if (!wire) {\n    throw new SpeechConfigurationError(\n      `openai-realtime does not support encoding ${String(encoding)}`,\n      ADAPTER,\n    );\n  }\n\n  const resolved: StreamingAudioFormat = {\n    encoding,\n    sampleRate: format?.sampleRate ?? wire.sampleRate,\n    channels: format?.channels ?? 1,\n  };\n  if (resolved.sampleRate !== wire.sampleRate || resolved.channels !== 1) {\n    throw new SpeechConfigurationError(\n      `openai-realtime requires ${encoding} at ${wire.sampleRate} Hz mono (got ${resolved.sampleRate} Hz, ${resolved.channels} channel(s)); resample before streaming`,\n      ADAPTER,\n    );\n  }\n  return resolved;\n}\n\nfunction turnDetectionToWire(\n  turnDetection: StreamingTurnDetection,\n): Record<string, unknown> | null {\n  switch (turnDetection.type) {\n    case 'manual':\n      return null;\n    case 'server_vad':\n      return compactJson({\n        type: 'server_vad',\n        threshold: turnDetection.threshold,\n        prefix_padding_ms: turnDetection.prefixPaddingMs,\n        silence_duration_ms: turnDetection.silenceDurationMs,\n      });\n    case 'semantic_vad':\n      return compactJson({\n        type: 'semantic_vad',\n        eagerness: turnDetection.eagerness,\n      });\n    default:\n      throw new SpeechConfigurationError(\n        `Unknown turn detection type: ${String((turnDetection as { type?: unknown }).type)}`,\n        ADAPTER,\n      );\n  }\n}\n\nfunction stringField(\n  record: Record<string, unknown>,\n  key: string,\n): string | undefined {\n  const value = record[key];\n  return typeof value === 'string' ? value : undefined;\n}\n\nfunction numberField(\n  record: Record<string, unknown>,\n  key: string,\n): number | undefined {\n  const value = record[key];\n  return typeof value === 'number' ? value : undefined;\n}\n","/**\n * vLLM realtime transcription adapter (`voxtral-realtime`), for Mistral's\n * Voxtral Realtime models (e.g. `mistralai/Voxtral-Mini-4B-Realtime-2602`)\n * served by vLLM at `/v1/realtime`.\n *\n * vLLM's protocol borrows OpenAI event names but is not OpenAI-compatible.\n * Verified against vLLM's `entrypoints/speech_to_text/realtime` source and a\n * live server:\n *\n * 1. The server sends `session.created` (`{ id, created }`) on connect.\n * 2. The client sends `session.update` with a top-level `model` (no nested\n *    `session` object, no acknowledgement; an unknown model yields an\n *    `error` with code `model_not_found`).\n * 3. A non-final `input_audio_buffer.commit` starts generation for a turn.\n * 4. `input_audio_buffer.append` carries base64 PCM16 at 16 kHz mono. Audio\n *    appended before the start commit is kept.\n * 5. The server streams `transcription.delta` (`{ delta }`, often empty) while\n *    audio arrives, at roughly real-time speed.\n * 6. `input_audio_buffer.commit` with `final: true` ends the turn; the server\n *    replies `transcription.done` (`{ text, usage }`, token usage only) and\n *    clears its buffer. A further start commit opens the next turn on the same\n *    socket.\n * 7. Errors arrive as `{ type: 'error', error: string, code }` and leave the\n *    socket open; this adapter treats every error as fatal.\n * 8. A turn whose audio and tokens exceed `max_model_len` (~12.5 tokens per\n *    audio second plus a 39-token prompt) ends in `error` `processing_error`\n *    and can take the engine down. The limit is per turn: after a final\n *    commit vLLM clears the turn's context. The session therefore rolls over\n *    to a new turn before `maxTurnSeconds` (default 270) of audio.\n * 9. A stray final commit leaves a stale end marker: the next turn ends at once\n *    with `prompt_tokens: 1` (any audio costs 39) and its audio is dropped.\n *    The session fails closed on that, and on any `done` for a turn it did\n *    not commit, rather than losing audio silently.\n */\n\nimport { SpeechConfigurationError } from '../shared/errors.js';\nimport { resolveOpenAICompatibleUrl } from '../shared/http.js';\nimport {\n  assertNoBrowserApiKey,\n  CLIENT_SECRET_SUBPROTOCOL_PREFIX,\n  mergeHeaderRecords,\n  realtimeConnector,\n} from '../shared/realtime-auth.js';\nimport {\n  DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n  DEFAULT_STREAMING_HIGH_WATER_MARK,\n  DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n  DEFAULT_STREAMING_TIMEOUT_MS,\n  type RealtimeProtocol,\n  type RealtimeProtocolEvent,\n  RealtimeTranscriptionSession,\n} from '../shared/realtime-session.js';\nimport type {\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTurnDetection,\n  StreamingTurnLimit,\n  VoxtralRealtimeTranscriberOptions,\n} from '../shared/streaming-types.js';\nimport {\n  DEFAULT_MAX_TURN_SECONDS,\n  describeTurnLimit,\n  resolveTurnLimit,\n  type TurnLimit,\n} from '../shared/turn-limit.js';\nimport {\n  resolveWebSocketFactory,\n  type SpeechWebSocketFactory,\n  toWebSocketUrl,\n} from '../shared/websocket.js';\n\nconst ADAPTER = 'voxtral-realtime';\n\n/** vLLM's default served model name for Voxtral Mini Realtime. */\nexport const VOXTRAL_REALTIME_DEFAULT_MODEL =\n  'mistralai/Voxtral-Mini-4B-Realtime-2602';\n\n/** The only input format vLLM's realtime endpoint accepts. */\nexport const VOXTRAL_REALTIME_AUDIO_FORMAT: Readonly<StreamingAudioFormat> =\n  Object.freeze({ encoding: 'pcm16', sampleRate: 16_000, channels: 1 });\n\n/** The vLLM realtime wire protocol. */\nexport const voxtralRealtimeProtocol = (): RealtimeProtocol => ({\n  provider: ADAPTER,\n  // vLLM never acknowledges a commit; `transcription.done` closes the turn.\n  commitAck: 'final',\n  // The start commit is sent lazily before a turn's first audio, so a session\n  // with no uncommitted audio has no open turn and needs no end commit.\n  endCommit: 'if-audio',\n  // vLLM's context (`max_model_len`) covers one turn; roll over before it.\n  maxTurnSeconds: DEFAULT_MAX_TURN_SECONDS,\n  sessionMessages(config) {\n    return [{ type: 'session.update', model: config.model }];\n  },\n  turnStartMessage() {\n    return { type: 'input_audio_buffer.commit' };\n  },\n  appendMessage(base64Audio) {\n    return { type: 'input_audio_buffer.append', audio: base64Audio };\n  },\n  commitMessage() {\n    // Every turn end is a final commit; the next turn gets a new start commit.\n    return { type: 'input_audio_buffer.commit', final: true };\n  },\n  parse: parseVoxtralRealtimeEvent,\n});\n\n/** Maps one vLLM realtime server event to normalised session events. */\nexport function parseVoxtralRealtimeEvent(\n  message: unknown,\n): RealtimeProtocolEvent[] {\n  if (!message || typeof message !== 'object') {\n    return [];\n  }\n\n  const event = message as Record<string, unknown>;\n  switch (event.type) {\n    case 'session.created':\n      return [\n        {\n          type: 'session',\n          sessionId: typeof event.id === 'string' ? event.id : undefined,\n        },\n      ];\n    case 'transcription.delta': {\n      const delta = typeof event.delta === 'string' ? event.delta : '';\n      return delta ? [{ type: 'partial', delta }] : [];\n    }\n    case 'transcription.done': {\n      const usage =\n        event.usage &&\n        typeof event.usage === 'object' &&\n        !Array.isArray(event.usage)\n          ? (event.usage as Record<string, unknown>)\n          : undefined;\n      // Any audio, even one sample, costs the turn's full audio prompt\n      // (39 tokens on Voxtral Mini Realtime); a turn that consumed no audio\n      // reports `prompt_tokens: 1`.\n      const promptTokens = usage?.prompt_tokens;\n      return [\n        {\n          type: 'final',\n          text: typeof event.text === 'string' ? event.text.trim() : '',\n          providerUsage: usage,\n          raw: event,\n          audioConsumed:\n            typeof promptTokens === 'number' ? promptTokens > 1 : undefined,\n        },\n      ];\n    }\n    case 'error':\n      return [\n        {\n          type: 'error',\n          message:\n            typeof event.error === 'string' ? event.error : 'unknown error',\n          code: typeof event.code === 'string' ? event.code : undefined,\n          raw: event,\n        },\n      ];\n    default:\n      return [];\n  }\n}\n\n/**\n * Realtime transcriber for Voxtral Realtime served by vLLM.\n *\n * - Audio: PCM16, 16 kHz, mono only; write whole samples (even byte counts).\n * - Turns: manual only. vLLM has no VAD; it streams partials continuously and\n *   produces one final per `commit()` / `end()`, plus one per automatic\n *   rollover when a turn reaches `maxTurnSeconds` (default 270).\n * - `language` and `prompt` are not sent: vLLM's realtime session accepts\n *   only `model`.\n * - Auth: `apiKey` → `Authorization: Bearer` (Node only). vLLM has no\n *   ephemeral tokens, so browsers must connect through the consumer's own\n *   proxy or token gateway, passing that gateway's short-lived `clientSecret`\n *   (sent as the `openai-insecure-api-key.<secret>` subprotocol next to\n *   `realtime`; the gateway must echo `realtime`).\n */\nexport class VoxtralRealtimeTranscriber implements StreamingTranscriber {\n  readonly type = ADAPTER;\n  readonly audioFormat: StreamingAudioFormat = {\n    ...VOXTRAL_REALTIME_AUDIO_FORMAT,\n  };\n\n  private readonly options: VoxtralRealtimeTranscriberOptions;\n  private readonly url: string;\n  private readonly createWebSocket: SpeechWebSocketFactory;\n  private readonly protocol = voxtralRealtimeProtocol();\n\n  constructor(options: VoxtralRealtimeTranscriberOptions = {}) {\n    assertNoBrowserApiKey(ADAPTER, options.apiKey);\n    if (!options.baseUrl?.trim()) {\n      throw new SpeechConfigurationError(\n        'voxtral-realtime requires baseUrl (the vLLM server, e.g. http://localhost:8000 or its /v1 root)',\n        ADAPTER,\n      );\n    }\n\n    this.options = options;\n    this.url = resolveVoxtralRealtimeUrl(options.baseUrl);\n    this.createWebSocket = resolveWebSocketFactory(ADAPTER, options);\n    assertFormat(options.format);\n    assertTurnDetection(options.turnDetection);\n    this.resolveTurnLimit();\n  }\n\n  turnLimit(\n    options: StreamingSessionOptions = {},\n  ): StreamingTurnLimit | undefined {\n    assertFormat(options.format);\n    return describeTurnLimit(\n      this.resolveTurnLimit(options),\n      VOXTRAL_REALTIME_AUDIO_FORMAT,\n    );\n  }\n\n  private resolveTurnLimit(\n    options: StreamingSessionOptions = {},\n  ): TurnLimit | undefined {\n    return resolveTurnLimit(\n      ADAPTER,\n      this.protocol.maxTurnSeconds,\n      options,\n      this.options,\n    );\n  }\n\n  start(options: StreamingSessionOptions = {}): StreamingSession {\n    assertFormat(options.format);\n    assertTurnDetection(options.turnDetection);\n    const turnLimit = this.resolveTurnLimit(options);\n    const connect = realtimeConnector({\n      adapter: ADAPTER,\n      url: this.url,\n      apiKey: this.options.apiKey,\n      headers: mergeHeaderRecords(this.options.headers, options.headers),\n      clientSecret: options.clientSecret ?? this.options.clientSecret,\n      // A plain vLLM server selects no subprotocol, so offer none unless a\n      // gateway token must travel in one.\n      protocols: (secret) =>\n        secret\n          ? ['realtime', `${CLIENT_SECRET_SUBPROTOCOL_PREFIX}${secret}`]\n          : [],\n    });\n\n    return new RealtimeTranscriptionSession({\n      protocol: this.protocol,\n      config: {\n        model:\n          options.model?.trim() ||\n          this.options.model?.trim() ||\n          VOXTRAL_REALTIME_DEFAULT_MODEL,\n        format: { ...VOXTRAL_REALTIME_AUDIO_FORMAT },\n        turnDetection: { type: 'manual' },\n        language: options.language ?? this.options.language,\n      },\n      connect,\n      createWebSocket: this.createWebSocket,\n      timeoutMs: this.options.timeoutMs ?? DEFAULT_STREAMING_TIMEOUT_MS,\n      connectTimeoutMs:\n        this.options.connectTimeoutMs ?? DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n      highWaterMark:\n        this.options.highWaterMark ?? DEFAULT_STREAMING_HIGH_WATER_MARK,\n      maxBufferedBytes:\n        this.options.maxBufferedBytes ?? DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n      signal: options.signal,\n      onUsage: [this.options.onUsage, options.onUsage],\n      turnLimit,\n    });\n  }\n}\n\n/**\n * Resolves vLLM's realtime WebSocket URL: `http(s)` becomes `ws(s)`, a server\n * root gets `/v1/realtime`, a versioned API root (e.g. a gateway's\n * `/stt/v1`) gets `/realtime`, and a URL ending in `/realtime` is kept.\n */\nexport function resolveVoxtralRealtimeUrl(baseUrl: string): string {\n  return toWebSocketUrl(\n    resolveOpenAICompatibleUrl(baseUrl, 'realtime'),\n    ADAPTER,\n  );\n}\n\nfunction assertFormat(format: Partial<StreamingAudioFormat> | undefined): void {\n  const expected = VOXTRAL_REALTIME_AUDIO_FORMAT;\n  if (\n    format &&\n    ((format.encoding !== undefined && format.encoding !== expected.encoding) ||\n      (format.sampleRate !== undefined &&\n        format.sampleRate !== expected.sampleRate) ||\n      (format.channels !== undefined && format.channels !== expected.channels))\n  ) {\n    throw new SpeechConfigurationError(\n      `voxtral-realtime requires pcm16 at 16000 Hz mono (got ${format.encoding ?? 'pcm16'} at ${format.sampleRate ?? 16_000} Hz, ${format.channels ?? 1} channel(s)); resample before streaming`,\n      ADAPTER,\n    );\n  }\n}\n\nfunction assertTurnDetection(\n  turnDetection: StreamingTurnDetection | undefined,\n): void {\n  if (turnDetection && turnDetection.type !== 'manual') {\n    throw new SpeechConfigurationError(\n      `voxtral-realtime has no server turn detection (got ${turnDetection.type}); use manual turns with commit()/end()`,\n      ADAPTER,\n    );\n  }\n}\n","import {\n  compactJson,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n  resolveOpenAICompatibleUrl,\n  voiceToString,\n} from '../shared/http.js';\nimport type {\n  OpenAICompatibleSpeechSynthesizerOptions,\n  SpeechSynthesizer,\n  SynthesisRequest,\n  SynthesizedSpeech,\n} from '../shared/types.js';\n\nexport class OpenAICompatibleSpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'openai-compatible' as const;\n\n  /** Explicit override, resolved relative to `baseUrl` (or absolute). */\n  private readonly speechPath?: string;\n  private readonly defaultModel: string;\n  private readonly defaultVoice: string;\n\n  constructor(options: OpenAICompatibleSpeechSynthesizerOptions) {\n    super(options);\n    this.speechPath = options.speechPath;\n    this.defaultModel = options.defaultModel ?? 'tts-1';\n    this.defaultVoice = options.defaultVoice ?? 'alloy';\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const outputFormat = request.outputFormat ?? 'mp3';\n    const payload = compactJson({\n      model: request.model ?? this.defaultModel,\n      input: request.text,\n      voice: voiceToString(request.voice, this.defaultVoice),\n      // biome-ignore lint/style/useNamingConvention: OpenAI-compatible API uses snake_case.\n      response_format: outputFormat,\n      speed: request.speed,\n    });\n\n    // Without an override, resolve the endpoint the way the transcriber does:\n    // a server root gets `/v1/audio/speech`, a `/vN` API root gets\n    // `/audio/speech`, and a full endpoint URL is used as-is. Resolved per\n    // call so an invalid baseUrl still fails at synthesize time.\n    const endpoint =\n      this.speechPath ??\n      resolveOpenAICompatibleUrl(this.baseUrl, 'audio/speech');\n\n    const speech = await this.post(\n      this.type,\n      endpoint,\n      {\n        headers: {\n          'content-type': 'application/json',\n        },\n        body: JSON.stringify(payload),\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n\n    return {\n      ...speech,\n      format: speech.format ?? outputFormat,\n      model: speech.model ?? String(payload.model),\n    };\n  }\n}\n","import {\n  DEFAULT_MAX_AUDIO_BYTES,\n  normalizeAudioInput,\n} from '../shared/audio.js';\nimport { SpeechProviderError } from '../shared/errors.js';\nimport {\n  appendOptionalFormValue,\n  HttpSpeechAdapter,\n  readTranscriptResponse,\n  resolveOpenAICompatibleUrl,\n} from '../shared/http.js';\nimport {\n  resolveRetryOptions,\n  type SpeechRetryOptions,\n} from '../shared/retry.js';\nimport type {\n  OpenAICompatibleTranscriberOptions,\n  TimestampGranularity,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptionResponseFormat,\n  TranscriptResult,\n} from '../shared/types.js';\nimport {\n  audioSecondsFromProviderUsage,\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from '../shared/usage.js';\n\nconst DEFAULT_MODEL = 'whisper-1';\n\n/**\n * Models that only accept `json`/`text` and reject `verbose_json` and\n * `timestamp_granularities[]` (e.g. `gpt-4o-transcribe`,\n * `gpt-4o-mini-transcribe`, optionally gateway-prefixed like\n * `openai/gpt-4o-transcribe`).\n */\nconst JSON_ONLY_MODEL = /(^|\\/)gpt-4o[\\w.-]*-transcribe/i;\n\n/** Provider 400/422 bodies that indicate an unsupported format option. */\nconst UNSUPPORTED_FORMAT_ERROR =\n  /verbose_json|response_format|timestamp_granularit/i;\n\nexport function isJsonOnlyTranscriptionModel(model: string): boolean {\n  return JSON_ONLY_MODEL.test(model);\n}\n\ninterface TranscriptionWireOptions {\n  model: string;\n  responseFormat: TranscriptionResponseFormat;\n  timestampGranularities?: TimestampGranularity[];\n}\n\n/**\n * OpenAI-compatible speech-to-text adapter (`POST <base>/audio/transcriptions`).\n * Works with OpenAI, Groq, Fireworks, LiteLLM/Bifrost gateways, vLLM,\n * speaches/faster-whisper-server, and whisper.cpp server.\n *\n * Server-side only: the adapter holds the API key.\n */\nexport class OpenAICompatibleTranscriber\n  extends HttpSpeechAdapter\n  implements Transcriber\n{\n  readonly type = 'openai-compatible' as const;\n\n  private readonly endpoint: string;\n  private readonly model: string;\n  private readonly responseFormat?: TranscriptionResponseFormat;\n  private readonly maxBytes: number;\n  private readonly retry: Required<SpeechRetryOptions> | false;\n  private readonly onUsage?: SpeechUsageCallback;\n\n  constructor(options: OpenAICompatibleTranscriberOptions) {\n    super(options);\n    this.endpoint = resolveOpenAICompatibleUrl(\n      options.baseUrl,\n      'audio/transcriptions',\n    );\n    this.model = options.model?.trim() || DEFAULT_MODEL;\n    this.responseFormat = options.responseFormat;\n    this.maxBytes = options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES;\n    this.retry = resolveRetryOptions(options.retry);\n    this.onUsage = options.onUsage;\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    const audio = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes: request.maxBytes ?? this.maxBytes,\n      adapter: this.type,\n      signal: request.signal,\n    });\n    const wire = this.resolveWireOptions(request);\n\n    const send = (options: TranscriptionWireOptions) => {\n      const form = new FormData();\n      form.append('file', audio.blob, audio.filename);\n      form.set('model', options.model);\n      appendOptionalFormValue(form, 'language', request.language);\n      appendOptionalFormValue(form, 'prompt', request.prompt);\n      appendOptionalFormValue(form, 'temperature', request.temperature);\n      form.set('response_format', options.responseFormat);\n      for (const granularity of options.timestampGranularities ?? []) {\n        form.append('timestamp_granularities[]', granularity);\n      }\n\n      return this.post(\n        this.type,\n        this.endpoint,\n        {\n          body: form,\n          headers: request.headers,\n          signal: request.signal,\n        },\n        (response) => readTranscriptResponse(response, this.type),\n        this.retry,\n      );\n    };\n\n    let transcript: TranscriptResult;\n    try {\n      transcript = await send(wire);\n    } catch (error) {\n      if (!isUnsupportedFormatError(error, wire)) {\n        throw error;\n      }\n      // The server rejected verbose_json/timestamps: fall back to plain JSON.\n      transcript = await send({\n        model: wire.model,\n        responseFormat: 'json',\n      });\n    }\n\n    const providerUsage = extractProviderUsage(transcript.raw);\n    const usage: SpeechUsage = {\n      operation: 'transcription',\n      provider: this.type,\n      model: transcript.model ?? wire.model,\n      audioSeconds:\n        transcript.durationSeconds ??\n        audioSecondsFromProviderUsage(providerUsage) ??\n        audio.durationSeconds,\n      bytes: audio.bytes,\n      providerUsage,\n    };\n    const result: TranscriptResult = {\n      ...transcript,\n      // Plain `text` responses end with a newline; JSON text is kept verbatim.\n      text:\n        transcript.raw === undefined ? transcript.text.trim() : transcript.text,\n      provider: this.type,\n      model: transcript.model ?? wire.model,\n      usage,\n    };\n\n    await reportSpeechUsage(usage, this.onUsage, request.onUsage);\n    return result;\n  }\n\n  private resolveWireOptions(\n    request: TranscriptionRequest,\n  ): TranscriptionWireOptions {\n    const model = request.model?.trim() || this.model;\n    const jsonOnly = isJsonOnlyTranscriptionModel(model);\n    let responseFormat =\n      request.responseFormat ??\n      this.responseFormat ??\n      (jsonOnly ? 'json' : 'verbose_json');\n\n    if (jsonOnly && responseFormat === 'verbose_json') {\n      responseFormat = 'json';\n    }\n\n    const timestampGranularities =\n      responseFormat === 'verbose_json' &&\n      request.timestampGranularities?.length\n        ? [...new Set(request.timestampGranularities)]\n        : undefined;\n\n    return { model, responseFormat, timestampGranularities };\n  }\n}\n\nfunction isUnsupportedFormatError(\n  error: unknown,\n  wire: TranscriptionWireOptions,\n): boolean {\n  return (\n    error instanceof SpeechProviderError &&\n    (error.status === 400 || error.status === 422) &&\n    (wire.responseFormat === 'verbose_json' ||\n      Boolean(wire.timestampGranularities?.length)) &&\n    UNSUPPORTED_FORMAT_ERROR.test(error.responseBody ?? '')\n  );\n}\n\nfunction extractProviderUsage(\n  raw: unknown,\n): Record<string, unknown> | undefined {\n  if (!raw || typeof raw !== 'object') {\n    return undefined;\n  }\n\n  const usage = (raw as Record<string, unknown>).usage;\n  return usage && typeof usage === 'object' && !Array.isArray(usage)\n    ? (usage as Record<string, unknown>)\n    : undefined;\n}\n","import {\n  createHappyVerticalSynthesisForm,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n} from '../shared/http.js';\nimport type {\n  Qwen3SpeechSynthesizerOptions,\n  SpeechSynthesizer,\n  SynthesisRequest,\n  SynthesizedSpeech,\n} from '../shared/types.js';\n\nexport class Qwen3SpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'qwen3-tts' as const;\n\n  private readonly speechPath: string;\n  private readonly defaultModel: string;\n  private readonly defaultVoice?: string;\n\n  constructor(options: Qwen3SpeechSynthesizerOptions) {\n    super(options);\n    this.speechPath = options.speechPath ?? '/v1/audio/speech';\n    this.defaultModel = options.defaultModel ?? 'qwen3-tts';\n    this.defaultVoice = options.defaultVoice;\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const form = createHappyVerticalSynthesisForm(request, this.defaultVoice);\n    const speech = await this.post(\n      this.type,\n      this.speechPath,\n      {\n        body: form,\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n\n    return {\n      ...speech,\n      format: speech.format ?? request.outputFormat,\n      model: speech.model ?? request.model ?? this.defaultModel,\n    };\n  }\n}\n","import { normalizeAudioInput } from '../shared/audio.js';\nimport {\n  appendOptionalFormValue,\n  createHappyVerticalSynthesisForm,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n  readTranscriptResponse,\n} from '../shared/http.js';\nimport type {\n  SpeechSynthesizer,\n  StudioServerSpeechSynthesizerOptions,\n  StudioServerTranscriberOptions,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from '../shared/types.js';\n\nexport class StudioServerTranscriber\n  extends HttpSpeechAdapter\n  implements Transcriber\n{\n  readonly type = 'studio-server' as const;\n  private readonly transcribePath: string;\n\n  constructor(options: StudioServerTranscriberOptions) {\n    super(options);\n    this.transcribePath = options.transcribePath ?? '/v1/transcribe';\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    const audio = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes: request.maxBytes,\n      adapter: this.type,\n      signal: request.signal,\n      deriveExtension: false,\n    });\n    const form = new FormData();\n    form.append('audio', audio.blob, audio.filename);\n    appendOptionalFormValue(form, 'language', request.language);\n\n    return this.post(\n      this.type,\n      this.transcribePath,\n      {\n        body: form,\n        headers: request.headers,\n        signal: request.signal,\n      },\n      (response) => readTranscriptResponse(response, this.type),\n    );\n  }\n}\n\nexport class StudioServerSpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'studio-server' as const;\n  private readonly synthesizePath: string;\n  private readonly defaultVoice?: string;\n\n  constructor(options: StudioServerSpeechSynthesizerOptions) {\n    super(options);\n    this.synthesizePath = options.synthesizePath ?? '/v1/tts/synthesize';\n    this.defaultVoice = options.defaultVoice;\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const form = createHappyVerticalSynthesisForm(request, this.defaultVoice);\n\n    return this.post(\n      this.type,\n      this.synthesizePath,\n      {\n        body: form,\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n  }\n}\n","/**\n * Streaming transcriber factory, environment configuration, and the\n * record-then-send `Transcriber` wrapper.\n */\n\nimport { OpenAIRealtimeTranscriber } from '../adapters/openai-realtime.js';\nimport { VoxtralRealtimeTranscriber } from '../adapters/voxtral-realtime.js';\nimport { DEFAULT_MAX_AUDIO_BYTES, normalizeAudioInput } from './audio.js';\nimport {\n  defaultEnv,\n  mergeHeaderInits,\n  parseHeadersEnv,\n  parseOptionalInteger,\n  readEnv,\n  type SpeechEnv,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n  SpeechError,\n} from './errors.js';\nimport { unwrapRawAudio } from './pcm.js';\nimport type {\n  GetStreamingTranscriberOptions,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTranscriberType,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport type {\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from './types.js';\n\n/** Streaming adapters `getStreamingTranscriber()` can build. */\nexport const STREAMING_TRANSCRIBER_TYPES: readonly StreamingTranscriberType[] =\n  ['openai-realtime', 'voxtral-realtime'];\n\n/** Environment variable names read for each streaming setting, in priority order. */\nexport const STREAMING_TRANSCRIBER_ENV_KEYS = {\n  type: ['HAVE_SPEECH_STREAMING_TYPE'],\n  baseUrl: ['HAVE_SPEECH_STREAMING_BASE_URL'],\n  model: ['HAVE_SPEECH_STREAMING_MODEL'],\n  apiKey: ['HAVE_SPEECH_STREAMING_API_KEY'],\n  language: ['HAVE_SPEECH_STREAMING_LANGUAGE'],\n  turnDetection: ['HAVE_SPEECH_STREAMING_TURN_DETECTION'],\n  timeoutMs: [\n    'HAVE_SPEECH_STREAMING_TIMEOUT',\n    'HAVE_SPEECH_STREAMING_TIMEOUT_MS',\n  ],\n  connectTimeoutMs: ['HAVE_SPEECH_STREAMING_CONNECT_TIMEOUT_MS'],\n  headers: ['HAVE_SPEECH_STREAMING_HEADERS'],\n  maxTurnSeconds: ['HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS'],\n} as const;\n\nexport interface StreamingFactoryContext {\n  env?: SpeechEnv;\n  /** Headers merged under explicit `headers` (env → context → options). */\n  headers?: HeadersInit;\n}\n\nexport function isStreamingTranscriberType(\n  type: string | undefined,\n): type is StreamingTranscriberType {\n  return (\n    type !== undefined &&\n    (STREAMING_TRANSCRIBER_TYPES as readonly string[]).includes(type)\n  );\n}\n\n/**\n * Creates a realtime streaming transcriber. Explicit options win over\n * `HAVE_SPEECH_STREAMING_*` environment variables. The type defaults to\n * `openai-realtime`.\n */\nexport function getStreamingTranscriber(\n  options: GetStreamingTranscriberOptions = {},\n  context: StreamingFactoryContext = {},\n): StreamingTranscriber {\n  const env = context.env ?? defaultEnv();\n  const keys = STREAMING_TRANSCRIBER_ENV_KEYS;\n  const type = options.type ?? readEnv(env, ...keys.type) ?? 'openai-realtime';\n\n  if (!isStreamingTranscriberType(type)) {\n    throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n\n  const headers = mergeHeaderInits(\n    parseHeadersEnv(env, keys.headers),\n    context.headers,\n    options.headers,\n  );\n  const resolved = {\n    ...options,\n    type,\n    baseUrl: (options.baseUrl ?? readEnv(env, ...keys.baseUrl))?.trim(),\n    model: options.model ?? readEnv(env, ...keys.model),\n    apiKey: options.apiKey ?? readEnv(env, ...keys.apiKey),\n    language: options.language ?? readEnv(env, ...keys.language),\n    turnDetection:\n      options.turnDetection ??\n      parseTurnDetection(readEnv(env, ...keys.turnDetection)),\n    timeoutMs:\n      options.timeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.timeoutMs)),\n    connectTimeoutMs:\n      options.connectTimeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.connectTimeoutMs)),\n    maxTurnSeconds:\n      options.maxTurnSeconds ??\n      parseMaxTurnSeconds(readEnv(env, ...keys.maxTurnSeconds)),\n    headers,\n  };\n\n  switch (type) {\n    case 'openai-realtime':\n      return new OpenAIRealtimeTranscriber({ ...resolved, type });\n    case 'voxtral-realtime':\n      return new VoxtralRealtimeTranscriber({ ...resolved, type });\n    default:\n      throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n}\n\nexport function parseTurnDetection(\n  value: string | undefined,\n): StreamingTurnDetection | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const normalized = value.toLowerCase();\n  if (normalized === 'server_vad' || normalized === 'semantic_vad') {\n    return { type: normalized };\n  }\n  if (normalized === 'manual' || normalized === 'none') {\n    return { type: 'manual' };\n  }\n  throw new SpeechConfigurationError(\n    'HAVE_SPEECH_STREAMING_TURN_DETECTION must be server_vad, semantic_vad, or manual',\n  );\n}\n\n/**\n * Parses `HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS`: a positive number of\n * seconds, or `Infinity` for no limit.\n */\nexport function parseMaxTurnSeconds(\n  value: string | undefined,\n): number | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const parsed = Number(value);\n  if (Number.isNaN(parsed) || parsed <= 0) {\n    throw new SpeechConfigurationError(\n      'HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS must be a positive number of seconds or Infinity',\n    );\n  }\n  return parsed;\n}\n\nexport interface StreamingTranscriberWrapperOptions {\n  /** Bytes per `write()`. Default 64 KiB. */\n  chunkBytes?: number;\n  /** Upload limit in bytes. Default 25 MB; `Infinity` disables it. */\n  maxBytes?: number;\n  /**\n   * Turn detection for wrapped sessions. Default `manual`: the whole clip is\n   * one turn, committed by `end()`.\n   */\n  turnDetection?: StreamingTurnDetection;\n}\n\n/**\n * Adapts a streaming transcriber to the request/response `Transcriber`\n * interface for record-then-send callers. Each `transcribe()` opens one\n * session, streams the recording, and returns `session.end()`.\n *\n * Input must be raw audio: a PCM16/G.711 WAV file, `audio/pcm` or\n * `audio/L16` (optionally with `rate`/`channels` parameters), `audio/pcmu`,\n * `audio/pcma`, or untyped bytes in the adapter's default format. WAV headers\n * and MIME parameters set the session format; compressed recordings are\n * rejected, so use an HTTP transcriber for WebM/MP4/Ogg.\n *\n * A clip longer than the adapter's per-turn limit (`maxTurnSeconds`) is\n * never sent as one turn: with rollover (the default) the session splits it\n * into several turns at quiet boundaries, and with `rollover: false` the clip\n * is rejected with `SPEECH_TURN_TOO_LONG` before connecting.\n */\nexport function wrapStreamingTranscriber(\n  streaming: StreamingTranscriber,\n  options: StreamingTranscriberWrapperOptions = {},\n): Transcriber {\n  const chunkBytes = options.chunkBytes ?? 64 * 1024;\n  if (!Number.isInteger(chunkBytes) || chunkBytes <= 0) {\n    throw new SpeechConfigurationError(\n      'chunkBytes must be a positive integer',\n      streaming.type,\n    );\n  }\n\n  return {\n    type: streaming.type,\n    async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n      const audio = await normalizeAudioInput(request.audio, {\n        mimeType: request.mimeType,\n        maxBytes:\n          request.maxBytes ?? options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES,\n        adapter: streaming.type,\n        signal: request.signal,\n        deriveExtension: false,\n      });\n      const bytes = new Uint8Array(await audio.blob.arrayBuffer());\n      const raw = unwrapRawAudio(bytes, audio.mimeType, streaming.type);\n      const declared =\n        request.audio &&\n        typeof request.audio === 'object' &&\n        'data' in request.audio\n          ? request.audio\n          : undefined;\n      const format = {\n        ...raw.format,\n        sampleRate: raw.format.sampleRate ?? declared?.sampleRate,\n        channels: raw.format.channels ?? declared?.channels,\n      };\n\n      const sessionOptions: StreamingSessionOptions = {\n        format: Object.fromEntries(\n          Object.entries(format).filter(([, value]) => value !== undefined),\n        ),\n        language: request.language,\n        model: request.model,\n        prompt: request.prompt,\n        headers: request.headers,\n        signal: request.signal,\n        onUsage: request.onUsage,\n        metadata: request.metadata,\n        turnDetection: options.turnDetection ?? { type: 'manual' },\n      };\n      const limit = streaming.turnLimit?.(sessionOptions);\n      if (\n        limit &&\n        !limit.rollover &&\n        raw.bytes.byteLength > limit.maxTurnBytes\n      ) {\n        throw new SpeechError(\n          `The recording is longer than one turn may be (maxTurnSeconds ${limit.maxTurnSeconds} s) and rollover is disabled; enable rollover or split the recording`,\n          'SPEECH_TURN_TOO_LONG',\n          streaming.type,\n        );\n      }\n\n      const session = streaming.start(sessionOptions);\n\n      try {\n        for (\n          let offset = 0;\n          offset < raw.bytes.byteLength;\n          offset += chunkBytes\n        ) {\n          await session.write(raw.bytes.subarray(offset, offset + chunkBytes));\n        }\n      } catch (error) {\n        session.abort(error);\n        throw error;\n      }\n      return session.end();\n    },\n  };\n}\n","import { OpenAICompatibleSpeechSynthesizer } from '../adapters/openai-compatible.js';\nimport { OpenAICompatibleTranscriber } from '../adapters/openai-compatible-transcriber.js';\nimport { Qwen3SpeechSynthesizer } from '../adapters/qwen3.js';\nimport {\n  StudioServerSpeechSynthesizer,\n  StudioServerTranscriber,\n} from '../adapters/studio-server.js';\nimport {\n  defaultEnv,\n  hasAnyEnv,\n  hasTranscriberEnv,\n  parseOptionalInteger,\n  readEnv,\n  resolveTranscriberConfig,\n  type SpeechEnv,\n  TRANSCRIBER_ENV_KEYS,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n} from './errors.js';\nimport { compactJson } from './http.js';\nimport {\n  getStreamingTranscriber,\n  isStreamingTranscriberType,\n  STREAMING_TRANSCRIBER_TYPES,\n  wrapStreamingTranscriber,\n} from './streaming-factory.js';\nimport type {\n  GetStreamingTranscriberOptions,\n  StreamingTranscriberType,\n} from './streaming-types.js';\nimport {\n  getOptionalTranscriberFactory,\n  optionalTranscriberEntry,\n  registeredOptionalTranscribers,\n} from './registry.js';\nimport type {\n  GetSpeechOptions,\n  GetSpeechSynthesizerOptions,\n  GetTranscriberOptions,\n  OpenAICompatibleSpeechSynthesizerOptions,\n  OpenAICompatibleTranscriberOptions,\n  Qwen3SpeechSynthesizerOptions,\n  Speech,\n  SpeechAdapterAvailability,\n  SpeechFetch,\n  SpeechSynthesizer,\n  StudioServerSpeechSynthesizerOptions,\n  StudioServerTranscriberOptions,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from './types.js';\n\nexport interface SpeechFactoryContext {\n  env?: SpeechEnv;\n  fetch?: SpeechFetch;\n  headers?: HeadersInit;\n}\n\n/**\n * Creates a speech service from explicit options and/or environment variables.\n *\n * Explicit options win over environment defaults. If no STT or TTS config is\n * available, the returned service is still usable for dependency injection, but\n * calling the missing operation throws a configuration error.\n */\nexport async function getSpeech(\n  options: GetSpeechOptions = {},\n  context: SpeechFactoryContext = {},\n): Promise<Speech> {\n  const transcriber =\n    options.transcriber === false\n      ? undefined\n      : await getOptionalTranscriber(options.transcriber, context);\n  const synthesizer =\n    options.synthesizer === false\n      ? undefined\n      : await getOptionalSpeechSynthesizer(options.synthesizer, context);\n\n  return {\n    transcriber,\n    synthesizer,\n    async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n      if (!transcriber) {\n        throw new SpeechConfigurationError('No STT provider configured');\n      }\n      return transcriber.transcribe(request);\n    },\n    async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n      if (!synthesizer) {\n        throw new SpeechConfigurationError('No TTS provider configured');\n      }\n      return synthesizer.synthesize(request);\n    },\n  };\n}\n\nexport async function getTranscriber(\n  options: GetTranscriberOptions = {},\n  context: SpeechFactoryContext = {},\n): Promise<Transcriber> {\n  const streamingType =\n    options.type ??\n    readEnv(context.env ?? defaultEnv(), ...TRANSCRIBER_ENV_KEYS.type);\n  if (isStreamingTranscriberType(streamingType)) {\n    return getWrappedStreamingTranscriber(streamingType, options, context);\n  }\n\n  const optional = await getRegisteredTranscriber(options, context);\n  if (optional) {\n    return optional;\n  }\n\n  const resolved = normalizeTranscriberOptions(options, context);\n\n  switch (resolved.type) {\n    case 'studio-server':\n      return new StudioServerTranscriber(resolved);\n    case 'openai-compatible':\n      return new OpenAICompatibleTranscriber(resolved);\n    default:\n      throw new InvalidSpeechAdapterError(\n        (resolved as { type?: string }).type ?? 'unknown',\n        'STT',\n      );\n  }\n}\n\nexport async function getSpeechSynthesizer(\n  options?: GetSpeechSynthesizerOptions,\n  context: SpeechFactoryContext = {},\n): Promise<SpeechSynthesizer> {\n  const resolved = normalizeSpeechSynthesizerOptions(options, context);\n\n  switch (resolved.type) {\n    case 'studio-server':\n      return new StudioServerSpeechSynthesizer(resolved);\n    case 'qwen3-tts':\n      return new Qwen3SpeechSynthesizer(resolved);\n    case 'openai-compatible':\n      return new OpenAICompatibleSpeechSynthesizer(resolved);\n    default:\n      throw new InvalidSpeechAdapterError(\n        (resolved as { type?: string }).type ?? 'unknown',\n        'TTS',\n      );\n  }\n}\n\nexport function getAvailableSpeechAdapters(): SpeechAdapterAvailability {\n  return {\n    transcribers: [\n      'studio-server',\n      'openai-compatible',\n      ...registeredOptionalTranscribers(),\n      ...STREAMING_TRANSCRIBER_TYPES,\n    ],\n    streamingTranscribers: [...STREAMING_TRANSCRIBER_TYPES],\n    synthesizers: ['studio-server', 'qwen3-tts', 'openai-compatible'],\n  };\n}\n\n/**\n * Builds a streaming adapter from `getTranscriber()` options and wraps it as\n * a record-then-send `Transcriber`. Settings come from explicit options, then\n * `options.streaming`, then `HAVE_SPEECH_STREAMING_*`.\n */\nfunction getWrappedStreamingTranscriber(\n  type: StreamingTranscriberType,\n  options: GetTranscriberOptions,\n  context: SpeechFactoryContext,\n): Transcriber {\n  const streaming = getStreamingTranscriber(\n    {\n      ...options.streaming,\n      ...(compactJson({\n        type,\n        baseUrl: options.baseUrl,\n        apiKey: options.apiKey,\n        model: options.model,\n        headers: options.headers,\n        timeoutMs: options.timeoutMs,\n        onUsage: options.onUsage,\n      }) as GetStreamingTranscriberOptions),\n    },\n    { env: context.env, headers: context.headers },\n  );\n  return wrapStreamingTranscriber(streaming, { maxBytes: options.maxBytes });\n}\n\n/**\n * Builds opt-in adapters (e.g. `local`) registered by their subpath entry.\n * Returns `undefined` for built-in types.\n */\nasync function getRegisteredTranscriber(\n  options: GetTranscriberOptions,\n  context: SpeechFactoryContext,\n): Promise<Transcriber | undefined> {\n  const type = resolveTranscriberConfig(options, context).type;\n  const entry = type ? optionalTranscriberEntry(type) : undefined;\n  if (!type || !entry) {\n    return undefined;\n  }\n\n  const factory = getOptionalTranscriberFactory(type);\n  if (!factory) {\n    throw new SpeechConfigurationError(\n      `The '${type}' transcriber is opt-in: import '${entry}' before calling getTranscriber()`,\n      type,\n    );\n  }\n  return factory({ ...options, type: type as Transcriber['type'] }, context);\n}\n\nasync function getOptionalTranscriber(\n  options: GetTranscriberOptions | undefined,\n  context: SpeechFactoryContext,\n): Promise<Transcriber | undefined> {\n  if (!options && !hasTranscriberEnv(context.env ?? defaultEnv())) {\n    return undefined;\n  }\n\n  return getTranscriber(options, context);\n}\n\nasync function getOptionalSpeechSynthesizer(\n  options: GetSpeechSynthesizerOptions | undefined,\n  context: SpeechFactoryContext,\n): Promise<SpeechSynthesizer | undefined> {\n  if (!options && !hasSynthesizerEnv(context.env ?? defaultEnv())) {\n    return undefined;\n  }\n\n  return getSpeechSynthesizer(options, context);\n}\n\nfunction normalizeTranscriberOptions(\n  options: GetTranscriberOptions = {},\n  context: SpeechFactoryContext,\n): StudioServerTranscriberOptions | OpenAICompatibleTranscriberOptions {\n  const resolved = resolveTranscriberConfig(options, context);\n  const type = resolved.type ?? 'studio-server';\n\n  if (type !== 'studio-server' && type !== 'openai-compatible') {\n    throw new InvalidSpeechAdapterError(type, 'STT');\n  }\n\n  if (!resolved.baseUrl) {\n    throw new SpeechConfigurationError('STT baseUrl is required', type);\n  }\n\n  const shared = {\n    baseUrl: resolved.baseUrl,\n    fetch: resolved.fetch,\n    headers: resolved.headers,\n    apiKey: resolved.apiKey,\n    timeoutMs: resolved.timeoutMs,\n  };\n\n  if (type === 'openai-compatible') {\n    return {\n      ...shared,\n      type,\n      model: resolved.model,\n      responseFormat: options.responseFormat,\n      maxBytes: resolved.maxBytes,\n      retry: options.retry,\n      onUsage: options.onUsage,\n    };\n  }\n\n  return {\n    ...shared,\n    type,\n    transcribePath: options.transcribePath ?? resolved.path,\n  };\n}\n\nfunction normalizeSpeechSynthesizerOptions(\n  options: GetSpeechSynthesizerOptions | undefined,\n  context: SpeechFactoryContext,\n):\n  | StudioServerSpeechSynthesizerOptions\n  | Qwen3SpeechSynthesizerOptions\n  | OpenAICompatibleSpeechSynthesizerOptions {\n  const env = context.env ?? defaultEnv();\n  const type =\n    options?.type ??\n    (readEnv(\n      env,\n      'HAVE_SPEECH_TTS_TYPE',\n      'HAVE_SPEECH_TTS_ADAPTER',\n      'TTS_ADAPTER',\n    ) as GetSpeechSynthesizerOptions['type'] | undefined);\n  const baseUrl =\n    options?.baseUrl ??\n    readEnv(env, 'HAVE_SPEECH_TTS_BASE_URL', 'TTS_BASE_URL');\n\n  if (!type) {\n    throw new SpeechConfigurationError('TTS provider type is required');\n  }\n\n  if (\n    type !== 'studio-server' &&\n    type !== 'qwen3-tts' &&\n    type !== 'openai-compatible'\n  ) {\n    throw new InvalidSpeechAdapterError(type, 'TTS');\n  }\n\n  if (!baseUrl?.trim()) {\n    throw new SpeechConfigurationError('TTS baseUrl is required', type);\n  }\n\n  const shared = {\n    type,\n    baseUrl: baseUrl.trim(),\n    fetch: options?.fetch ?? context.fetch,\n    headers: options?.headers ?? context.headers,\n    apiKey:\n      options?.apiKey ??\n      readEnv(env, 'HAVE_SPEECH_TTS_API_KEY', 'TTS_API_KEY', 'SPEECH_API_KEY'),\n    timeoutMs:\n      options?.timeoutMs ??\n      parseOptionalInteger(\n        readEnv(env, 'HAVE_SPEECH_TTS_TIMEOUT_MS', 'TTS_TIMEOUT_MS'),\n      ),\n  };\n\n  if (type === 'studio-server') {\n    return {\n      ...shared,\n      type,\n      synthesizePath:\n        options?.synthesizePath ??\n        readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  if (type === 'qwen3-tts') {\n    return {\n      ...shared,\n      type,\n      speechPath:\n        options?.speechPath ?? readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultModel:\n        options?.defaultModel ??\n        readEnv(env, 'HAVE_SPEECH_TTS_MODEL', 'TTS_MODEL'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  if (type === 'openai-compatible') {\n    return {\n      ...shared,\n      type,\n      speechPath:\n        options?.speechPath ?? readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultModel:\n        options?.defaultModel ??\n        readEnv(env, 'HAVE_SPEECH_TTS_MODEL', 'TTS_MODEL'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  throw new InvalidSpeechAdapterError(type, 'TTS');\n}\n\nfunction hasSynthesizerEnv(env: SpeechEnv): boolean {\n  return hasAnyEnv(\n    env,\n    'HAVE_SPEECH_TTS_TYPE',\n    'HAVE_SPEECH_TTS_ADAPTER',\n    'HAVE_SPEECH_TTS_BASE_URL',\n    'TTS_ADAPTER',\n    'TTS_BASE_URL',\n  );\n}\n","/**\n * Server-side minting of short-lived browser credentials for realtime\n * transcription.\n *\n * A browser must never hold the long-lived provider key. Your server calls\n * {@link createStreamingClientSecret} once per tenant session (after its own\n * auth check), records the returned tenant/session attribution, and hands the\n * browser only `value` and `expiresAt`. The browser passes `value` as the\n * streaming adapter's `clientSecret`.\n *\n * - `openai-realtime`: `POST /v1/realtime/client_secrets` with\n *   `expires_after: { anchor: 'created_at', seconds }` and a\n *   `session.type: \"transcription\"` config, returning an `ek_…` secret\n *   (https://developers.openai.com/api/reference/resources/realtime/subresources/client_secrets/methods/create).\n * - `voxtral-realtime`: vLLM has no ephemeral-token mechanism, so this throws\n *   `SpeechConfigurationError`. Browser Voxtral needs your own proxy or token\n *   gateway in front of vLLM.\n */\n\nimport {\n  OPENAI_REALTIME_DEFAULT_MODEL,\n  OPENAI_REALTIME_DEFAULT_URL,\n  openAIRealtimeTranscriptionSession,\n  resolveOpenAIRealtimeFormat,\n} from '../adapters/openai-realtime.js';\nimport {\n  defaultEnv,\n  mergeHeaderInits,\n  parseHeadersEnv,\n  parseOptionalInteger,\n  readEnv,\n  type SpeechEnv,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n  SpeechProviderError,\n} from './errors.js';\nimport {\n  HttpSpeechAdapter,\n  redactSecret,\n  resolveOpenAICompatibleUrl,\n} from './http.js';\nimport { headerSecrets } from './realtime-auth.js';\nimport {\n  isStreamingTranscriberType,\n  parseTurnDetection,\n  STREAMING_TRANSCRIBER_ENV_KEYS,\n} from './streaming-factory.js';\nimport type {\n  OpenAIRealtimeTranscriberOptions,\n  StreamingAudioFormat,\n  StreamingTranscriberType,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport type { SpeechFetch } from './types.js';\nimport { isBrowserRuntime } from './websocket.js';\n\n/** Default client-secret lifetime: long enough to connect, short if leaked. */\nexport const DEFAULT_CLIENT_SECRET_TTL_SECONDS = 60;\n/** OpenAI's accepted `expires_after.seconds` range. */\nexport const CLIENT_SECRET_TTL_RANGE = { min: 10, max: 7200 } as const;\n\nexport interface CreateStreamingClientSecretOptions {\n  /** Streaming adapter. Default `HAVE_SPEECH_STREAMING_TYPE`, then `openai-realtime`. */\n  type?: StreamingTranscriberType;\n  /** Long-lived server key. Default `HAVE_SPEECH_STREAMING_API_KEY`. */\n  apiKey?: string;\n  /**\n   * API base. Accepts the same values as the streaming adapter\n   * (`https://api.openai.com`, `…/v1`, or `wss://…/v1/realtime`). Default\n   * `HAVE_SPEECH_STREAMING_BASE_URL`, then `https://api.openai.com/v1`.\n   */\n  baseUrl?: string;\n  /** Secret lifetime in seconds, 10–7200. Default 60. */\n  ttlSeconds?: number;\n  /**\n   * Tenant the secret is minted for. Echoed in the result for your usage\n   * ledger; OpenAI's transcription client secrets carry no attribution field.\n   */\n  tenantId?: string;\n  /** Your session id (e.g. the dictation session), echoed in the result. */\n  sessionId?: string;\n  /** Extra attribution echoed in the result. Never sent to the provider. */\n  metadata?: Record<string, unknown>;\n  /** Transcription model the secret is bound to. Default `HAVE_SPEECH_STREAMING_MODEL`, then `gpt-4o-transcribe`. */\n  model?: string;\n  language?: string;\n  prompt?: string;\n  /** Default `HAVE_SPEECH_STREAMING_TURN_DETECTION`, then `server_vad`. */\n  turnDetection?: StreamingTurnDetection;\n  /** Default PCM16 24 kHz mono. */\n  format?: Partial<StreamingAudioFormat>;\n  noiseReduction?: OpenAIRealtimeTranscriberOptions['noiseReduction'];\n  include?: string[];\n  /**\n   * Extra `transcription` fields. The typed `model`, `language`, and `prompt`\n   * take precedence, so the secret is bound to the model the result reports.\n   */\n  transcriptionOptions?: Record<string, unknown>;\n  /**\n   * Headers for the mint request, e.g. a per-tenant gateway virtual key\n   * (`x-bf-vk`) for provider-side attribution. Merged over\n   * `HAVE_SPEECH_STREAMING_HEADERS`.\n   */\n  headers?: HeadersInit;\n  fetch?: SpeechFetch;\n  signal?: AbortSignal;\n  /** Request timeout. Default `HAVE_SPEECH_STREAMING_TIMEOUT[_MS]`, then 10000. */\n  timeoutMs?: number;\n}\n\n/** A minted browser credential plus the attribution it was minted for. */\nexport interface StreamingClientSecretResult {\n  type: StreamingTranscriberType;\n  /** The short-lived secret. Send it to the browser as `clientSecret`. */\n  value: string;\n  /** Expiry as epoch milliseconds. */\n  expiresAt: number;\n  ttlSeconds: number;\n  model: string;\n  tenantId?: string;\n  sessionId?: string;\n  metadata?: Record<string, unknown>;\n  /** The provider's echoed session config. */\n  session?: Record<string, unknown>;\n}\n\nexport interface StreamingClientSecretContext {\n  env?: SpeechEnv;\n}\n\nconst OPENAI_CLIENT_SECRET_ENDPOINT = 'realtime/client_secrets';\n\n/**\n * Mints a short-lived, session-scoped client secret for browser streaming.\n * Server-side only: throws in a browser runtime, and for adapters without an\n * ephemeral-token mechanism (`voxtral-realtime`).\n */\nexport async function createStreamingClientSecret(\n  options: CreateStreamingClientSecretOptions = {},\n  context: StreamingClientSecretContext = {},\n): Promise<StreamingClientSecretResult> {\n  const env = context.env ?? defaultEnv();\n  const keys = STREAMING_TRANSCRIBER_ENV_KEYS;\n  const type = options.type ?? readEnv(env, ...keys.type) ?? 'openai-realtime';\n  if (!isStreamingTranscriberType(type)) {\n    throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n\n  if (isBrowserRuntime()) {\n    throw new SpeechConfigurationError(\n      'createStreamingClientSecret runs on your server: it needs the long-lived API key, which must never reach a browser',\n      type,\n    );\n  }\n\n  if (type === 'voxtral-realtime') {\n    throw new SpeechConfigurationError(\n      'voxtral-realtime cannot mint browser credentials: vLLM has no ephemeral-token mechanism. Put your own proxy or token gateway in front of vLLM, mint its short-lived tokens yourself, and never send the vLLM API key to a browser',\n      type,\n    );\n  }\n\n  const ttlSeconds = options.ttlSeconds ?? DEFAULT_CLIENT_SECRET_TTL_SECONDS;\n  if (\n    !Number.isInteger(ttlSeconds) ||\n    ttlSeconds < CLIENT_SECRET_TTL_RANGE.min ||\n    ttlSeconds > CLIENT_SECRET_TTL_RANGE.max\n  ) {\n    throw new SpeechConfigurationError(\n      `ttlSeconds must be an integer from ${CLIENT_SECRET_TTL_RANGE.min} to ${CLIENT_SECRET_TTL_RANGE.max}`,\n      type,\n    );\n  }\n\n  const apiKey = options.apiKey ?? readEnv(env, ...keys.apiKey);\n  if (!apiKey) {\n    throw new SpeechConfigurationError(\n      'createStreamingClientSecret requires apiKey (or HAVE_SPEECH_STREAMING_API_KEY)',\n      type,\n    );\n  }\n\n  const model =\n    options.model?.trim() ||\n    readEnv(env, ...keys.model)?.trim() ||\n    OPENAI_REALTIME_DEFAULT_MODEL;\n  const session = openAIRealtimeTranscriptionSession(\n    {\n      model,\n      format: resolveOpenAIRealtimeFormat(options.format),\n      turnDetection: options.turnDetection ??\n        parseTurnDetection(readEnv(env, ...keys.turnDetection)) ?? {\n          type: 'server_vad',\n        },\n      language: options.language ?? readEnv(env, ...keys.language),\n      prompt: options.prompt,\n    },\n    options,\n  );\n\n  const endpoint = resolveClientSecretUrl(\n    options.baseUrl?.trim() ||\n      readEnv(env, ...keys.baseUrl)?.trim() ||\n      OPENAI_REALTIME_DEFAULT_URL,\n  );\n  const headers = mergeHeaderInits(\n    parseHeadersEnv(env, keys.headers),\n    options.headers,\n  );\n  const client = new ClientSecretClient({\n    baseUrl: endpoint,\n    apiKey,\n    fetch: options.fetch,\n    headers,\n    timeoutMs:\n      options.timeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.timeoutMs)) ??\n      10_000,\n  });\n\n  const body = await client\n    .mint(\n      endpoint,\n      {\n        expires_after: { anchor: 'created_at', seconds: ttlSeconds },\n        session,\n      },\n      options.signal,\n    )\n    .catch((error: unknown) => {\n      throw redactProviderError(error, headerSecrets(headers));\n    });\n\n  const value = body.value;\n  if (typeof value !== 'string' || !value) {\n    throw new SpeechProviderError(\n      type,\n      'openai-realtime client secret response did not include a value',\n    );\n  }\n  const expiresAtSeconds = body.expires_at;\n\n  return {\n    type,\n    value,\n    expiresAt:\n      typeof expiresAtSeconds === 'number' && Number.isFinite(expiresAtSeconds)\n        ? expiresAtSeconds * 1000\n        : Date.now() + ttlSeconds * 1000,\n    ttlSeconds,\n    model,\n    tenantId: options.tenantId,\n    sessionId: options.sessionId,\n    metadata: options.metadata,\n    session:\n      body.session && typeof body.session === 'object'\n        ? (body.session as Record<string, unknown>)\n        : undefined,\n  };\n}\n\n/**\n * Resolves the client-secret endpoint from any accepted realtime base:\n * `ws(s)` becomes `http(s)`, a trailing `/realtime` and query are dropped,\n * then `/v1/realtime/client_secrets` (or `/realtime/client_secrets` under a\n * versioned root) is appended.\n */\nexport function resolveClientSecretUrl(baseUrl: string): string {\n  const url = new URL(baseUrl);\n  if (url.protocol === 'ws:') {\n    url.protocol = 'http:';\n  } else if (url.protocol === 'wss:') {\n    url.protocol = 'https:';\n  }\n  url.search = '';\n  url.hash = '';\n  url.pathname = url.pathname.replace(/\\/+$/, '').replace(/\\/realtime$/, '');\n  return resolveOpenAICompatibleUrl(\n    url.toString(),\n    OPENAI_CLIENT_SECRET_ENDPOINT,\n  );\n}\n\n/**\n * The HTTP layer redacts the API key from provider error bodies; this also\n * removes gateway credentials sent in `headers` (e.g. `x-bf-vk`).\n */\nfunction redactProviderError(error: unknown, secrets: string[]): unknown {\n  if (!(error instanceof SpeechProviderError) || secrets.length === 0) {\n    return error;\n  }\n  let responseBody = error.responseBody;\n  let message = error.message;\n  for (const secret of secrets) {\n    responseBody = redactSecret(responseBody, secret);\n    message = redactSecret(message, secret) ?? message;\n  }\n  if (responseBody === error.responseBody && message === error.message) {\n    return error;\n  }\n  return new SpeechProviderError(error.adapter ?? 'openai-realtime', message, {\n    status: error.status,\n    responseBody,\n    retryAfterMs: error.retryAfterMs,\n  });\n}\n\nclass ClientSecretClient extends HttpSpeechAdapter {\n  mint(\n    endpoint: string,\n    payload: Record<string, unknown>,\n    signal: AbortSignal | undefined,\n  ): Promise<Record<string, unknown>> {\n    return this.post(\n      'openai-realtime',\n      endpoint,\n      {\n        body: JSON.stringify(payload),\n        headers: { 'content-type': 'application/json' },\n        signal,\n      },\n      async (response) => {\n        const json = (await response.json().catch(() => undefined)) as unknown;\n        if (!json || typeof json !== 'object' || Array.isArray(json)) {\n          throw new SpeechProviderError(\n            'openai-realtime',\n            'openai-realtime client secret response was not a JSON object',\n          );\n        }\n        return json as Record<string, unknown>;\n      },\n    );\n  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Default `2`. `0` disables retries. */\n  maxRetries?: number;\n  /** First backoff delay in milliseconds. Default `500`. */\n  initialDelayMs?: number;\n  /**\n   * Upper bound for any single wait in milliseconds. Default `30000`. A\n   * `Retry-After` longer than this ends retrying rather than retrying early.\n   */\n  maxDelayMs?: number;\n}\n\nexport const DEFAULT_SPEECH_RETRY: Required<SpeechRetryOptions> = {\n  maxRetries: 2,\n  initialDelayMs: 500,\n  maxDelayMs: 30_000,\n};\n\nexport function resolveRetryOptions(\n  retry: SpeechRetryOptions | false | undefined,\n  defaults: Required<SpeechRetryOptions> | false = DEFAULT_SPEECH_RETRY,\n): Required<SpeechRetryOptions> | false {\n  if (retry === false) {\n    return false;\n  }\n\n  if (retry === undefined) {\n    return defaults;\n  }\n\n  const base = defaults || DEFAULT_SPEECH_RETRY;\n  return {\n    maxRetries: Math.floor(\n      finiteRetryValue('maxRetries', retry.maxRetries, base.maxRetries),\n    ),\n    initialDelayMs: finiteRetryValue(\n      'initialDelayMs',\n      retry.initialDelayMs,\n      base.initialDelayMs,\n    ),\n    maxDelayMs: finiteRetryValue(\n      'maxDelayMs',\n      retry.maxDelayMs,\n      base.maxDelayMs,\n    ),\n  };\n}\n\n/**\n * Rejects non-finite values (`NaN`, `Infinity`): they would make the retry\n * budget or delay cap unbounded. Negative values clamp to `0`.\n */\nfunction finiteRetryValue(\n  name: keyof SpeechRetryOptions,\n  value: number | undefined,\n  fallback: number,\n): number {\n  if (value === undefined) {\n    return Math.max(0, fallback);\n  }\n\n  if (typeof value !== 'number' || !Number.isFinite(value)) {\n    throw new SpeechConfigurationError(\n      `Speech retry option ${name} must be a finite number`,\n    );\n  }\n\n  return Math.max(0, value);\n}\n\nexport function isRetryableStatus(status: number | undefined): boolean {\n  return status === 429 || (status !== undefined && status >= 500);\n}\n\n/**\n * Parses a `Retry-After` header (delta-seconds or HTTP-date) into\n * milliseconds. Returns `undefined` when absent or unparseable.\n */\nexport function parseRetryAfter(\n  value: string | null | undefined,\n  now: number = Date.now(),\n): number | undefined {\n  if (!value?.trim()) {\n    return undefined;\n  }\n\n  const trimmed = value.trim();\n  if (/^\\d+(\\.\\d+)?$/.test(trimmed)) {\n    return Math.round(Number(trimmed) * 1000);\n  }\n\n  const date = Date.parse(trimmed);\n  return Number.isNaN(date) ? undefined : Math.max(0, date - now);\n}\n\n/**\n * Runs `attempt` until it succeeds, throws a non-retryable error, or the\n * retry budget is exhausted. Retryable errors are {@link SpeechProviderError}\n * instances whose `status` is 429 or 5xx.\n */\nexport async function withSpeechRetry<T>(\n  attempt: (attemptNumber: number) => Promise<T>,\n  retry: Required<SpeechRetryOptions> | false,\n  signal?: AbortSignal,\n): Promise<T> {\n  const maxRetries = retry ? retry.maxRetries : 0;\n\n  for (let attemptNumber = 0; ; attemptNumber += 1) {\n    try {\n      return await attempt(attemptNumber);\n    } catch (error) {\n      if (\n        !retry ||\n        attemptNumber >= maxRetries ||\n        signal?.aborted ||\n        !(error instanceof SpeechProviderError) ||\n        !isRetryableStatus(error.status)\n      ) {\n        throw error;\n      }\n\n      const backoff = Math.min(\n        retry.maxDelayMs,\n        retry.initialDelayMs * 2 ** attemptNumber,\n      );\n      const retryAfterMs = error.retryAfterMs;\n      if (retryAfterMs !== undefined && retryAfterMs > retry.maxDelayMs) {\n        throw error;\n      }\n\n      await sleep(Math.max(backoff, retryAfterMs ?? 0), signal);\n    }\n  }\n}\n\nfunction sleep(ms: number, signal?: AbortSignal): Promise<void> {\n  return new Promise((resolve, reject) => {\n    if (signal?.aborted) {\n      reject(signal.reason);\n      return;\n    }\n\n    const onAbort = () => {\n      clearTimeout(timer);\n      reject(signal?.reason);\n    };\n    const timer = setTimeout(() => {\n      signal?.removeEventListener('abort', onAbort);\n      resolve();\n    }, ms);\n    signal?.addEventListener('abort', onAbort, { once: true });\n  });\n}\n","import { SpeechConfigurationError, SpeechProviderError } from './errors.js';\nimport {\n  parseRetryAfter,\n  type SpeechRetryOptions,\n  withSpeechRetry,\n} from './retry.js';\nimport type {\n  HttpSpeechOptions,\n  SpeechFetch,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  TranscriptResult,\n  TranscriptSegment,\n  WordTiming,\n} from './types.js';\n\n/**\n * Parses a base URL, dropping any fragment and keeping any query string.\n * Throws {@link SpeechConfigurationError} for a blank or unparseable value.\n * The value is not echoed in the error because it may carry credentials.\n */\nfunction parseBaseUrl(baseUrl: string): URL {\n  const trimmed = baseUrl.trim();\n  if (!trimmed) {\n    throw new SpeechConfigurationError('Speech adapter baseUrl is required');\n  }\n\n  let url: URL;\n  try {\n    url = new URL(trimmed);\n  } catch {\n    throw new SpeechConfigurationError(\n      'Speech adapter baseUrl is not a valid absolute URL',\n    );\n  }\n\n  url.hash = '';\n  return url;\n}\n\n/**\n * Returns `baseUrl` with a trailing `/` on its path so relative paths resolve\n * beneath it. The query string is preserved and any fragment is dropped.\n */\nexport function normalizeBaseUrl(baseUrl: string): string {\n  const url = parseBaseUrl(baseUrl);\n  if (!url.pathname.endsWith('/')) {\n    url.pathname = `${url.pathname}/`;\n  }\n\n  return url.toString();\n}\n\n/**\n * Resolves `path` against `baseUrl`. A relative path without its own query\n * inherits the base URL's query (for example a gateway `api-version`); an\n * absolute URL is returned unchanged.\n */\nexport function resolveSpeechUrl(baseUrl: string, path: string): string {\n  const base = new URL(normalizeBaseUrl(baseUrl));\n  const isAbsolute = /^[a-z][a-z\\d+.-]*:/i.test(path);\n  const resolved = new URL(isAbsolute ? path : path.replace(/^\\//, ''), base);\n  if (!isAbsolute && !resolved.search && base.search) {\n    resolved.search = base.search;\n  }\n\n  return resolved.toString();\n}\n\n/**\n * Resolves an OpenAI-compatible endpoint from a base URL. Accepts a server\n * root (`https://api.openai.com` → `/v1/<resource>`), an API root ending in a\n * version segment (`http://gateway/stt/v1` → `/<resource>`), or the full\n * endpoint URL (used as-is). Only the path is rewritten: the query string is\n * preserved and any fragment is dropped.\n */\nexport function resolveOpenAICompatibleUrl(\n  baseUrl: string,\n  resource: string,\n): string {\n  const trimmedResource = resource.replace(/^\\/+|\\/+$/g, '');\n  const url = parseBaseUrl(baseUrl);\n  const pathname = url.pathname.replace(/\\/+$/, '');\n\n  if (pathname.endsWith(`/${trimmedResource}`)) {\n    url.pathname = pathname;\n  } else if (/\\/v\\d+$/.test(pathname)) {\n    url.pathname = `${pathname}/${trimmedResource}`;\n  } else {\n    url.pathname = `${pathname}/v1/${trimmedResource}`;\n  }\n\n  return url.toString();\n}\n\nexport function resolveFetch(fetchOverride?: SpeechFetch): SpeechFetch {\n  if (fetchOverride) {\n    return fetchOverride;\n  }\n\n  const globalFetch = globalThis.fetch;\n  if (!globalFetch) {\n    throw new SpeechConfigurationError(\n      'No fetch implementation available for speech adapter',\n    );\n  }\n\n  return globalFetch.bind(globalThis);\n}\n\nexport function mergeHeaders(\n  options: Pick<HttpSpeechOptions, 'apiKey' | 'headers'>,\n  extra?: HeadersInit,\n): Headers {\n  const headers = new Headers(options.headers);\n\n  if (options.apiKey && !headers.has('authorization')) {\n    headers.set('authorization', `Bearer ${options.apiKey}`);\n  }\n\n  if (extra) {\n    new Headers(extra).forEach((value, key) => {\n      headers.set(key, value);\n    });\n  }\n\n  return headers;\n}\n\nexport abstract class HttpSpeechAdapter {\n  protected readonly baseUrl: string;\n  protected readonly fetchImpl: SpeechFetch;\n  protected readonly apiKey?: string;\n  protected readonly headers?: HeadersInit;\n  protected readonly timeoutMs?: number;\n\n  constructor(options: HttpSpeechOptions) {\n    this.baseUrl = options.baseUrl;\n    this.fetchImpl = resolveFetch(options.fetch);\n    this.apiKey = options.apiKey;\n    this.headers = options.headers;\n    this.timeoutMs = options.timeoutMs;\n  }\n\n  /**\n   * POSTs to `path` (relative to `baseUrl`, or an absolute URL). The optional\n   * `retry` policy retries 429/5xx responses; `timeoutMs` applies per attempt.\n   * The request body must be re-sendable (string, Blob, FormData) when\n   * retries are enabled.\n   */\n  protected async post<T>(\n    adapterName: string,\n    path: string,\n    init: Omit<RequestInit, 'method'>,\n    readResponse: (response: Response) => Promise<T>,\n    retry: Required<SpeechRetryOptions> | false = false,\n  ): Promise<T> {\n    return withSpeechRetry(\n      () => this.postOnce(adapterName, path, init, readResponse),\n      retry,\n      init.signal ?? undefined,\n    );\n  }\n\n  private async postOnce<T>(\n    adapterName: string,\n    path: string,\n    init: Omit<RequestInit, 'method'>,\n    readResponse: (response: Response) => Promise<T>,\n  ): Promise<T> {\n    const timeoutController =\n      this.timeoutMs && this.timeoutMs > 0 ? new AbortController() : undefined;\n    const timeout =\n      timeoutController && this.timeoutMs\n        ? setTimeout(() => timeoutController.abort(), this.timeoutMs)\n        : undefined;\n    const requestSignal = composeAbortSignals(\n      init.signal,\n      timeoutController?.signal,\n    );\n\n    try {\n      // Fail fast when the caller aborted before the request was sent.\n      requestSignal.signal?.throwIfAborted();\n      const response = await this.fetchImpl(\n        resolveSpeechUrl(this.baseUrl, path),\n        {\n          ...init,\n          method: 'POST',\n          headers: mergeHeaders(\n            { apiKey: this.apiKey, headers: this.headers },\n            init.headers,\n          ),\n          signal: requestSignal.signal,\n        },\n      );\n      await assertOk(response, adapterName, this.apiKey);\n      return await readResponse(response);\n    } finally {\n      if (timeout) {\n        clearTimeout(timeout);\n      }\n      requestSignal.cleanup();\n    }\n  }\n}\n\nfunction composeAbortSignals(\n  ...signals: Array<AbortSignal | undefined | null>\n): { signal?: AbortSignal; cleanup: () => void } {\n  const activeSignals = signals.filter((signal): signal is AbortSignal =>\n    Boolean(signal),\n  );\n\n  if (activeSignals.length === 0) {\n    return { cleanup: () => undefined };\n  }\n\n  if (activeSignals.length === 1) {\n    return { signal: activeSignals[0], cleanup: () => undefined };\n  }\n\n  const controller = new AbortController();\n  let cleanup = () => undefined;\n\n  const abortFrom = (signal: AbortSignal) => {\n    if (!controller.signal.aborted) {\n      controller.abort(signal.reason);\n    }\n    cleanup();\n  };\n\n  const onAbort = (event: Event) => {\n    abortFrom(event.target as AbortSignal);\n  };\n\n  cleanup = () => {\n    for (const signal of activeSignals) {\n      signal.removeEventListener('abort', onAbort);\n    }\n  };\n\n  for (const signal of activeSignals) {\n    if (signal.aborted) {\n      abortFrom(signal);\n      return { signal: controller.signal, cleanup: () => undefined };\n    }\n  }\n\n  for (const signal of activeSignals) {\n    signal.addEventListener('abort', onAbort, { once: true });\n  }\n\n  return { signal: controller.signal, cleanup };\n}\n\nasync function assertOk(\n  response: Response,\n  adapterName: string,\n  apiKey?: string,\n): Promise<void> {\n  if (response.ok) {\n    return;\n  }\n\n  const responseBody = await response.text().catch(() => undefined);\n  throw new SpeechProviderError(\n    adapterName,\n    `${adapterName} speech request failed with HTTP ${response.status}`,\n    {\n      status: response.status,\n      responseBody: redactSecret(responseBody, apiKey),\n      retryAfterMs: parseRetryAfter(response.headers.get('retry-after')),\n    },\n  );\n}\n\n/** Removes an API key from provider-supplied text before it is surfaced. */\nexport function redactSecret(\n  text: string | undefined,\n  secret: string | undefined,\n): string | undefined {\n  if (!text || !secret) {\n    return text;\n  }\n\n  return text.split(secret).join('[REDACTED]');\n}\n\nexport function appendOptionalFormValue(\n  form: FormData,\n  name: string,\n  value: unknown,\n): void {\n  if (value === undefined || value === null) {\n    return;\n  }\n\n  form.append(name, String(value));\n}\n\nexport function createHappyVerticalSynthesisForm(\n  request: SynthesisRequest,\n  defaultVoice?: string,\n): FormData {\n  const form = new FormData();\n  const voice =\n    request.voice && typeof request.voice !== 'string'\n      ? request.voice\n      : undefined;\n  form.set('text', request.text);\n  appendOptionalFormValue(\n    form,\n    'language',\n    request.language ?? voice?.language,\n  );\n  appendOptionalFormValue(\n    form,\n    'speaker',\n    voiceToString(request.voice, defaultVoice),\n  );\n  appendOptionalFormValue(form, 'speed', request.speed);\n\n  appendOptionalFormValue(form, 'voice_prompt', voice?.prompt);\n\n  return form;\n}\n\nexport function voiceToString(\n  voice:\n    | string\n    | { id?: string; name?: string; speakerId?: string }\n    | undefined,\n  fallback?: string,\n): string | undefined {\n  if (!voice) {\n    return fallback;\n  }\n\n  if (typeof voice === 'string') {\n    return voice;\n  }\n\n  return voice.id ?? voice.name ?? voice.speakerId ?? fallback;\n}\n\nexport function compactJson(\n  value: Record<string, unknown>,\n): Record<string, unknown> {\n  return Object.fromEntries(\n    Object.entries(value).filter(([, entryValue]) => entryValue !== undefined),\n  );\n}\n\nexport function contentTypeToFormat(contentType: string): string | undefined {\n  const match = /^audio\\/([^;]+)/.exec(contentType);\n  return match?.[1];\n}\n\nexport function arrayBufferFromBytes(bytes: Uint8Array): ArrayBuffer {\n  const copy = new Uint8Array(bytes.byteLength);\n  copy.set(bytes);\n  return copy.buffer;\n}\n\nexport function decodeBase64(base64: string): ArrayBuffer {\n  if (typeof Buffer !== 'undefined') {\n    return arrayBufferFromBytes(Buffer.from(base64, 'base64'));\n  }\n\n  const binary = atob(base64);\n  const bytes = new Uint8Array(binary.length);\n  for (let index = 0; index < binary.length; index += 1) {\n    bytes[index] = binary.charCodeAt(index);\n  }\n  return bytes.buffer;\n}\n\nfunction getNumber(\n  value: Record<string, unknown>,\n  ...keys: string[]\n): number | undefined {\n  for (const key of keys) {\n    const candidate = value[key];\n    if (typeof candidate === 'number') {\n      return candidate;\n    }\n  }\n\n  return undefined;\n}\n\nfunction getString(\n  value: Record<string, unknown>,\n  ...keys: string[]\n): string | undefined {\n  for (const key of keys) {\n    const candidate = value[key];\n    if (typeof candidate === 'string') {\n      return candidate;\n    }\n  }\n\n  return undefined;\n}\n\nexport function normalizeWordTimings(value: unknown): WordTiming[] | undefined {\n  if (!Array.isArray(value)) {\n    return undefined;\n  }\n\n  return value.flatMap((entry): WordTiming[] => {\n    if (!entry || typeof entry !== 'object') {\n      return [];\n    }\n\n    const record = entry as Record<string, unknown>;\n    const word = getString(record, 'word', 'text', 'token');\n    const startSeconds = getNumber(\n      record,\n      'startSeconds',\n      'start',\n      'start_time',\n    );\n    const endSeconds = getNumber(record, 'endSeconds', 'end', 'end_time');\n\n    if (!word || startSeconds === undefined || endSeconds === undefined) {\n      return [];\n    }\n\n    return [\n      {\n        word,\n        startSeconds,\n        endSeconds,\n        confidence: getNumber(record, 'confidence'),\n        speakerId: getString(record, 'speakerId', 'speaker', 'speaker_id'),\n      },\n    ];\n  });\n}\n\nexport function normalizeTranscriptSegments(\n  value: unknown,\n): TranscriptSegment[] | undefined {\n  if (!Array.isArray(value)) {\n    return undefined;\n  }\n\n  return value.flatMap((entry): TranscriptSegment[] => {\n    if (!entry || typeof entry !== 'object') {\n      return [];\n    }\n\n    const record = entry as Record<string, unknown>;\n    const text = getString(record, 'text', 'transcript');\n    if (!text) {\n      return [];\n    }\n\n    return [\n      {\n        text,\n        startSeconds: getNumber(record, 'startSeconds', 'start', 'start_time'),\n        endSeconds: getNumber(record, 'endSeconds', 'end', 'end_time'),\n        confidence: getNumber(record, 'confidence'),\n        speakerId: getString(record, 'speakerId', 'speaker', 'speaker_id'),\n      },\n    ];\n  });\n}\n\nexport async function readTranscriptResponse(\n  response: Response,\n  adapterName: string,\n): Promise<TranscriptResult> {\n  const contentType = response.headers.get('content-type') ?? '';\n\n  if (!contentType.includes('application/json')) {\n    const text = await response.text();\n    return {\n      text,\n      provider: adapterName,\n    };\n  }\n\n  const json = (await response.json()) as Record<string, unknown>;\n  const text = getString(json, 'text', 'transcript', 'transcription') ?? '';\n\n  return {\n    text,\n    language: getString(json, 'language', 'lang'),\n    durationSeconds: getNumber(json, 'durationSeconds', 'duration'),\n    words: normalizeWordTimings(\n      json.words ?? json.wordTimings ?? json.word_timings,\n    ),\n    segments: normalizeTranscriptSegments(json.segments),\n    provider: getString(json, 'provider') ?? adapterName,\n    model: getString(json, 'model'),\n    raw: json,\n  };\n}\n\nexport async function readSynthesizedSpeechResponse(\n  response: Response,\n  adapterName: string,\n): Promise<SynthesizedSpeech> {\n  const responseContentType =\n    response.headers.get('content-type') ?? 'application/octet-stream';\n\n  if (!responseContentType.includes('application/json')) {\n    const audio = await response.arrayBuffer();\n    const sampleRate = parseHeaderNumber(response, 'x-sample-rate');\n    return {\n      audio,\n      contentType: responseContentType,\n      format: contentTypeToFormat(responseContentType),\n      sampleRate,\n      provider: adapterName,\n    };\n  }\n\n  const json = (await response.json()) as Record<string, unknown>;\n  const contentType =\n    getString(json, 'contentType', 'content_type', 'mimeType', 'mime_type') ??\n    'application/octet-stream';\n  const encodedAudio = getString(\n    json,\n    'audio',\n    'audioContent',\n    'audio_content',\n  );\n\n  if (!encodedAudio) {\n    throw new SpeechProviderError(\n      adapterName,\n      `${adapterName} JSON speech response did not include base64 audio`,\n    );\n  }\n\n  return {\n    audio: decodeBase64(encodedAudio),\n    contentType,\n    format:\n      getString(json, 'format', 'response_format') ??\n      contentTypeToFormat(contentType),\n    sampleRate: getNumber(json, 'sampleRate', 'sample_rate'),\n    channels: getNumber(json, 'channels'),\n    durationSeconds: getNumber(json, 'durationSeconds', 'duration'),\n    words: normalizeWordTimings(\n      json.words ?? json.wordTimings ?? json.word_timings,\n    ),\n    provider: getString(json, 'provider') ?? adapterName,\n    model: getString(json, 'model'),\n    raw: json,\n  };\n}\n\nfunction parseHeaderNumber(\n  response: Response,\n  headerName: string,\n): number | undefined {\n  const value = response.headers.get(headerName);\n  if (!value) {\n    return undefined;\n  }\n\n  const parsed = Number(value);\n  return Number.isFinite(parsed) ? parsed : undefined;\n}\n","/**\n * Credential handling shared by realtime (WebSocket) adapters.\n *\n * Node callers authenticate with a long-lived `apiKey` in the\n * `Authorization` header. Browsers cannot set WebSocket headers and must never\n * hold a long-lived key, so they pass a short-lived `clientSecret`, which is\n * sent as a WebSocket subprotocol.\n */\n\nimport { SpeechConfigurationError } from './errors.js';\nimport type { RealtimeConnection } from './realtime-session.js';\nimport type { StreamingClientSecret } from './streaming-types.js';\nimport { isBrowserRuntime } from './websocket.js';\n\n/** Subprotocol prefix that carries a client secret (OpenAI's convention). */\nexport const CLIENT_SECRET_SUBPROTOCOL_PREFIX = 'openai-insecure-api-key.';\n\n/**\n * Long-lived OpenAI-style keys (`sk-…`, `sk-proj-…`, `sk-svcacct-…`). A\n * browser `clientSecret` that looks like one is refused: browsers accept only\n * short-lived tokens such as OpenAI's `ek_…` client secrets.\n */\nconst LONG_LIVED_KEY = /^sk-/;\n\n/** Throws when an adapter is given a long-lived `apiKey` in a browser. */\nexport function assertNoBrowserApiKey(\n  adapter: string,\n  apiKey: string | undefined,\n): void {\n  if (apiKey && isBrowserRuntime()) {\n    throw new SpeechConfigurationError(\n      `${adapter} refuses apiKey in a browser; mint a short-lived clientSecret on your server instead`,\n      adapter,\n    );\n  }\n}\n\n/** Resolves a static or per-session client secret and validates its shape. */\nexport async function resolveClientSecret(\n  adapter: string,\n  secret: StreamingClientSecret | undefined,\n): Promise<string | undefined> {\n  const value = typeof secret === 'function' ? await secret() : secret;\n  const trimmed = value?.trim();\n  if (trimmed === undefined) {\n    return undefined;\n  }\n  if (!/^[\\x21-\\x7e]+$/.test(trimmed)) {\n    throw new SpeechConfigurationError(\n      'clientSecret must be a non-empty token without spaces',\n      adapter,\n    );\n  }\n  if (isBrowserRuntime() && LONG_LIVED_KEY.test(trimmed)) {\n    throw new SpeechConfigurationError(\n      'clientSecret looks like a long-lived API key; browsers accept only short-lived tokens minted by your server (see createStreamingClientSecret)',\n      adapter,\n    );\n  }\n  return trimmed;\n}\n\nexport interface RealtimeConnectorOptions {\n  adapter: string;\n  url: string;\n  apiKey?: string;\n  headers: Record<string, string>;\n  clientSecret?: StreamingClientSecret;\n  /** Subprotocols to offer, given the resolved client secret (if any). */\n  protocols: (secret: string | undefined) => string[];\n}\n\n/**\n * Builds the per-session `connect` callback: refuses handshake headers in a\n * browser, resolves the client secret, and sets `Authorization` from\n * `apiKey` unless a header already supplies it.\n */\nexport function realtimeConnector(\n  options: RealtimeConnectorOptions,\n): () => Promise<RealtimeConnection> {\n  if (isBrowserRuntime() && Object.keys(options.headers).length > 0) {\n    throw new SpeechConfigurationError(\n      'Browsers cannot send WebSocket headers; remove `headers` or connect from a server',\n      options.adapter,\n    );\n  }\n\n  return async () => {\n    const secret = await resolveClientSecret(\n      options.adapter,\n      options.clientSecret,\n    );\n    const headers = { ...options.headers };\n    // Gateway credentials (e.g. `x-bf-vk`, an explicit Authorization) are\n    // redacted from surfaced provider text just like the API key.\n    const secrets: string[] = headerSecrets(headers);\n    if (secret) {\n      secrets.push(secret);\n    }\n    if (options.apiKey) {\n      secrets.push(options.apiKey);\n      if (!hasHeader(headers, 'authorization')) {\n        headers.authorization = `Bearer ${options.apiKey}`;\n      }\n    }\n    return {\n      url: options.url,\n      protocols: options.protocols(secret),\n      headers,\n      secrets,\n    };\n  };\n}\n\n/** Values of other headers shorter than this are not treated as credentials. */\nconst MIN_HEADER_SECRET_LENGTH = 8;\n\n/**\n * Header names that carry credentials (`authorization`, `x-api-key`,\n * `x-bf-vk`, `*-token`, `*-secret`, …). Their values are redacted at any\n * length.\n */\nconst CREDENTIAL_HEADER = /auth|key|token|secret|password|credential|-vk$/i;\n\n/**\n * Values of caller-supplied headers to redact from provider error text.\n * Credential-named headers are redacted whatever their length; for\n * `Authorization`, the credential after the scheme is included too. Other\n * headers may also carry tenant secrets, so their values are redacted when\n * they are at least 8 characters long, which keeps short values such as\n * `1` from mangling messages.\n */\nexport function headerSecrets(headers: HeadersInit | undefined): string[] {\n  const secrets: string[] = [];\n  for (const [key, value] of Object.entries(mergeHeaderRecords(headers))) {\n    const credential = CREDENTIAL_HEADER.test(key);\n    const candidates = [value];\n    if (key.toLowerCase() === 'authorization') {\n      candidates.push(value.replace(/^\\S+\\s+/, ''));\n    }\n    for (const candidate of candidates) {\n      const trimmed = candidate.trim();\n      const long = trimmed.length >= MIN_HEADER_SECRET_LENGTH;\n      if (trimmed && (credential || long) && !secrets.includes(trimmed)) {\n        secrets.push(trimmed);\n      }\n    }\n  }\n  // Longest first, so a full `Bearer <token>` is replaced before its token.\n  return secrets.sort((left, right) => right.length - left.length);\n}\n\nexport function mergeHeaderRecords(\n  ...sets: Array<HeadersInit | undefined>\n): Record<string, string> {\n  const merged: Record<string, string> = {};\n  for (const set of sets) {\n    if (set) {\n      new Headers(set).forEach((value, key) => {\n        merged[key] = value;\n      });\n    }\n  }\n  return merged;\n}\n\nfunction hasHeader(headers: Record<string, string>, name: string): boolean {\n  return Object.keys(headers).some((key) => key.toLowerCase() === name);\n}\n","/**\n * Raw-audio helpers for streaming adapters: byte views, base64 encoding,\n * duration from byte counts, and WAV/`audio/L16` unwrapping for the\n * record-then-send wrapper. Nothing here decodes compressed audio or resamples.\n */\n\nimport { normalizeMimeType } from './audio.js';\nimport { SpeechConfigurationError } from './errors.js';\nimport type {\n  StreamingAudioChunk,\n  StreamingAudioEncoding,\n  StreamingAudioFormat,\n} from './streaming-types.js';\n\nconst ENCODING_BYTES_PER_SAMPLE: Record<StreamingAudioEncoding, number> = {\n  pcm16: 2,\n  g711_ulaw: 1,\n  g711_alaw: 1,\n};\n\n/** Returns a byte view over a chunk without copying. */\nexport function chunkToBytes(\n  chunk: StreamingAudioChunk,\n  adapter?: string,\n): Uint8Array {\n  if (chunk instanceof Uint8Array) {\n    return chunk;\n  }\n  if (chunk instanceof ArrayBuffer) {\n    return new Uint8Array(chunk);\n  }\n  if (ArrayBuffer.isView(chunk)) {\n    return new Uint8Array(chunk.buffer, chunk.byteOffset, chunk.byteLength);\n  }\n  throw new SpeechConfigurationError(\n    'Streaming audio chunks must be Uint8Array, ArrayBuffer, or an ArrayBuffer view',\n    adapter,\n  );\n}\n\nexport function encodeBase64(bytes: Uint8Array): string {\n  if (typeof Buffer !== 'undefined') {\n    return Buffer.from(\n      bytes.buffer,\n      bytes.byteOffset,\n      bytes.byteLength,\n    ).toString('base64');\n  }\n\n  let binary = '';\n  const step = 0x8000;\n  for (let index = 0; index < bytes.length; index += step) {\n    binary += String.fromCharCode(...bytes.subarray(index, index + step));\n  }\n  return btoa(binary);\n}\n\n/** Bytes per second of audio in `format`. */\nexport function bytesPerSecond(format: StreamingAudioFormat): number {\n  return (\n    format.sampleRate *\n    format.channels *\n    ENCODING_BYTES_PER_SAMPLE[format.encoding]\n  );\n}\n\n/** Audio duration in seconds for `bytes` of `format`, rounded to milliseconds. */\nexport function audioSecondsForBytes(\n  bytes: number,\n  format: StreamingAudioFormat,\n): number {\n  return Math.round((bytes / bytesPerSecond(format)) * 1000) / 1000;\n}\n\nexport interface RawAudio {\n  bytes: Uint8Array;\n  /** Format fields the input declared (WAV header, MIME parameters, or `AudioInput`). */\n  format: Partial<StreamingAudioFormat>;\n}\n\nconst WAV_TYPES = new Set([\n  'audio/wav',\n  'audio/wave',\n  'audio/x-wav',\n  'audio/vnd.wave',\n]);\nconst RAW_PCM_TYPES = new Set([\n  'audio/pcm',\n  'audio/l16',\n  'audio/raw',\n  'audio/x-raw',\n]);\n/** Untyped bytes: the streaming transcriber's default format applies. */\nconst UNTYPED = 'application/octet-stream';\nconst MULAW_TYPES = new Set(['audio/pcmu', 'audio/basic', 'audio/x-mulaw']);\nconst ALAW_TYPES = new Set(['audio/pcma', 'audio/x-alaw']);\n\n/**\n * Interprets a finished recording as raw audio for a streaming session.\n * Accepts PCM/G.711 WAV files, `audio/pcm`/`audio/L16` (with optional `rate`\n * and `channels` parameters), `audio/pcmu`/`audio/pcma`, and untyped bytes.\n * Compressed containers are rejected with a `SpeechConfigurationError`.\n */\nexport function unwrapRawAudio(\n  bytes: Uint8Array,\n  mimeType: string,\n  adapter: string,\n): RawAudio {\n  const essence = normalizeMimeType(mimeType) ?? 'application/octet-stream';\n  const params = parseMimeParameters(mimeType);\n  const rate = parsePositiveInt(params.rate);\n  const channels = parsePositiveInt(params.channels);\n\n  if (WAV_TYPES.has(essence) || isRiffWave(bytes)) {\n    return parseWav(bytes, adapter);\n  }\n  if (essence === UNTYPED) {\n    // No encoding: `start()` keeps the adapter's `audioFormat` (e.g. G.711).\n    return { bytes, format: compact({ sampleRate: rate, channels }) };\n  }\n  if (RAW_PCM_TYPES.has(essence)) {\n    if (essence === 'audio/l16' && bytes.byteLength % 2 !== 0) {\n      throw new SpeechConfigurationError(\n        `audio/L16 payload has an odd byte length (${bytes.byteLength}); 16-bit samples need an even number of bytes`,\n        adapter,\n      );\n    }\n    return {\n      // RFC 2586: audio/L16 samples are big-endian; pcm16 is little-endian.\n      bytes: essence === 'audio/l16' ? swapBytePairs(bytes) : bytes,\n      format: compact({ encoding: 'pcm16', sampleRate: rate, channels }),\n    };\n  }\n  if (MULAW_TYPES.has(essence)) {\n    return {\n      bytes,\n      format: compact({ encoding: 'g711_ulaw', sampleRate: rate, channels }),\n    };\n  }\n  if (ALAW_TYPES.has(essence)) {\n    return {\n      bytes,\n      format: compact({ encoding: 'g711_alaw', sampleRate: rate, channels }),\n    };\n  }\n\n  throw new SpeechConfigurationError(\n    `${adapter} streams raw audio and cannot decode ${essence}; send PCM WAV or raw PCM, or use an HTTP transcriber for compressed recordings`,\n    adapter,\n  );\n}\n\n/** Returns a copy with each 16-bit sample's bytes swapped. */\nfunction swapBytePairs(bytes: Uint8Array): Uint8Array {\n  const swapped = new Uint8Array(bytes.byteLength);\n  for (let index = 0; index + 1 < bytes.byteLength; index += 2) {\n    swapped[index] = bytes[index + 1] ?? 0;\n    swapped[index + 1] = bytes[index] ?? 0;\n  }\n  return swapped;\n}\n\nfunction isRiffWave(bytes: Uint8Array): boolean {\n  return (\n    bytes.byteLength >= 12 &&\n    ascii(bytes, 0, 4) === 'RIFF' &&\n    ascii(bytes, 8, 4) === 'WAVE'\n  );\n}\n\nfunction parseWav(bytes: Uint8Array, adapter: string): RawAudio {\n  if (!isRiffWave(bytes)) {\n    throw new SpeechConfigurationError('Invalid WAV header', adapter);\n  }\n\n  const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);\n  let offset = 12;\n  let format: Partial<StreamingAudioFormat> | undefined;\n\n  while (offset + 8 <= bytes.byteLength) {\n    const id = ascii(bytes, offset, 4);\n    const size = view.getUint32(offset + 4, true);\n    const body = offset + 8;\n\n    if (id === 'fmt ' && body + 16 <= bytes.byteLength) {\n      const audioFormat = view.getUint16(body, true);\n      const channels = view.getUint16(body + 2, true);\n      const sampleRate = view.getUint32(body + 4, true);\n      const bitsPerSample = view.getUint16(body + 14, true);\n      format = {\n        encoding: wavEncoding(audioFormat, bitsPerSample, adapter),\n        sampleRate,\n        channels,\n      };\n    } else if (id === 'data') {\n      if (!format) {\n        throw new SpeechConfigurationError(\n          'WAV data chunk precedes its fmt chunk',\n          adapter,\n        );\n      }\n      const end = Math.min(bytes.byteLength, body + size);\n      return { bytes: bytes.subarray(body, end), format };\n    }\n\n    // Chunks are padded to an even size.\n    offset = body + size + (size % 2);\n  }\n\n  throw new SpeechConfigurationError('WAV file has no data chunk', adapter);\n}\n\nfunction wavEncoding(\n  audioFormat: number,\n  bitsPerSample: number,\n  adapter: string,\n): StreamingAudioEncoding {\n  // 0xfffe is WAVE_FORMAT_EXTENSIBLE; its common PCM case is treated as PCM.\n  if ((audioFormat === 1 || audioFormat === 0xfffe) && bitsPerSample === 16) {\n    return 'pcm16';\n  }\n  if (audioFormat === 7 && bitsPerSample === 8) {\n    return 'g711_ulaw';\n  }\n  if (audioFormat === 6 && bitsPerSample === 8) {\n    return 'g711_alaw';\n  }\n  throw new SpeechConfigurationError(\n    `Unsupported WAV encoding (format ${audioFormat}, ${bitsPerSample}-bit); expected 16-bit PCM or 8-bit G.711`,\n    adapter,\n  );\n}\n\nfunction ascii(bytes: Uint8Array, offset: number, length: number): string {\n  return String.fromCharCode(...bytes.subarray(offset, offset + length));\n}\n\nfunction parseMimeParameters(mimeType: string): Record<string, string> {\n  const params: Record<string, string> = {};\n  for (const part of mimeType.split(';').slice(1)) {\n    const [key, value] = part.split('=');\n    if (key && value) {\n      params[key.trim().toLowerCase()] = value.trim();\n    }\n  }\n  return params;\n}\n\nfunction parsePositiveInt(value: string | undefined): number | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const parsed = Number.parseInt(value, 10);\n  return Number.isFinite(parsed) && parsed > 0 ? parsed : undefined;\n}\n\nfunction compact(\n  format: Partial<StreamingAudioFormat>,\n): Partial<StreamingAudioFormat> {\n  return Object.fromEntries(\n    Object.entries(format).filter(([, value]) => value !== undefined),\n  ) as Partial<StreamingAudioFormat>;\n}\n","/**\n * Per-turn audio limits for realtime sessions.\n *\n * Some providers cap the audio of one turn rather than of a session: vLLM's\n * realtime endpoint keeps a turn's audio and generated tokens in one model\n * context (`max_model_len`) and clears it after each final commit. A\n * {@link TurnSplitter} measures the audio of the current turn from byte counts\n * and the negotiated format, and picks where to roll over to a new turn: at\n * the first quiet stretch inside a window before the limit, or at the limit\n * itself when no quiet stretch turns up. The silence check is a plain RMS\n * energy test, not a voice-activity model.\n */\n\nimport { SpeechConfigurationError } from './errors.js';\nimport { bytesPerSecond } from './pcm.js';\nimport type {\n  StreamingAudioEncoding,\n  StreamingAudioFormat,\n  StreamingRolloverOptions,\n  StreamingSessionSettings,\n  StreamingTurnLimit,\n} from './streaming-types.js';\n\n/** Default per-turn audio cap for providers that need one (`voxtral-realtime`). */\nexport const DEFAULT_MAX_TURN_SECONDS = 270;\n/** How long before the cap the rollover starts looking for silence. */\nexport const DEFAULT_ROLLOVER_WINDOW_SECONDS = 15;\n/** RMS level (0-1 of full scale, about -40 dBFS) below which audio is quiet. */\nexport const DEFAULT_ROLLOVER_SILENCE_THRESHOLD = 0.01;\n/** Quiet audio needed before the rollover commits. */\nexport const DEFAULT_ROLLOVER_MIN_SILENCE_MS = 300;\n/** Energy is measured over frames of this length. */\nconst ANALYSIS_FRAME_MS = 20;\n\nconst BYTES_PER_SAMPLE: Record<StreamingAudioEncoding, number> = {\n  pcm16: 2,\n  g711_ulaw: 1,\n  g711_alaw: 1,\n};\n\n/** A resolved per-turn limit; `undefined` means turns are unbounded. */\nexport interface TurnLimit {\n  maxTurnSeconds: number;\n  /** `false`: writes past the cap reject with `SPEECH_TURN_TOO_LONG`. */\n  rollover: boolean;\n  windowSeconds: number;\n  silenceThreshold: number;\n  minSilenceMs: number;\n}\n\n/**\n * Resolves the turn limit from settings, highest priority first (session\n * options, then adapter options, which already carry environment values),\n * falling back to `defaultMaxTurnSeconds`. Returns `undefined` when no limit\n * applies or `maxTurnSeconds` is `Infinity`.\n */\nexport function resolveTurnLimit(\n  adapter: string,\n  defaultMaxTurnSeconds: number | undefined,\n  ...settings: Array<\n    Pick<StreamingSessionSettings, 'maxTurnSeconds' | 'rollover'> | undefined\n  >\n): TurnLimit | undefined {\n  const maxTurnSeconds =\n    settings.find((entry) => entry?.maxTurnSeconds !== undefined)\n      ?.maxTurnSeconds ?? defaultMaxTurnSeconds;\n  if (maxTurnSeconds === undefined) {\n    return undefined;\n  }\n  if (\n    typeof maxTurnSeconds !== 'number' ||\n    Number.isNaN(maxTurnSeconds) ||\n    maxTurnSeconds <= 0\n  ) {\n    throw new SpeechConfigurationError(\n      `maxTurnSeconds must be a positive number of seconds or Infinity (got ${String(maxTurnSeconds)})`,\n      adapter,\n    );\n  }\n  if (maxTurnSeconds === Number.POSITIVE_INFINITY) {\n    return undefined;\n  }\n\n  // On/off comes from the first layer that sets `rollover`; each tuning\n  // field from the first layer whose `rollover` object sets that field, so a\n  // `rollover: true` session keeps the adapter's tuning.\n  const rollover = settings.find(\n    (entry) => entry?.rollover !== undefined,\n  )?.rollover;\n  const tuning = <K extends keyof StreamingRolloverOptions>(key: K) =>\n    settings\n      .map((entry) => entry?.rollover)\n      .find(\n        (value): value is StreamingRolloverOptions =>\n          typeof value === 'object' &&\n          value !== null &&\n          value[key] !== undefined,\n      )?.[key];\n  const limit: TurnLimit = {\n    maxTurnSeconds,\n    rollover: rollover !== false,\n    windowSeconds: tuning('windowSeconds') ?? DEFAULT_ROLLOVER_WINDOW_SECONDS,\n    silenceThreshold:\n      tuning('silenceThreshold') ?? DEFAULT_ROLLOVER_SILENCE_THRESHOLD,\n    minSilenceMs: tuning('minSilenceMs') ?? DEFAULT_ROLLOVER_MIN_SILENCE_MS,\n  };\n  assertNonNegative(adapter, 'rollover.windowSeconds', limit.windowSeconds);\n  assertNonNegative(adapter, 'rollover.minSilenceMs', limit.minSilenceMs);\n  assertNonNegative(\n    adapter,\n    'rollover.silenceThreshold',\n    limit.silenceThreshold,\n  );\n  if (limit.silenceThreshold > 1) {\n    throw new SpeechConfigurationError(\n      `rollover.silenceThreshold is an RMS level from 0 to 1 (got ${limit.silenceThreshold})`,\n      adapter,\n    );\n  }\n  return limit;\n}\n\nfunction assertNonNegative(adapter: string, name: string, value: unknown) {\n  if (typeof value !== 'number' || !Number.isFinite(value) || value < 0) {\n    throw new SpeechConfigurationError(\n      `${name} must be a non-negative finite number (got ${String(value)})`,\n      adapter,\n    );\n  }\n}\n\n/** Bytes of one sample frame (one sample per channel). */\nexport function blockAlign(format: StreamingAudioFormat): number {\n  return BYTES_PER_SAMPLE[format.encoding] * format.channels;\n}\n\n/** Whole sample frames in `seconds` of audio, as bytes (rounded down). */\nexport function bytesForSeconds(\n  seconds: number,\n  format: StreamingAudioFormat,\n): number {\n  const align = blockAlign(format);\n  // The epsilon keeps decimal seconds such as 0.3 from losing a frame.\n  return Math.floor((seconds * bytesPerSecond(format)) / align + 1e-9) * align;\n}\n\n/** Digital silence in `format`: zero samples, encoded. */\nexport function silence(\n  bytes: number,\n  format: StreamingAudioFormat,\n): Uint8Array {\n  const fill =\n    format.encoding === 'g711_ulaw'\n      ? 0xff\n      : format.encoding === 'g711_alaw'\n        ? 0xd5\n        : 0;\n  return new Uint8Array(bytes).fill(fill);\n}\n\n/** Bytes of audio one turn may hold: whole sample frames, at least one. */\nexport function turnCapBytes(\n  limit: TurnLimit,\n  format: StreamingAudioFormat,\n): number {\n  return Math.max(\n    blockAlign(format),\n    bytesForSeconds(limit.maxTurnSeconds, format),\n  );\n}\n\n/** The public view of a resolved limit for a session format. */\nexport function describeTurnLimit(\n  limit: TurnLimit | undefined,\n  format: StreamingAudioFormat,\n): StreamingTurnLimit | undefined {\n  return (\n    limit && {\n      maxTurnSeconds: limit.maxTurnSeconds,\n      rollover: limit.rollover,\n      maxTurnBytes: turnCapBytes(limit, format),\n    }\n  );\n}\n\n/**\n * Tracks the audio of the current turn and finds rollover points.\n *\n * Feed every chunk in write order to {@link split} (rollover on) or\n * {@link add} (rollover off), and call {@link reset} whenever the turn ends\n * for another reason (a manual commit).\n */\nexport class TurnSplitter {\n  /** Bytes of the current turn so far. */\n  turnBytes = 0;\n  readonly capBytes: number;\n  private readonly windowStartBytes: number;\n  private readonly frameBytes: number;\n  private readonly minSilenceBytes: number;\n  private readonly threshold: number;\n  private readonly encoding: StreamingAudioEncoding;\n  private readonly sampleBytes: number;\n  // Energy of the analysis frame being filled.\n  private frameFill = 0;\n  private frameSumSquares = 0;\n  private frameSamples = 0;\n  /** Trailing bytes of a sample split across chunks. */\n  private carry: number[] = [];\n  /** Consecutive quiet bytes ending at the last complete frame. */\n  private quietRun = 0;\n\n  constructor(limit: TurnLimit, format: StreamingAudioFormat) {\n    const align = blockAlign(format);\n    this.capBytes = turnCapBytes(limit, format);\n    this.windowStartBytes = Math.max(\n      0,\n      this.capBytes - bytesForSeconds(limit.windowSeconds, format),\n    );\n    this.frameBytes = Math.max(\n      align,\n      bytesForSeconds(ANALYSIS_FRAME_MS / 1000, format),\n    );\n    this.minSilenceBytes = Math.max(\n      this.frameBytes,\n      bytesForSeconds(limit.minSilenceMs / 1000, format),\n    );\n    this.threshold = limit.silenceThreshold;\n    this.encoding = format.encoding;\n    this.sampleBytes = BYTES_PER_SAMPLE[format.encoding];\n  }\n\n  /** Ends the current turn: the next byte starts a new one. */\n  reset(): void {\n    this.turnBytes = 0;\n    this.frameFill = 0;\n    this.frameSumSquares = 0;\n    this.frameSamples = 0;\n    this.carry = [];\n    this.quietRun = 0;\n  }\n\n  /** Whether `bytes` more would push the current turn past the cap. */\n  wouldOverflow(bytes: number): boolean {\n    return this.turnBytes + bytes > this.capBytes;\n  }\n\n  /** Counts `bytes` toward the current turn without analysing them. */\n  add(bytes: number): void {\n    this.turnBytes += bytes;\n  }\n\n  /**\n   * Feeds one chunk and returns the offsets inside it where the turn should\n   * end (each in `1..chunk.byteLength`, ascending). The splitter resets at\n   * each offset, so bytes after it count toward the next turn.\n   */\n  split(chunk: Uint8Array): number[] {\n    const cuts: number[] = [];\n    let offset = 0;\n    while (offset < chunk.byteLength) {\n      if (this.turnBytes < this.windowStartBytes) {\n        // Before the window only the length matters.\n        const skip = Math.min(\n          chunk.byteLength - offset,\n          this.windowStartBytes - this.turnBytes,\n        );\n        offset += skip;\n        this.turnBytes += skip;\n        // With no window (`windowSeconds: 0`) the window starts at the cap.\n        if (this.turnBytes >= this.capBytes) {\n          cuts.push(offset);\n          this.reset();\n        }\n        continue;\n      }\n\n      const take = Math.min(\n        chunk.byteLength - offset,\n        this.frameBytes - this.frameFill,\n        this.capBytes - this.turnBytes,\n      );\n      this.measure(chunk, offset, take);\n      offset += take;\n      this.turnBytes += take;\n      this.frameFill += take;\n\n      let cut = false;\n      if (this.frameFill >= this.frameBytes) {\n        const quiet = this.frameIsQuiet();\n        this.quietRun = quiet ? this.quietRun + this.frameFill : 0;\n        this.frameFill = 0;\n        this.frameSumSquares = 0;\n        this.frameSamples = 0;\n        cut = this.quietRun >= this.minSilenceBytes;\n      }\n      if (cut || this.turnBytes >= this.capBytes) {\n        cuts.push(offset);\n        this.reset();\n      }\n    }\n    return cuts;\n  }\n\n  private frameIsQuiet(): boolean {\n    if (this.frameSamples === 0) {\n      return false;\n    }\n    const rms = Math.sqrt(this.frameSumSquares / this.frameSamples) / 32_768;\n    return rms < this.threshold;\n  }\n\n  private measure(chunk: Uint8Array, start: number, length: number): void {\n    const end = start + length;\n    let index = start;\n    // Complete a sample left over from the previous chunk.\n    while (this.carry.length > 0 && index < end) {\n      this.carry.push(chunk[index] ?? 0);\n      index += 1;\n      if (this.carry.length === this.sampleBytes) {\n        this.addSample(this.carry);\n        this.carry = [];\n      }\n    }\n    for (; index + this.sampleBytes <= end; index += this.sampleBytes) {\n      this.addSampleAt(chunk, index);\n    }\n    for (; index < end; index += 1) {\n      this.carry.push(chunk[index] ?? 0);\n    }\n  }\n\n  private addSample(bytes: number[]): void {\n    this.addLinear(\n      this.encoding === 'pcm16'\n        ? pcm16Sample(bytes[0] ?? 0, bytes[1] ?? 0)\n        : this.decodeG711(bytes[0] ?? 0),\n    );\n  }\n\n  private addSampleAt(chunk: Uint8Array, index: number): void {\n    this.addLinear(\n      this.encoding === 'pcm16'\n        ? pcm16Sample(chunk[index] ?? 0, chunk[index + 1] ?? 0)\n        : this.decodeG711(chunk[index] ?? 0),\n    );\n  }\n\n  private addLinear(sample: number): void {\n    this.frameSumSquares += sample * sample;\n    this.frameSamples += 1;\n  }\n\n  private decodeG711(byte: number): number {\n    return this.encoding === 'g711_ulaw' ? decodeMuLaw(byte) : decodeALaw(byte);\n  }\n}\n\n/** Signed 16-bit little-endian sample. */\nfunction pcm16Sample(low: number, high: number): number {\n  const value = low | (high << 8);\n  return value >= 0x8000 ? value - 0x10000 : value;\n}\n\n/** G.711 mu-law byte to a 16-bit linear sample (ITU-T G.711). */\nexport function decodeMuLaw(byte: number): number {\n  const value = ~byte & 0xff;\n  const exponent = (value >> 4) & 0x07;\n  const mantissa = value & 0x0f;\n  const magnitude = (((mantissa << 3) + 0x84) << exponent) - 0x84;\n  return value & 0x80 ? -magnitude : magnitude;\n}\n\n/** G.711 A-law byte to a 16-bit linear sample (ITU-T G.711). */\nexport function decodeALaw(byte: number): number {\n  const value = byte ^ 0x55;\n  const exponent = (value >> 4) & 0x07;\n  const mantissa = value & 0x0f;\n  const magnitude =\n    exponent === 0\n      ? (mantissa << 4) + 8\n      : ((mantissa << 4) + 0x108) << (exponent - 1);\n  return value & 0x80 ? magnitude : -magnitude;\n}\n","/**\n * Protocol-agnostic realtime transcription session over a WebSocket.\n *\n * The session owns the socket lifecycle, write queue and backpressure,\n * timeouts, turn bookkeeping, events, and the final `TranscriptResult`. A\n * {@link RealtimeProtocol} supplies the wire format, so a sibling adapter\n * (for example a vLLM-served Voxtral realtime endpoint) only has to map its\n * own messages.\n */\n\nimport {\n  SpeechConfigurationError,\n  SpeechError,\n  SpeechProviderError,\n} from './errors.js';\nimport { redactSecret } from './http.js';\nimport {\n  audioSecondsForBytes,\n  bytesPerSecond,\n  chunkToBytes,\n  encodeBase64,\n} from './pcm.js';\nimport type {\n  StreamingAudioChunk,\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionEventName,\n  StreamingSessionEvents,\n  StreamingSessionListener,\n  StreamingSessionState,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport {\n  blockAlign,\n  silence,\n  type TurnLimit,\n  TurnSplitter,\n} from './turn-limit.js';\nimport type { TranscriptResult, TranscriptSegment } from './types.js';\nimport {\n  audioSecondsFromProviderUsage,\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from './usage.js';\nimport {\n  messageDataToText,\n  type SpeechWebSocket,\n  type SpeechWebSocketCloseEvent,\n  type SpeechWebSocketFactory,\n} from './websocket.js';\n\nexport const DEFAULT_STREAMING_TIMEOUT_MS = 30_000;\nexport const DEFAULT_STREAMING_CONNECT_TIMEOUT_MS = 10_000;\nexport const DEFAULT_STREAMING_HIGH_WATER_MARK = 1024 * 1024;\nexport const DEFAULT_STREAMING_MAX_BUFFERED_BYTES = 16 * 1024 * 1024;\nconst DRAIN_POLL_MS = 10;\n\n/** Normalised events a protocol parser returns for one server message. */\nexport type RealtimeProtocolEvent =\n  | { type: 'session'; sessionId?: string }\n  /** The provider committed a turn (from a client commit or server VAD). */\n  | { type: 'committed'; itemId?: string }\n  /** The provider rejected a commit because its buffer was empty. Not fatal. */\n  | { type: 'commit_empty' }\n  | { type: 'partial'; itemId?: string; delta: string }\n  | {\n      type: 'final';\n      itemId?: string;\n      text: string;\n      providerUsage?: Record<string, unknown>;\n      raw?: unknown;\n      /**\n       * `false` when the provider reports that the turn consumed no audio\n       * (vLLM: `usage.prompt_tokens <= 1`). A turn that was sent audio but\n       * consumed none means the provider's turn state is out of sync, and the\n       * session fails instead of silently dropping that audio.\n       */\n      audioConsumed?: boolean;\n    }\n  | { type: 'speech_started'; itemId?: string; audioMs?: number }\n  | { type: 'speech_stopped'; itemId?: string; audioMs?: number }\n  /** A fatal provider error. */\n  | { type: 'error'; message: string; code?: string; raw?: unknown };\n\n/** Resolved per-session settings handed to the protocol. */\nexport interface RealtimeSessionConfig {\n  model: string;\n  format: StreamingAudioFormat;\n  turnDetection: StreamingTurnDetection;\n  language?: string;\n  prompt?: string;\n}\n\n/** Wire format of one realtime provider. */\nexport interface RealtimeProtocol {\n  /** Adapter name used for errors and usage, e.g. `openai-realtime`. */\n  readonly provider: string;\n  /**\n   * How a client commit is acknowledged: by a `committed` event (OpenAI), or\n   * only by the turn's `final` event (servers without a commit ack).\n   */\n  readonly commitAck: 'committed' | 'final';\n  /**\n   * Whether `end()` commits only when uncommitted audio remains\n   * (`if-audio`), or always sends a final commit (`always`, for servers that\n   * need an explicit end-of-stream marker).\n   */\n  readonly endCommit: 'if-audio' | 'always';\n  /**\n   * Default per-turn audio cap in seconds, for providers whose model context\n   * covers one turn (vLLM resets it after each final commit). Adapters pass\n   * it to `resolveTurnLimit()`; a session with a turn limit rolls over to a\n   * new turn before the cap, or rejects writes past it. Unset: no default.\n   */\n  readonly maxTurnSeconds?: number;\n  /**\n   * Shortest audio, in seconds, the provider accepts in one commit (OpenAI:\n   * 100 ms; shorter commits are rejected as empty). A session with a turn\n   * limit pads a shorter, non-empty turn with silence before committing it,\n   * so the tail left after a rollover is transcribed instead of dropped.\n   */\n  readonly minCommitSeconds?: number;\n  /** Messages sent once the socket opens (session configuration). */\n  sessionMessages(config: RealtimeSessionConfig): unknown[];\n  /** One audio chunk, already base64-encoded. */\n  appendMessage(base64Audio: string): unknown;\n  /** Ends the current turn. `final` is true for the commit sent by `end()`. */\n  commitMessage(options: { final: boolean }): unknown;\n  /**\n   * Opens a turn before its first audio, for servers that only start\n   * generating after an explicit start signal (vLLM's non-final\n   * `input_audio_buffer.commit`). When set, the session sends it before the\n   * first append of every turn, and holds audio written after a commit until\n   * every outstanding turn has been acknowledged, so a server that resets its\n   * buffer between turns never drops audio.\n   */\n  turnStartMessage?(): unknown;\n  /** Maps one parsed JSON server message to zero or more events. */\n  parse(message: unknown): RealtimeProtocolEvent[];\n}\n\n/** Where and how to open the socket; resolved per session. */\nexport interface RealtimeConnection {\n  url: string;\n  protocols: string[];\n  headers: Record<string, string>;\n  /** Credentials to redact from surfaced provider text. */\n  secrets: string[];\n}\n\nexport interface RealtimeSessionInit {\n  protocol: RealtimeProtocol;\n  config: RealtimeSessionConfig;\n  /** Resolves the connection (may mint a short-lived token). */\n  connect: () => Promise<RealtimeConnection>;\n  createWebSocket: SpeechWebSocketFactory;\n  timeoutMs: number;\n  connectTimeoutMs: number;\n  highWaterMark: number;\n  maxBufferedBytes: number;\n  signal?: AbortSignal;\n  /** Called in order with the aggregated usage when `end()` succeeds. */\n  onUsage: Array<SpeechUsageCallback | undefined>;\n  /** Per-turn audio limit; unset means turns are unbounded. */\n  turnLimit?: TurnLimit;\n}\n\ntype QueueItem =\n  | {\n      kind: 'audio';\n      base64: string;\n      bytes: number;\n      resolve: () => void;\n      reject: (error: Error) => void;\n    }\n  | { kind: 'commit'; final: boolean };\n\ninterface FinalRecord {\n  itemId?: string;\n  text: string;\n  providerUsage?: Record<string, unknown>;\n  raw?: unknown;\n  arrival: number;\n}\n\nconst noop = () => undefined;\n\nexport class RealtimeTranscriptionSession implements StreamingSession {\n  state: StreamingSessionState = 'connecting';\n  sessionId?: string;\n  readonly ready: Promise<void>;\n\n  private readonly init: RealtimeSessionInit;\n  private readonly provider: string;\n  private resolveReady: () => void = noop;\n  private rejectReady: (error: Error) => void = noop;\n  private socket?: SpeechWebSocket;\n  private opened = false;\n  private secrets: string[] = [];\n  private queue: QueueItem[] = [];\n  private queued = 0;\n  private pumping = false;\n  private totalBytes = 0;\n  private uncommittedBytes = 0;\n  private outstandingCommits = 0;\n  /** Per outstanding commit, in order: whether it carried audio. */\n  private readonly commitHadAudio: boolean[] = [];\n  private turnOpen = false;\n  /** `Date.now()` of the last server message; bounds inactivity waits. */\n  private lastServerMessageAt = 0;\n  private readonly itemOrder: string[] = [];\n  private readonly pendingItems = new Set<string>();\n  private readonly vadStoppedItems = new Set<string>();\n  private readonly partials = new Map<string, string>();\n  private readonly finals: FinalRecord[] = [];\n  private readonly listeners = new Map<\n    StreamingSessionEventName,\n    Set<(event: never) => void>\n  >();\n  private error?: Error;\n  private endPromise?: Promise<TranscriptResult>;\n  private endWaiter?: { resolve: () => void; reject: (error: Error) => void };\n  private connectTimer?: ReturnType<typeof setTimeout>;\n  private endTimer?: ReturnType<typeof setTimeout>;\n  private closeEmitted = false;\n  /** Measures the current turn in write order when a turn limit applies. */\n  private readonly splitter?: TurnSplitter;\n  private readonly onAbort = () => {\n    this.fail(toError(this.init.signal?.reason, this.provider));\n  };\n\n  constructor(init: RealtimeSessionInit) {\n    this.init = init;\n    this.provider = init.protocol.provider;\n    assertTurnLimitFitsProtocol(init.protocol, init.turnLimit);\n    if (init.turnLimit) {\n      this.splitter = new TurnSplitter(init.turnLimit, init.config.format);\n    }\n    this.ready = new Promise<void>((resolve, reject) => {\n      this.resolveReady = resolve;\n      this.rejectReady = reject;\n    });\n    this.ready.catch(noop);\n\n    if (init.signal?.aborted) {\n      this.fail(toError(init.signal.reason, this.provider));\n      return;\n    }\n    init.signal?.addEventListener('abort', this.onAbort, { once: true });\n    void this.connect();\n  }\n\n  get queuedBytes(): number {\n    return this.queued;\n  }\n\n  write(chunk: StreamingAudioChunk): Promise<void> {\n    const promise = this.enqueueAudio(chunk);\n    promise.catch(noop);\n    return promise;\n  }\n\n  commit(): void {\n    if (this.isTerminal() || this.endPromise) {\n      return;\n    }\n    this.queue.push({ kind: 'commit', final: false });\n    this.splitter?.reset();\n    void this.pump();\n  }\n\n  end(): Promise<TranscriptResult> {\n    if (this.endPromise) {\n      return this.endPromise;\n    }\n\n    if (this.error) {\n      this.endPromise = Promise.reject(this.error);\n      return this.endPromise;\n    }\n\n    this.state = 'ending';\n    this.queue.push({ kind: 'commit', final: true });\n    const settled = new Promise<void>((resolve, reject) => {\n      this.endWaiter = { resolve, reject };\n    });\n    this.armEndTimer();\n    this.endPromise = settled.then(() => this.finish());\n    void this.pump();\n    this.checkEnded();\n    return this.endPromise;\n  }\n\n  /**\n   * (Re)starts the end timeout. It measures provider inactivity: every server\n   * message restarts it, so a long recording that is still being transcribed\n   * is not cut off, while a silent provider still fails after `timeoutMs`.\n   */\n  private armEndTimer(): void {\n    clearTimeout(this.endTimer);\n    this.endTimer = setTimeout(() => {\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} timed out after ${this.init.timeoutMs} ms waiting for the final transcript`,\n        ),\n      );\n    }, this.init.timeoutMs);\n  }\n\n  abort(reason?: unknown): void {\n    this.fail(toError(reason, this.provider));\n  }\n\n  on<E extends StreamingSessionEventName>(\n    event: E,\n    listener: StreamingSessionListener<E>,\n  ): () => void {\n    let set = this.listeners.get(event);\n    if (!set) {\n      set = new Set();\n      this.listeners.set(event, set);\n    }\n    set.add(listener as (event: never) => void);\n    return () => this.off(event, listener);\n  }\n\n  off<E extends StreamingSessionEventName>(\n    event: E,\n    listener: StreamingSessionListener<E>,\n  ): void {\n    this.listeners.get(event)?.delete(listener as (event: never) => void);\n  }\n\n  private enqueueAudio(chunk: StreamingAudioChunk): Promise<void> {\n    if (this.error) {\n      return Promise.reject(this.error);\n    }\n    if (this.endPromise || this.isTerminal()) {\n      return Promise.reject(\n        new SpeechError(\n          'Cannot write to a streaming session after end()',\n          'SPEECH_SESSION_ENDED',\n          this.provider,\n        ),\n      );\n    }\n\n    let bytes: Uint8Array;\n    try {\n      bytes = chunkToBytes(chunk, this.provider);\n    } catch (error) {\n      return Promise.reject(error);\n    }\n    if (bytes.byteLength === 0) {\n      return Promise.resolve();\n    }\n    if (this.queued + bytes.byteLength > this.init.maxBufferedBytes) {\n      return Promise.reject(\n        new SpeechError(\n          `Streaming write would queue more than maxBufferedBytes (${this.init.maxBufferedBytes}); await write() before sending more audio`,\n          'SPEECH_BACKPRESSURE',\n          this.provider,\n        ),\n      );\n    }\n\n    const splitter = this.splitter;\n    let cuts: number[] = [];\n    if (splitter && !this.init.turnLimit?.rollover) {\n      if (splitter.wouldOverflow(bytes.byteLength)) {\n        return Promise.reject(\n          new SpeechError(\n            `Streaming write would push the current turn past maxTurnSeconds (${this.init.turnLimit?.maxTurnSeconds} s); call commit() before writing more audio`,\n            'SPEECH_TURN_TOO_LONG',\n            this.provider,\n          ),\n        );\n      }\n      splitter.add(bytes.byteLength);\n    } else if (splitter) {\n      cuts = splitter.split(bytes);\n    }\n\n    return new Promise<void>((resolve, reject) => {\n      // A rollover splits the chunk: each part before a cut ends with an\n      // automatic commit, and the write settles with its last part.\n      const pushPart = (start: number, end: number) => {\n        const part = bytes.subarray(start, end);\n        this.queue.push({\n          kind: 'audio',\n          // Encode now so callers may reuse their buffer once write() returns.\n          base64: encodeBase64(part),\n          bytes: part.byteLength,\n          resolve: end === bytes.byteLength ? resolve : noop,\n          reject,\n        });\n        this.queued += part.byteLength;\n      };\n      // Cuts are ascending offsets in 1..byteLength, so no part is empty.\n      let start = 0;\n      for (const cut of cuts) {\n        pushPart(start, cut);\n        this.queue.push({ kind: 'commit', final: false });\n        start = cut;\n      }\n      if (start < bytes.byteLength) {\n        pushPart(start, bytes.byteLength);\n      }\n      void this.pump();\n    });\n  }\n\n  private async connect(): Promise<void> {\n    this.connectTimer = setTimeout(() => {\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} connection timed out after ${this.init.connectTimeoutMs} ms`,\n        ),\n      );\n    }, this.init.connectTimeoutMs);\n\n    try {\n      const connection = await this.init.connect();\n      if (this.isTerminal()) {\n        return;\n      }\n      this.secrets = connection.secrets;\n      const socket = this.init.createWebSocket(connection.url, {\n        protocols: connection.protocols,\n        headers: connection.headers,\n      });\n      this.socket = socket;\n      try {\n        socket.binaryType = 'arraybuffer';\n      } catch {\n        // Some implementations expose a read-only binaryType.\n      }\n      socket.addEventListener('open', () => this.onOpen());\n      socket.addEventListener('message', (event) => this.onMessage(event.data));\n      socket.addEventListener('error', () => {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} WebSocket error`,\n          ),\n        );\n      });\n      socket.addEventListener('close', (event) => this.onClose(event));\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n    }\n  }\n\n  private onOpen(): void {\n    clearTimeout(this.connectTimer);\n    if (this.isTerminal()) {\n      this.closeSocket();\n      return;\n    }\n\n    this.opened = true;\n    for (const message of this.init.protocol.sessionMessages(\n      this.init.config,\n    )) {\n      if (!this.send(message)) {\n        return;\n      }\n    }\n    if (this.state === 'connecting') {\n      this.state = 'open';\n    }\n    this.resolveReady();\n    this.emit('open', { sessionId: this.sessionId });\n    void this.pump();\n  }\n\n  private onMessage(data: unknown): void {\n    if (this.isTerminal()) {\n      return;\n    }\n\n    this.lastServerMessageAt = Date.now();\n    if (this.endWaiter) {\n      this.armEndTimer();\n    }\n\n    const text = messageDataToText(data);\n    if (text === undefined) {\n      return;\n    }\n\n    let parsed: unknown;\n    try {\n      parsed = JSON.parse(text);\n    } catch {\n      return;\n    }\n\n    let events: RealtimeProtocolEvent[];\n    try {\n      events = this.init.protocol.parse(parsed);\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n      return;\n    }\n\n    for (const event of events) {\n      if (this.isTerminal()) {\n        return;\n      }\n      this.handle(event);\n    }\n    this.checkEnded();\n  }\n\n  private handle(event: RealtimeProtocolEvent): void {\n    switch (event.type) {\n      case 'session':\n        this.sessionId = event.sessionId ?? this.sessionId;\n        return;\n      case 'committed':\n        if (event.itemId && !this.pendingItems.has(event.itemId)) {\n          this.pendingItems.add(event.itemId);\n          this.itemOrder.push(event.itemId);\n        }\n        // A turn whose speech_stopped we saw was committed by server VAD, so\n        // it does not acknowledge one of our commits.\n        if (\n          this.init.protocol.commitAck === 'committed' &&\n          !(event.itemId && this.vadStoppedItems.has(event.itemId))\n        ) {\n          this.ackCommit();\n        }\n        return;\n      case 'commit_empty':\n        this.ackCommit();\n        return;\n      case 'partial': {\n        const key = event.itemId ?? '';\n        const text = (this.partials.get(key) ?? '') + event.delta;\n        this.partials.set(key, text);\n        this.emit('partial', {\n          delta: event.delta,\n          text,\n          itemId: event.itemId,\n        });\n        return;\n      }\n      case 'final': {\n        let serverEnded = false;\n        if (this.init.protocol.commitAck === 'final') {\n          if (this.outstandingCommits === 0) {\n            // A final we did not ask for: the server ended the open turn by\n            // itself (e.g. a server-side generation limit).\n            serverEnded = true;\n          } else if (this.ackCommit() && event.audioConsumed === false) {\n            // Our turn carried audio but the server consumed none. vLLM does\n            // this when a stale end-of-turn marker (left by a commit that\n            // crossed a server-ended turn) ends the turn at once.\n            this.fail(\n              new SpeechProviderError(\n                this.provider,\n                `${this.provider} finished a turn without transcribing its audio (the server's turn state is out of sync, e.g. after a commit crossed a server-ended turn); start a new session`,\n              ),\n            );\n            return;\n          }\n        }\n        if (event.itemId) {\n          this.pendingItems.delete(event.itemId);\n        }\n        this.partials.delete(event.itemId ?? '');\n        this.finals.push({\n          itemId: event.itemId,\n          text: event.text,\n          providerUsage: event.providerUsage,\n          raw: event.raw,\n          arrival: this.finals.length,\n        });\n        this.emit('final', {\n          text: event.text,\n          itemId: event.itemId,\n          providerUsage: event.providerUsage,\n          raw: event.raw,\n        });\n        if (serverEnded) {\n          // Fail closed: audio sent after the cut-off was never transcribed,\n          // and the server may now be out of step with our turns.\n          this.fail(\n            new SpeechProviderError(\n              this.provider,\n              `${this.provider} ended a turn before it was committed (for example at a server generation limit); audio sent after that point was not transcribed. Call commit() more often and start a new session`,\n            ),\n          );\n        }\n        return;\n      }\n      case 'speech_started':\n        this.emit(event.type, { itemId: event.itemId, audioMs: event.audioMs });\n        return;\n      case 'speech_stopped':\n        if (event.itemId) {\n          this.vadStoppedItems.add(event.itemId);\n        }\n        this.emit(event.type, { itemId: event.itemId, audioMs: event.audioMs });\n        return;\n      case 'error':\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            this.redact(\n              `${this.provider} error${event.code ? ` (${event.code})` : ''}: ${event.message}`,\n            ) ?? `${this.provider} error`,\n            {\n              responseBody:\n                event.raw === undefined\n                  ? undefined\n                  : this.redact(JSON.stringify(event.raw)),\n            },\n          ),\n        );\n        return;\n    }\n  }\n\n  /** Acknowledges the oldest outstanding commit; returns whether it carried audio. */\n  private ackCommit(): boolean {\n    this.outstandingCommits = Math.max(0, this.outstandingCommits - 1);\n    return this.commitHadAudio.shift() ?? false;\n  }\n\n  private async pump(): Promise<void> {\n    if (this.pumping || !this.opened) {\n      return;\n    }\n\n    this.pumping = true;\n    try {\n      while (this.queue.length > 0 && !this.isTerminal()) {\n        await this.waitForDrain();\n        if (!(await this.openTurnIfNeeded())) {\n          break;\n        }\n        const item = this.queue.shift();\n        if (!item || this.isTerminal()) {\n          if (item?.kind === 'audio') {\n            item.reject(this.error ?? new Error('Session closed'));\n          }\n          break;\n        }\n\n        if (item.kind === 'commit') {\n          this.sendCommit(item.final);\n          continue;\n        }\n\n        this.queued -= item.bytes;\n        if (!this.send(this.init.protocol.appendMessage(item.base64))) {\n          item.reject(this.error ?? new Error('Session closed'));\n          break;\n        }\n        this.totalBytes += item.bytes;\n        this.uncommittedBytes += item.bytes;\n        await this.waitForDrain();\n        if (this.error) {\n          item.reject(this.error);\n        } else {\n          item.resolve();\n        }\n      }\n    } finally {\n      this.pumping = false;\n    }\n    this.checkEnded();\n  }\n\n  private sendCommit(final: boolean): void {\n    const needed =\n      this.uncommittedBytes > 0 ||\n      (final && this.init.protocol.endCommit === 'always');\n    if (!needed || !this.padShortTurn()) {\n      return;\n    }\n    if (this.send(this.init.protocol.commitMessage({ final }))) {\n      this.commitHadAudio.push(this.uncommittedBytes > 0);\n      this.uncommittedBytes = 0;\n      this.outstandingCommits += 1;\n      this.turnOpen = false;\n    }\n  }\n\n  /**\n   * In a session with a turn limit, tops up a non-empty turn shorter than the\n   * protocol's `minCommitSeconds` with silence (not counted as streamed\n   * audio). Rollover can leave such a tail, which the provider would\n   * otherwise reject as empty and drop. Returns false when the send failed.\n   */\n  private padShortTurn(): boolean {\n    const minSeconds = this.init.protocol.minCommitSeconds;\n    if (!this.init.turnLimit || !minSeconds || this.uncommittedBytes === 0) {\n      return true;\n    }\n    const format = this.init.config.format;\n    const align = blockAlign(format);\n    // Round up to whole sample frames so the turn reaches the minimum.\n    const minBytes =\n      Math.ceil((minSeconds * bytesPerSecond(format)) / align - 1e-9) * align;\n    const missing = minBytes - this.uncommittedBytes;\n    if (missing <= 0) {\n      return true;\n    }\n    if (\n      !this.send(\n        this.init.protocol.appendMessage(\n          encodeBase64(silence(missing, format)),\n        ),\n      )\n    ) {\n      return false;\n    }\n    this.uncommittedBytes += missing;\n    return true;\n  }\n\n  /**\n   * For protocols with an explicit turn start: before the next audio, waits\n   * for every outstanding turn to be acknowledged, then opens a new turn. Like\n   * the end timeout, the wait is bounded by provider inactivity: it fails only\n   * after `timeoutMs` without any server message. Returns false when the\n   * session has failed.\n   */\n  private async openTurnIfNeeded(): Promise<boolean> {\n    const start = this.init.protocol.turnStartMessage;\n    if (!start || this.turnOpen || this.queue[0]?.kind !== 'audio') {\n      return true;\n    }\n\n    const waitStartedAt = Date.now();\n    while (!this.isTerminal() && this.outstandingCommits > 0) {\n      const lastActivity = Math.max(waitStartedAt, this.lastServerMessageAt);\n      if (Date.now() - lastActivity >= this.init.timeoutMs) {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} timed out after ${this.init.timeoutMs} ms without a server message while waiting for the previous turn to finish`,\n          ),\n        );\n        break;\n      }\n      await new Promise((resolve) => setTimeout(resolve, DRAIN_POLL_MS));\n    }\n    if (this.isTerminal() || !this.send(start.call(this.init.protocol))) {\n      return false;\n    }\n    this.turnOpen = true;\n    return true;\n  }\n\n  /**\n   * Waits until the socket's send buffer is at or below `highWaterMark`. A\n   * socket that stays congested for `timeoutMs` fails the session, so a\n   * stalled connection can never leave `write()` pending forever.\n   */\n  private async waitForDrain(): Promise<void> {\n    const deadline = Date.now() + this.init.timeoutMs;\n    while (\n      !this.isTerminal() &&\n      this.socket &&\n      this.socket.bufferedAmount > this.init.highWaterMark\n    ) {\n      if (Date.now() >= deadline) {\n        this.fail(\n          new SpeechProviderError(\n            this.provider,\n            `${this.provider} socket did not drain below highWaterMark within ${this.init.timeoutMs} ms`,\n          ),\n        );\n        return;\n      }\n      await new Promise((resolve) => setTimeout(resolve, DRAIN_POLL_MS));\n    }\n  }\n\n  private send(message: unknown): boolean {\n    try {\n      this.socket?.send(JSON.stringify(message));\n      return true;\n    } catch (error) {\n      this.fail(toError(error, this.provider));\n      return false;\n    }\n  }\n\n  private checkEnded(): void {\n    if (\n      !this.endWaiter ||\n      this.isTerminal() ||\n      !this.opened ||\n      this.pumping ||\n      this.queue.length > 0 ||\n      this.outstandingCommits > 0 ||\n      this.pendingItems.size > 0\n    ) {\n      return;\n    }\n\n    clearTimeout(this.endTimer);\n    const waiter = this.endWaiter;\n    this.endWaiter = undefined;\n    waiter.resolve();\n  }\n\n  private async finish(): Promise<TranscriptResult> {\n    this.state = 'closed';\n    this.cleanup();\n    this.closeSocket();\n\n    const ordered = [...this.finals].sort(\n      (left, right) => this.rank(left) - this.rank(right),\n    );\n    const segments: TranscriptSegment[] = ordered\n      .map((final) => final.text.trim())\n      .filter(Boolean)\n      .map((text) => ({ text }));\n    const providerUsage = sumUsage(ordered.map((final) => final.providerUsage));\n    const durationSeconds = audioSecondsForBytes(\n      this.totalBytes,\n      this.init.config.format,\n    );\n    const usage: SpeechUsage = {\n      operation: 'transcription',\n      provider: this.provider,\n      model: this.init.config.model,\n      audioSeconds:\n        audioSecondsFromProviderUsage(providerUsage) ?? durationSeconds,\n      bytes: this.totalBytes,\n      providerUsage,\n    };\n    const result: TranscriptResult = {\n      text: segments.map((segment) => segment.text).join(' '),\n      language: this.init.config.language,\n      durationSeconds,\n      segments,\n      provider: this.provider,\n      model: this.init.config.model,\n      raw: {\n        sessionId: this.sessionId,\n        finals: ordered.map((final) => final.raw),\n      },\n      usage,\n    };\n\n    await reportSpeechUsage(usage, ...this.init.onUsage);\n    return result;\n  }\n\n  private rank(final: FinalRecord): number {\n    const index = final.itemId ? this.itemOrder.indexOf(final.itemId) : -1;\n    return index >= 0 ? index : this.itemOrder.length + final.arrival;\n  }\n\n  private onClose(event: SpeechWebSocketCloseEvent): void {\n    clearTimeout(this.connectTimer);\n    if (!this.isTerminal()) {\n      const code = event.code ?? 1005;\n      this.fail(\n        new SpeechProviderError(\n          this.provider,\n          `${this.provider} socket closed unexpectedly (code ${code}${event.reason ? `: ${this.redact(event.reason)}` : ''})`,\n        ),\n      );\n    }\n    if (!this.closeEmitted) {\n      this.closeEmitted = true;\n      this.emit('close', {\n        code: event.code ?? 1005,\n        reason: this.redact(event.reason ?? '') ?? '',\n      });\n    }\n  }\n\n  private fail(error: Error): void {\n    if (this.isTerminal()) {\n      return;\n    }\n\n    this.error = error;\n    this.state = 'failed';\n    this.cleanup();\n    const queue = this.queue;\n    this.queue = [];\n    this.queued = 0;\n    for (const item of queue) {\n      if (item.kind === 'audio') {\n        item.reject(error);\n      }\n    }\n    this.rejectReady(error);\n    const waiter = this.endWaiter;\n    this.endWaiter = undefined;\n    waiter?.reject(error);\n    this.emit('error', error);\n    this.closeSocket();\n  }\n\n  private cleanup(): void {\n    clearTimeout(this.connectTimer);\n    clearTimeout(this.endTimer);\n    this.init.signal?.removeEventListener('abort', this.onAbort);\n  }\n\n  private closeSocket(): void {\n    try {\n      this.socket?.close(1000);\n    } catch {\n      // Already closing.\n    }\n  }\n\n  private emit<E extends StreamingSessionEventName>(\n    event: E,\n    payload: StreamingSessionEvents[E],\n  ): void {\n    for (const listener of [...(this.listeners.get(event) ?? [])]) {\n      try {\n        (listener as StreamingSessionListener<E>)(payload);\n      } catch (error) {\n        // A throwing listener fails the session so the bug surfaces via end().\n        if (event !== 'error' && event !== 'close') {\n          this.fail(toError(error, this.provider));\n        }\n      }\n    }\n  }\n\n  private isTerminal(): boolean {\n    return this.state === 'closed' || this.state === 'failed';\n  }\n\n  private redact(text: string | undefined): string | undefined {\n    let result = text;\n    for (const secret of this.secrets) {\n      result = redactSecret(result, secret);\n    }\n    return result;\n  }\n}\n\n/**\n * Throws when a turn cap is below the provider's minimum commit length:\n * padding a short turn up to that minimum would then exceed the cap.\n */\nexport function assertTurnLimitFitsProtocol(\n  protocol: Pick<RealtimeProtocol, 'provider' | 'minCommitSeconds'>,\n  limit: Pick<TurnLimit, 'maxTurnSeconds'> | undefined,\n): void {\n  const minimum = protocol.minCommitSeconds;\n  if (limit && minimum && limit.maxTurnSeconds < minimum) {\n    throw new SpeechConfigurationError(\n      `maxTurnSeconds (${limit.maxTurnSeconds}) is below ${protocol.provider}'s minimum commit of ${minimum} s`,\n      protocol.provider,\n    );\n  }\n}\n\nfunction toError(value: unknown, provider: string): Error {\n  if (value instanceof Error) {\n    return value;\n  }\n  return new SpeechError(\n    value === undefined ? 'Streaming session aborted' : String(value),\n    'SPEECH_ABORTED',\n    provider,\n  );\n}\n\n/**\n * Sums numeric fields of per-turn provider usage blocks (recursively), keeping\n * the first value of non-numeric fields such as `type`.\n */\nexport function sumUsage(\n  blocks: Array<Record<string, unknown> | undefined>,\n): Record<string, unknown> | undefined {\n  const present = blocks.filter(\n    (block): block is Record<string, unknown> =>\n      Boolean(block) && typeof block === 'object',\n  );\n  if (present.length === 0) {\n    return undefined;\n  }\n\n  const total: Record<string, unknown> = {};\n  for (const block of present) {\n    for (const [key, value] of Object.entries(block)) {\n      const current = total[key];\n      if (typeof value === 'number') {\n        total[key] = (typeof current === 'number' ? current : 0) + value;\n      } else if (value && typeof value === 'object' && !Array.isArray(value)) {\n        total[key] = sumUsage([\n          current as Record<string, unknown> | undefined,\n          value as Record<string, unknown>,\n        ]);\n      } else if (current === undefined) {\n        total[key] = value;\n      }\n    }\n  }\n  return total;\n}\n","/**\n * OpenAI Realtime transcription adapter (`openai-realtime`).\n *\n * Targets the GA Realtime API transcription session (no `OpenAI-Beta`\n * header): connect to `wss://api.openai.com/v1/realtime?intent=transcription`,\n * send `session.update` with `session.type: \"transcription\"`, stream\n * `input_audio_buffer.append`, and read\n * `conversation.item.input_audio_transcription.delta` / `.completed`.\n * Reference: https://developers.openai.com/api/docs/guides/realtime-transcription\n * and https://developers.openai.com/api/reference/resources/realtime/client-events\n */\n\nimport { SpeechConfigurationError } from '../shared/errors.js';\nimport { compactJson, resolveOpenAICompatibleUrl } from '../shared/http.js';\nimport {\n  assertNoBrowserApiKey,\n  CLIENT_SECRET_SUBPROTOCOL_PREFIX,\n  mergeHeaderRecords,\n  realtimeConnector,\n} from '../shared/realtime-auth.js';\nimport {\n  assertTurnLimitFitsProtocol,\n  DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n  DEFAULT_STREAMING_HIGH_WATER_MARK,\n  DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n  DEFAULT_STREAMING_TIMEOUT_MS,\n  type RealtimeProtocol,\n  type RealtimeProtocolEvent,\n  type RealtimeSessionConfig,\n  RealtimeTranscriptionSession,\n} from '../shared/realtime-session.js';\nimport type {\n  OpenAIRealtimeTranscriberOptions,\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTurnDetection,\n  StreamingTurnLimit,\n} from '../shared/streaming-types.js';\nimport {\n  describeTurnLimit,\n  resolveTurnLimit,\n  type TurnLimit,\n} from '../shared/turn-limit.js';\nimport {\n  resolveWebSocketFactory,\n  type SpeechWebSocketFactory,\n  toWebSocketUrl,\n} from '../shared/websocket.js';\n\nconst ADAPTER = 'openai-realtime';\nexport const OPENAI_REALTIME_DEFAULT_URL = 'wss://api.openai.com/v1/realtime';\nexport const OPENAI_REALTIME_DEFAULT_MODEL = 'gpt-4o-transcribe';\n\n/** GA realtime input formats: 24 kHz mono PCM16, or 8 kHz mono G.711. */\nconst WIRE_FORMATS: Record<\n  StreamingAudioFormat['encoding'],\n  { type: string; sampleRate: number }\n> = {\n  pcm16: { type: 'audio/pcm', sampleRate: 24_000 },\n  g711_ulaw: { type: 'audio/pcmu', sampleRate: 8_000 },\n  g711_alaw: { type: 'audio/pcma', sampleRate: 8_000 },\n};\n\n/** The OpenAI Realtime transcription wire protocol. */\nexport const openAIRealtimeProtocol = (\n  extras: Pick<\n    OpenAIRealtimeTranscriberOptions,\n    'noiseReduction' | 'include' | 'transcriptionOptions'\n  > = {},\n): RealtimeProtocol => ({\n  provider: ADAPTER,\n  commitAck: 'committed',\n  endCommit: 'if-audio',\n  // No per-turn cap by default: server VAD ends turns. Manual sessions may opt\n  // in with `maxTurnSeconds`.\n  // OpenAI rejects a commit with under 100 ms of audio as empty.\n  minCommitSeconds: 0.1,\n  sessionMessages(config: RealtimeSessionConfig) {\n    return [\n      {\n        type: 'session.update',\n        session: openAIRealtimeTranscriptionSession(config, extras),\n      },\n    ];\n  },\n  appendMessage(base64Audio: string) {\n    return { type: 'input_audio_buffer.append', audio: base64Audio };\n  },\n  commitMessage() {\n    return { type: 'input_audio_buffer.commit' };\n  },\n  parse: parseOpenAIRealtimeEvent,\n});\n\n/**\n * The GA transcription session object (`session.type: \"transcription\"`),\n * shared by `session.update` and the client-secret mint request.\n */\nexport function openAIRealtimeTranscriptionSession(\n  config: RealtimeSessionConfig,\n  extras: Pick<\n    OpenAIRealtimeTranscriberOptions,\n    'noiseReduction' | 'include' | 'transcriptionOptions'\n  > = {},\n): Record<string, unknown> {\n  const input: Record<string, unknown> = {\n    format:\n      config.format.encoding === 'pcm16'\n        ? { type: 'audio/pcm', rate: config.format.sampleRate }\n        : { type: WIRE_FORMATS[config.format.encoding].type },\n    // Extension fields first: the typed settings must win, so the model\n    // sent (or bound to a minted client secret) always matches the model\n    // reported in usage.\n    transcription: {\n      ...compactJson(extras.transcriptionOptions ?? {}),\n      ...compactJson({\n        model: config.model,\n        language: config.language,\n        prompt: config.prompt,\n      }),\n    },\n    turn_detection: turnDetectionToWire(config.turnDetection),\n  };\n  if (extras.noiseReduction) {\n    input.noise_reduction = { type: extras.noiseReduction };\n  }\n\n  return compactJson({\n    type: 'transcription',\n    audio: { input },\n    include: extras.include,\n  });\n}\n\n/** Maps one OpenAI Realtime server event to normalised session events. */\nexport function parseOpenAIRealtimeEvent(\n  message: unknown,\n): RealtimeProtocolEvent[] {\n  if (!message || typeof message !== 'object') {\n    return [];\n  }\n\n  const event = message as Record<string, unknown>;\n  const itemId = stringField(event, 'item_id');\n\n  switch (event.type) {\n    case 'session.created':\n    case 'session.updated':\n    case 'transcription_session.created':\n    case 'transcription_session.updated': {\n      const session = event.session as Record<string, unknown> | undefined;\n      return [\n        { type: 'session', sessionId: session && stringField(session, 'id') },\n      ];\n    }\n    case 'input_audio_buffer.committed':\n      return [{ type: 'committed', itemId }];\n    case 'input_audio_buffer.speech_started':\n      return [\n        {\n          type: 'speech_started',\n          itemId,\n          audioMs: numberField(event, 'audio_start_ms'),\n        },\n      ];\n    case 'input_audio_buffer.speech_stopped':\n      return [\n        {\n          type: 'speech_stopped',\n          itemId,\n          audioMs: numberField(event, 'audio_end_ms'),\n        },\n      ];\n    case 'conversation.item.input_audio_transcription.delta': {\n      const delta = stringField(event, 'delta');\n      return delta ? [{ type: 'partial', itemId, delta }] : [];\n    }\n    case 'conversation.item.input_audio_transcription.completed': {\n      const usage = event.usage;\n      return [\n        {\n          type: 'final',\n          itemId,\n          text: stringField(event, 'transcript') ?? '',\n          providerUsage:\n            usage && typeof usage === 'object' && !Array.isArray(usage)\n              ? (usage as Record<string, unknown>)\n              : undefined,\n          raw: event,\n        },\n      ];\n    }\n    case 'conversation.item.input_audio_transcription.failed': {\n      const error = (event.error ?? {}) as Record<string, unknown>;\n      return [\n        {\n          type: 'error',\n          message: stringField(error, 'message') ?? 'transcription failed',\n          code: stringField(error, 'code'),\n          raw: event,\n        },\n      ];\n    }\n    case 'error': {\n      const error = (event.error ?? {}) as Record<string, unknown>;\n      const code = stringField(error, 'code');\n      if (code === 'input_audio_buffer_commit_empty') {\n        return [{ type: 'commit_empty' }];\n      }\n      return [\n        {\n          type: 'error',\n          message: stringField(error, 'message') ?? 'unknown error',\n          code,\n          raw: event,\n        },\n      ];\n    }\n    default:\n      return [];\n  }\n}\n\n/**\n * Realtime transcriber for OpenAI and OpenAI-Realtime-compatible servers.\n *\n * Auth: `apiKey` is sent as `Authorization: Bearer` (Node only). In browsers,\n * pass `clientSecret` (an ephemeral client secret minted by your server); it\n * is sent as the `openai-insecure-api-key.<secret>` subprotocol. Supplying\n * `apiKey` or `headers` in a browser throws.\n */\nexport class OpenAIRealtimeTranscriber implements StreamingTranscriber {\n  readonly type = ADAPTER;\n  readonly audioFormat: StreamingAudioFormat;\n\n  private readonly options: OpenAIRealtimeTranscriberOptions;\n  private readonly url: string;\n  private readonly createWebSocket: SpeechWebSocketFactory;\n  private readonly protocol: RealtimeProtocol;\n\n  constructor(options: OpenAIRealtimeTranscriberOptions = {}) {\n    assertNoBrowserApiKey(ADAPTER, options.apiKey);\n\n    this.options = options;\n    this.url = resolveRealtimeUrl(\n      options.baseUrl ?? OPENAI_REALTIME_DEFAULT_URL,\n    );\n    this.createWebSocket = resolveWebSocketFactory(ADAPTER, options);\n    this.audioFormat = resolveOpenAIRealtimeFormat(options.format);\n    this.protocol = openAIRealtimeProtocol(options);\n    resolveTurnLimit(ADAPTER, undefined, options);\n  }\n\n  /**\n   * A turn limit applies only to `manual` sessions with `maxTurnSeconds` set;\n   * under server or semantic VAD the provider ends turns itself.\n   */\n  turnLimit(\n    options: StreamingSessionOptions = {},\n  ): StreamingTurnLimit | undefined {\n    return describeTurnLimit(\n      this.resolveTurnLimit(options),\n      this.sessionFormat(options),\n    );\n  }\n\n  private sessionFormat(\n    options: StreamingSessionOptions,\n  ): StreamingAudioFormat {\n    // A different encoding starts from that encoding's defaults.\n    return resolveOpenAIRealtimeFormat(\n      options.format?.encoding &&\n        options.format.encoding !== this.audioFormat.encoding\n        ? options.format\n        : { ...this.audioFormat, ...options.format },\n    );\n  }\n\n  private resolveTurnLimit(\n    options: StreamingSessionOptions,\n  ): TurnLimit | undefined {\n    const limit = resolveTurnLimit(\n      ADAPTER,\n      this.protocol.maxTurnSeconds,\n      options,\n      this.options,\n    );\n    if (this.turnDetection(options).type !== 'manual') {\n      return undefined;\n    }\n    assertTurnLimitFitsProtocol(this.protocol, limit);\n    return limit;\n  }\n\n  private turnDetection(\n    options: StreamingSessionOptions,\n  ): StreamingTurnDetection {\n    return (\n      options.turnDetection ??\n      this.options.turnDetection ?? { type: 'server_vad' }\n    );\n  }\n\n  start(options: StreamingSessionOptions = {}): StreamingSession {\n    const format = this.sessionFormat(options);\n    const connect = realtimeConnector({\n      adapter: ADAPTER,\n      url: this.url,\n      apiKey: this.options.apiKey,\n      headers: mergeHeaderRecords(this.options.headers, options.headers),\n      clientSecret: options.clientSecret ?? this.options.clientSecret,\n      protocols: (secret) =>\n        secret\n          ? ['realtime', `${CLIENT_SECRET_SUBPROTOCOL_PREFIX}${secret}`]\n          : ['realtime'],\n    });\n\n    return new RealtimeTranscriptionSession({\n      protocol: this.protocol,\n      config: {\n        model:\n          options.model?.trim() ||\n          this.options.model?.trim() ||\n          OPENAI_REALTIME_DEFAULT_MODEL,\n        format,\n        turnDetection: this.turnDetection(options),\n        language: options.language ?? this.options.language,\n        prompt: options.prompt ?? this.options.prompt,\n      },\n      connect,\n      createWebSocket: this.createWebSocket,\n      timeoutMs: this.options.timeoutMs ?? DEFAULT_STREAMING_TIMEOUT_MS,\n      connectTimeoutMs:\n        this.options.connectTimeoutMs ?? DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n      highWaterMark:\n        this.options.highWaterMark ?? DEFAULT_STREAMING_HIGH_WATER_MARK,\n      maxBufferedBytes:\n        this.options.maxBufferedBytes ?? DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n      signal: options.signal,\n      onUsage: [this.options.onUsage, options.onUsage],\n      turnLimit: this.resolveTurnLimit(options),\n    });\n  }\n}\n\n/**\n * Resolves a realtime WebSocket URL: `http(s)` becomes `ws(s)`, a server root\n * gets `/v1/realtime`, a versioned API root gets `/realtime`, and\n * `intent=transcription` is added unless an `intent` is already present. No\n * `model` query parameter is added: the model travels in `session.update`.\n */\nexport function resolveRealtimeUrl(baseUrl: string): string {\n  const url = new URL(\n    toWebSocketUrl(resolveOpenAICompatibleUrl(baseUrl, 'realtime'), ADAPTER),\n  );\n  if (!url.searchParams.has('intent')) {\n    url.searchParams.set('intent', 'transcription');\n  }\n  return url.toString();\n}\n\nexport function resolveOpenAIRealtimeFormat(\n  format: Partial<StreamingAudioFormat> | undefined,\n): StreamingAudioFormat {\n  const encoding = format?.encoding ?? 'pcm16';\n  const wire = WIRE_FORMATS[encoding];\n  if (!wire) {\n    throw new SpeechConfigurationError(\n      `openai-realtime does not support encoding ${String(encoding)}`,\n      ADAPTER,\n    );\n  }\n\n  const resolved: StreamingAudioFormat = {\n    encoding,\n    sampleRate: format?.sampleRate ?? wire.sampleRate,\n    channels: format?.channels ?? 1,\n  };\n  if (resolved.sampleRate !== wire.sampleRate || resolved.channels !== 1) {\n    throw new SpeechConfigurationError(\n      `openai-realtime requires ${encoding} at ${wire.sampleRate} Hz mono (got ${resolved.sampleRate} Hz, ${resolved.channels} channel(s)); resample before streaming`,\n      ADAPTER,\n    );\n  }\n  return resolved;\n}\n\nfunction turnDetectionToWire(\n  turnDetection: StreamingTurnDetection,\n): Record<string, unknown> | null {\n  switch (turnDetection.type) {\n    case 'manual':\n      return null;\n    case 'server_vad':\n      return compactJson({\n        type: 'server_vad',\n        threshold: turnDetection.threshold,\n        prefix_padding_ms: turnDetection.prefixPaddingMs,\n        silence_duration_ms: turnDetection.silenceDurationMs,\n      });\n    case 'semantic_vad':\n      return compactJson({\n        type: 'semantic_vad',\n        eagerness: turnDetection.eagerness,\n      });\n    default:\n      throw new SpeechConfigurationError(\n        `Unknown turn detection type: ${String((turnDetection as { type?: unknown }).type)}`,\n        ADAPTER,\n      );\n  }\n}\n\nfunction stringField(\n  record: Record<string, unknown>,\n  key: string,\n): string | undefined {\n  const value = record[key];\n  return typeof value === 'string' ? value : undefined;\n}\n\nfunction numberField(\n  record: Record<string, unknown>,\n  key: string,\n): number | undefined {\n  const value = record[key];\n  return typeof value === 'number' ? value : undefined;\n}\n","/**\n * vLLM realtime transcription adapter (`voxtral-realtime`), for Mistral's\n * Voxtral Realtime models (e.g. `mistralai/Voxtral-Mini-4B-Realtime-2602`)\n * served by vLLM at `/v1/realtime`.\n *\n * vLLM's protocol borrows OpenAI event names but is not OpenAI-compatible.\n * Verified against vLLM's `entrypoints/speech_to_text/realtime` source and a\n * live server:\n *\n * 1. The server sends `session.created` (`{ id, created }`) on connect.\n * 2. The client sends `session.update` with a top-level `model` (no nested\n *    `session` object, no acknowledgement; an unknown model yields an\n *    `error` with code `model_not_found`).\n * 3. A non-final `input_audio_buffer.commit` starts generation for a turn.\n * 4. `input_audio_buffer.append` carries base64 PCM16 at 16 kHz mono. Audio\n *    appended before the start commit is kept.\n * 5. The server streams `transcription.delta` (`{ delta }`, often empty) while\n *    audio arrives, at roughly real-time speed.\n * 6. `input_audio_buffer.commit` with `final: true` ends the turn; the server\n *    replies `transcription.done` (`{ text, usage }`, token usage only) and\n *    clears its buffer. A further start commit opens the next turn on the same\n *    socket.\n * 7. Errors arrive as `{ type: 'error', error: string, code }` and leave the\n *    socket open; this adapter treats every error as fatal.\n * 8. A turn whose audio and tokens exceed `max_model_len` (~12.5 tokens per\n *    audio second plus a 39-token prompt) ends in `error` `processing_error`\n *    and can take the engine down. The limit is per turn: after a final\n *    commit vLLM clears the turn's context. The session therefore rolls over\n *    to a new turn before `maxTurnSeconds` (default 270) of audio.\n * 9. A stray final commit leaves a stale end marker: the next turn ends at once\n *    with `prompt_tokens: 1` (any audio costs 39) and its audio is dropped.\n *    The session fails closed on that, and on any `done` for a turn it did\n *    not commit, rather than losing audio silently.\n */\n\nimport { SpeechConfigurationError } from '../shared/errors.js';\nimport { resolveOpenAICompatibleUrl } from '../shared/http.js';\nimport {\n  assertNoBrowserApiKey,\n  CLIENT_SECRET_SUBPROTOCOL_PREFIX,\n  mergeHeaderRecords,\n  realtimeConnector,\n} from '../shared/realtime-auth.js';\nimport {\n  DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n  DEFAULT_STREAMING_HIGH_WATER_MARK,\n  DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n  DEFAULT_STREAMING_TIMEOUT_MS,\n  type RealtimeProtocol,\n  type RealtimeProtocolEvent,\n  RealtimeTranscriptionSession,\n} from '../shared/realtime-session.js';\nimport type {\n  StreamingAudioFormat,\n  StreamingSession,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTurnDetection,\n  StreamingTurnLimit,\n  VoxtralRealtimeTranscriberOptions,\n} from '../shared/streaming-types.js';\nimport {\n  DEFAULT_MAX_TURN_SECONDS,\n  describeTurnLimit,\n  resolveTurnLimit,\n  type TurnLimit,\n} from '../shared/turn-limit.js';\nimport {\n  resolveWebSocketFactory,\n  type SpeechWebSocketFactory,\n  toWebSocketUrl,\n} from '../shared/websocket.js';\n\nconst ADAPTER = 'voxtral-realtime';\n\n/** vLLM's default served model name for Voxtral Mini Realtime. */\nexport const VOXTRAL_REALTIME_DEFAULT_MODEL =\n  'mistralai/Voxtral-Mini-4B-Realtime-2602';\n\n/** The only input format vLLM's realtime endpoint accepts. */\nexport const VOXTRAL_REALTIME_AUDIO_FORMAT: Readonly<StreamingAudioFormat> =\n  Object.freeze({ encoding: 'pcm16', sampleRate: 16_000, channels: 1 });\n\n/** The vLLM realtime wire protocol. */\nexport const voxtralRealtimeProtocol = (): RealtimeProtocol => ({\n  provider: ADAPTER,\n  // vLLM never acknowledges a commit; `transcription.done` closes the turn.\n  commitAck: 'final',\n  // The start commit is sent lazily before a turn's first audio, so a session\n  // with no uncommitted audio has no open turn and needs no end commit.\n  endCommit: 'if-audio',\n  // vLLM's context (`max_model_len`) covers one turn; roll over before it.\n  maxTurnSeconds: DEFAULT_MAX_TURN_SECONDS,\n  sessionMessages(config) {\n    return [{ type: 'session.update', model: config.model }];\n  },\n  turnStartMessage() {\n    return { type: 'input_audio_buffer.commit' };\n  },\n  appendMessage(base64Audio) {\n    return { type: 'input_audio_buffer.append', audio: base64Audio };\n  },\n  commitMessage() {\n    // Every turn end is a final commit; the next turn gets a new start commit.\n    return { type: 'input_audio_buffer.commit', final: true };\n  },\n  parse: parseVoxtralRealtimeEvent,\n});\n\n/** Maps one vLLM realtime server event to normalised session events. */\nexport function parseVoxtralRealtimeEvent(\n  message: unknown,\n): RealtimeProtocolEvent[] {\n  if (!message || typeof message !== 'object') {\n    return [];\n  }\n\n  const event = message as Record<string, unknown>;\n  switch (event.type) {\n    case 'session.created':\n      return [\n        {\n          type: 'session',\n          sessionId: typeof event.id === 'string' ? event.id : undefined,\n        },\n      ];\n    case 'transcription.delta': {\n      const delta = typeof event.delta === 'string' ? event.delta : '';\n      return delta ? [{ type: 'partial', delta }] : [];\n    }\n    case 'transcription.done': {\n      const usage =\n        event.usage &&\n        typeof event.usage === 'object' &&\n        !Array.isArray(event.usage)\n          ? (event.usage as Record<string, unknown>)\n          : undefined;\n      // Any audio, even one sample, costs the turn's full audio prompt\n      // (39 tokens on Voxtral Mini Realtime); a turn that consumed no audio\n      // reports `prompt_tokens: 1`.\n      const promptTokens = usage?.prompt_tokens;\n      return [\n        {\n          type: 'final',\n          text: typeof event.text === 'string' ? event.text.trim() : '',\n          providerUsage: usage,\n          raw: event,\n          audioConsumed:\n            typeof promptTokens === 'number' ? promptTokens > 1 : undefined,\n        },\n      ];\n    }\n    case 'error':\n      return [\n        {\n          type: 'error',\n          message:\n            typeof event.error === 'string' ? event.error : 'unknown error',\n          code: typeof event.code === 'string' ? event.code : undefined,\n          raw: event,\n        },\n      ];\n    default:\n      return [];\n  }\n}\n\n/**\n * Realtime transcriber for Voxtral Realtime served by vLLM.\n *\n * - Audio: PCM16, 16 kHz, mono only; write whole samples (even byte counts).\n * - Turns: manual only. vLLM has no VAD; it streams partials continuously and\n *   produces one final per `commit()` / `end()`, plus one per automatic\n *   rollover when a turn reaches `maxTurnSeconds` (default 270).\n * - `language` and `prompt` are not sent: vLLM's realtime session accepts\n *   only `model`.\n * - Auth: `apiKey` → `Authorization: Bearer` (Node only). vLLM has no\n *   ephemeral tokens, so browsers must connect through the consumer's own\n *   proxy or token gateway, passing that gateway's short-lived `clientSecret`\n *   (sent as the `openai-insecure-api-key.<secret>` subprotocol next to\n *   `realtime`; the gateway must echo `realtime`).\n */\nexport class VoxtralRealtimeTranscriber implements StreamingTranscriber {\n  readonly type = ADAPTER;\n  readonly audioFormat: StreamingAudioFormat = {\n    ...VOXTRAL_REALTIME_AUDIO_FORMAT,\n  };\n\n  private readonly options: VoxtralRealtimeTranscriberOptions;\n  private readonly url: string;\n  private readonly createWebSocket: SpeechWebSocketFactory;\n  private readonly protocol = voxtralRealtimeProtocol();\n\n  constructor(options: VoxtralRealtimeTranscriberOptions = {}) {\n    assertNoBrowserApiKey(ADAPTER, options.apiKey);\n    if (!options.baseUrl?.trim()) {\n      throw new SpeechConfigurationError(\n        'voxtral-realtime requires baseUrl (the vLLM server, e.g. http://localhost:8000 or its /v1 root)',\n        ADAPTER,\n      );\n    }\n\n    this.options = options;\n    this.url = resolveVoxtralRealtimeUrl(options.baseUrl);\n    this.createWebSocket = resolveWebSocketFactory(ADAPTER, options);\n    assertFormat(options.format);\n    assertTurnDetection(options.turnDetection);\n    this.resolveTurnLimit();\n  }\n\n  turnLimit(\n    options: StreamingSessionOptions = {},\n  ): StreamingTurnLimit | undefined {\n    assertFormat(options.format);\n    return describeTurnLimit(\n      this.resolveTurnLimit(options),\n      VOXTRAL_REALTIME_AUDIO_FORMAT,\n    );\n  }\n\n  private resolveTurnLimit(\n    options: StreamingSessionOptions = {},\n  ): TurnLimit | undefined {\n    return resolveTurnLimit(\n      ADAPTER,\n      this.protocol.maxTurnSeconds,\n      options,\n      this.options,\n    );\n  }\n\n  start(options: StreamingSessionOptions = {}): StreamingSession {\n    assertFormat(options.format);\n    assertTurnDetection(options.turnDetection);\n    const turnLimit = this.resolveTurnLimit(options);\n    const connect = realtimeConnector({\n      adapter: ADAPTER,\n      url: this.url,\n      apiKey: this.options.apiKey,\n      headers: mergeHeaderRecords(this.options.headers, options.headers),\n      clientSecret: options.clientSecret ?? this.options.clientSecret,\n      // A plain vLLM server selects no subprotocol, so offer none unless a\n      // gateway token must travel in one.\n      protocols: (secret) =>\n        secret\n          ? ['realtime', `${CLIENT_SECRET_SUBPROTOCOL_PREFIX}${secret}`]\n          : [],\n    });\n\n    return new RealtimeTranscriptionSession({\n      protocol: this.protocol,\n      config: {\n        model:\n          options.model?.trim() ||\n          this.options.model?.trim() ||\n          VOXTRAL_REALTIME_DEFAULT_MODEL,\n        format: { ...VOXTRAL_REALTIME_AUDIO_FORMAT },\n        turnDetection: { type: 'manual' },\n        language: options.language ?? this.options.language,\n      },\n      connect,\n      createWebSocket: this.createWebSocket,\n      timeoutMs: this.options.timeoutMs ?? DEFAULT_STREAMING_TIMEOUT_MS,\n      connectTimeoutMs:\n        this.options.connectTimeoutMs ?? DEFAULT_STREAMING_CONNECT_TIMEOUT_MS,\n      highWaterMark:\n        this.options.highWaterMark ?? DEFAULT_STREAMING_HIGH_WATER_MARK,\n      maxBufferedBytes:\n        this.options.maxBufferedBytes ?? DEFAULT_STREAMING_MAX_BUFFERED_BYTES,\n      signal: options.signal,\n      onUsage: [this.options.onUsage, options.onUsage],\n      turnLimit,\n    });\n  }\n}\n\n/**\n * Resolves vLLM's realtime WebSocket URL: `http(s)` becomes `ws(s)`, a server\n * root gets `/v1/realtime`, a versioned API root (e.g. a gateway's\n * `/stt/v1`) gets `/realtime`, and a URL ending in `/realtime` is kept.\n */\nexport function resolveVoxtralRealtimeUrl(baseUrl: string): string {\n  return toWebSocketUrl(\n    resolveOpenAICompatibleUrl(baseUrl, 'realtime'),\n    ADAPTER,\n  );\n}\n\nfunction assertFormat(format: Partial<StreamingAudioFormat> | undefined): void {\n  const expected = VOXTRAL_REALTIME_AUDIO_FORMAT;\n  if (\n    format &&\n    ((format.encoding !== undefined && format.encoding !== expected.encoding) ||\n      (format.sampleRate !== undefined &&\n        format.sampleRate !== expected.sampleRate) ||\n      (format.channels !== undefined && format.channels !== expected.channels))\n  ) {\n    throw new SpeechConfigurationError(\n      `voxtral-realtime requires pcm16 at 16000 Hz mono (got ${format.encoding ?? 'pcm16'} at ${format.sampleRate ?? 16_000} Hz, ${format.channels ?? 1} channel(s)); resample before streaming`,\n      ADAPTER,\n    );\n  }\n}\n\nfunction assertTurnDetection(\n  turnDetection: StreamingTurnDetection | undefined,\n): void {\n  if (turnDetection && turnDetection.type !== 'manual') {\n    throw new SpeechConfigurationError(\n      `voxtral-realtime has no server turn detection (got ${turnDetection.type}); use manual turns with commit()/end()`,\n      ADAPTER,\n    );\n  }\n}\n","import {\n  compactJson,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n  resolveOpenAICompatibleUrl,\n  voiceToString,\n} from '../shared/http.js';\nimport type {\n  OpenAICompatibleSpeechSynthesizerOptions,\n  SpeechSynthesizer,\n  SynthesisRequest,\n  SynthesizedSpeech,\n} from '../shared/types.js';\n\nexport class OpenAICompatibleSpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'openai-compatible' as const;\n\n  /** Explicit override, resolved relative to `baseUrl` (or absolute). */\n  private readonly speechPath?: string;\n  private readonly defaultModel: string;\n  private readonly defaultVoice: string;\n\n  constructor(options: OpenAICompatibleSpeechSynthesizerOptions) {\n    super(options);\n    this.speechPath = options.speechPath;\n    this.defaultModel = options.defaultModel ?? 'tts-1';\n    this.defaultVoice = options.defaultVoice ?? 'alloy';\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const outputFormat = request.outputFormat ?? 'mp3';\n    const payload = compactJson({\n      model: request.model ?? this.defaultModel,\n      input: request.text,\n      voice: voiceToString(request.voice, this.defaultVoice),\n      // biome-ignore lint/style/useNamingConvention: OpenAI-compatible API uses snake_case.\n      response_format: outputFormat,\n      speed: request.speed,\n    });\n\n    // Without an override, resolve the endpoint the way the transcriber does:\n    // a server root gets `/v1/audio/speech`, a `/vN` API root gets\n    // `/audio/speech`, and a full endpoint URL is used as-is. Resolved per\n    // call so an invalid baseUrl still fails at synthesize time.\n    const endpoint =\n      this.speechPath ??\n      resolveOpenAICompatibleUrl(this.baseUrl, 'audio/speech');\n\n    const speech = await this.post(\n      this.type,\n      endpoint,\n      {\n        headers: {\n          'content-type': 'application/json',\n        },\n        body: JSON.stringify(payload),\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n\n    return {\n      ...speech,\n      format: speech.format ?? outputFormat,\n      model: speech.model ?? String(payload.model),\n    };\n  }\n}\n","import {\n  DEFAULT_MAX_AUDIO_BYTES,\n  normalizeAudioInput,\n} from '../shared/audio.js';\nimport { SpeechProviderError } from '../shared/errors.js';\nimport {\n  appendOptionalFormValue,\n  HttpSpeechAdapter,\n  readTranscriptResponse,\n  resolveOpenAICompatibleUrl,\n} from '../shared/http.js';\nimport {\n  resolveRetryOptions,\n  type SpeechRetryOptions,\n} from '../shared/retry.js';\nimport type {\n  OpenAICompatibleTranscriberOptions,\n  TimestampGranularity,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptionResponseFormat,\n  TranscriptResult,\n} from '../shared/types.js';\nimport {\n  audioSecondsFromProviderUsage,\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from '../shared/usage.js';\n\nconst DEFAULT_MODEL = 'whisper-1';\n\n/**\n * Models that only accept `json`/`text` and reject `verbose_json` and\n * `timestamp_granularities[]` (e.g. `gpt-4o-transcribe`,\n * `gpt-4o-mini-transcribe`, optionally gateway-prefixed like\n * `openai/gpt-4o-transcribe`).\n */\nconst JSON_ONLY_MODEL = /(^|\\/)gpt-4o[\\w.-]*-transcribe/i;\n\n/** Provider 400/422 bodies that indicate an unsupported format option. */\nconst UNSUPPORTED_FORMAT_ERROR =\n  /verbose_json|response_format|timestamp_granularit/i;\n\nexport function isJsonOnlyTranscriptionModel(model: string): boolean {\n  return JSON_ONLY_MODEL.test(model);\n}\n\ninterface TranscriptionWireOptions {\n  model: string;\n  responseFormat: TranscriptionResponseFormat;\n  timestampGranularities?: TimestampGranularity[];\n}\n\n/**\n * OpenAI-compatible speech-to-text adapter (`POST <base>/audio/transcriptions`).\n * Works with OpenAI, Groq, Fireworks, LiteLLM/Bifrost gateways, vLLM,\n * speaches/faster-whisper-server, and whisper.cpp server.\n *\n * Server-side only: the adapter holds the API key.\n */\nexport class OpenAICompatibleTranscriber\n  extends HttpSpeechAdapter\n  implements Transcriber\n{\n  readonly type = 'openai-compatible' as const;\n\n  private readonly endpoint: string;\n  private readonly model: string;\n  private readonly responseFormat?: TranscriptionResponseFormat;\n  private readonly maxBytes: number;\n  private readonly retry: Required<SpeechRetryOptions> | false;\n  private readonly onUsage?: SpeechUsageCallback;\n\n  constructor(options: OpenAICompatibleTranscriberOptions) {\n    super(options);\n    this.endpoint = resolveOpenAICompatibleUrl(\n      options.baseUrl,\n      'audio/transcriptions',\n    );\n    this.model = options.model?.trim() || DEFAULT_MODEL;\n    this.responseFormat = options.responseFormat;\n    this.maxBytes = options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES;\n    this.retry = resolveRetryOptions(options.retry);\n    this.onUsage = options.onUsage;\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    const audio = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes: request.maxBytes ?? this.maxBytes,\n      adapter: this.type,\n      signal: request.signal,\n    });\n    const wire = this.resolveWireOptions(request);\n\n    const send = (options: TranscriptionWireOptions) => {\n      const form = new FormData();\n      form.append('file', audio.blob, audio.filename);\n      form.set('model', options.model);\n      appendOptionalFormValue(form, 'language', request.language);\n      appendOptionalFormValue(form, 'prompt', request.prompt);\n      appendOptionalFormValue(form, 'temperature', request.temperature);\n      form.set('response_format', options.responseFormat);\n      for (const granularity of options.timestampGranularities ?? []) {\n        form.append('timestamp_granularities[]', granularity);\n      }\n\n      return this.post(\n        this.type,\n        this.endpoint,\n        {\n          body: form,\n          headers: request.headers,\n          signal: request.signal,\n        },\n        (response) => readTranscriptResponse(response, this.type),\n        this.retry,\n      );\n    };\n\n    let transcript: TranscriptResult;\n    try {\n      transcript = await send(wire);\n    } catch (error) {\n      if (!isUnsupportedFormatError(error, wire)) {\n        throw error;\n      }\n      // The server rejected verbose_json/timestamps: fall back to plain JSON.\n      transcript = await send({\n        model: wire.model,\n        responseFormat: 'json',\n      });\n    }\n\n    const providerUsage = extractProviderUsage(transcript.raw);\n    const usage: SpeechUsage = {\n      operation: 'transcription',\n      provider: this.type,\n      model: transcript.model ?? wire.model,\n      audioSeconds:\n        transcript.durationSeconds ??\n        audioSecondsFromProviderUsage(providerUsage) ??\n        audio.durationSeconds,\n      bytes: audio.bytes,\n      providerUsage,\n    };\n    const result: TranscriptResult = {\n      ...transcript,\n      // Plain `text` responses end with a newline; JSON text is kept verbatim.\n      text:\n        transcript.raw === undefined ? transcript.text.trim() : transcript.text,\n      provider: this.type,\n      model: transcript.model ?? wire.model,\n      usage,\n    };\n\n    await reportSpeechUsage(usage, this.onUsage, request.onUsage);\n    return result;\n  }\n\n  private resolveWireOptions(\n    request: TranscriptionRequest,\n  ): TranscriptionWireOptions {\n    const model = request.model?.trim() || this.model;\n    const jsonOnly = isJsonOnlyTranscriptionModel(model);\n    let responseFormat =\n      request.responseFormat ??\n      this.responseFormat ??\n      (jsonOnly ? 'json' : 'verbose_json');\n\n    if (jsonOnly && responseFormat === 'verbose_json') {\n      responseFormat = 'json';\n    }\n\n    const timestampGranularities =\n      responseFormat === 'verbose_json' &&\n      request.timestampGranularities?.length\n        ? [...new Set(request.timestampGranularities)]\n        : undefined;\n\n    return { model, responseFormat, timestampGranularities };\n  }\n}\n\nfunction isUnsupportedFormatError(\n  error: unknown,\n  wire: TranscriptionWireOptions,\n): boolean {\n  return (\n    error instanceof SpeechProviderError &&\n    (error.status === 400 || error.status === 422) &&\n    (wire.responseFormat === 'verbose_json' ||\n      Boolean(wire.timestampGranularities?.length)) &&\n    UNSUPPORTED_FORMAT_ERROR.test(error.responseBody ?? '')\n  );\n}\n\nfunction extractProviderUsage(\n  raw: unknown,\n): Record<string, unknown> | undefined {\n  if (!raw || typeof raw !== 'object') {\n    return undefined;\n  }\n\n  const usage = (raw as Record<string, unknown>).usage;\n  return usage && typeof usage === 'object' && !Array.isArray(usage)\n    ? (usage as Record<string, unknown>)\n    : undefined;\n}\n","import {\n  createHappyVerticalSynthesisForm,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n} from '../shared/http.js';\nimport type {\n  Qwen3SpeechSynthesizerOptions,\n  SpeechSynthesizer,\n  SynthesisRequest,\n  SynthesizedSpeech,\n} from '../shared/types.js';\n\nexport class Qwen3SpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'qwen3-tts' as const;\n\n  private readonly speechPath: string;\n  private readonly defaultModel: string;\n  private readonly defaultVoice?: string;\n\n  constructor(options: Qwen3SpeechSynthesizerOptions) {\n    super(options);\n    this.speechPath = options.speechPath ?? '/v1/audio/speech';\n    this.defaultModel = options.defaultModel ?? 'qwen3-tts';\n    this.defaultVoice = options.defaultVoice;\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const form = createHappyVerticalSynthesisForm(request, this.defaultVoice);\n    const speech = await this.post(\n      this.type,\n      this.speechPath,\n      {\n        body: form,\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n\n    return {\n      ...speech,\n      format: speech.format ?? request.outputFormat,\n      model: speech.model ?? request.model ?? this.defaultModel,\n    };\n  }\n}\n","import { normalizeAudioInput } from '../shared/audio.js';\nimport {\n  appendOptionalFormValue,\n  createHappyVerticalSynthesisForm,\n  HttpSpeechAdapter,\n  readSynthesizedSpeechResponse,\n  readTranscriptResponse,\n} from '../shared/http.js';\nimport type {\n  SpeechSynthesizer,\n  StudioServerSpeechSynthesizerOptions,\n  StudioServerTranscriberOptions,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from '../shared/types.js';\n\nexport class StudioServerTranscriber\n  extends HttpSpeechAdapter\n  implements Transcriber\n{\n  readonly type = 'studio-server' as const;\n  private readonly transcribePath: string;\n\n  constructor(options: StudioServerTranscriberOptions) {\n    super(options);\n    this.transcribePath = options.transcribePath ?? '/v1/transcribe';\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    const audio = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes: request.maxBytes,\n      adapter: this.type,\n      signal: request.signal,\n      deriveExtension: false,\n    });\n    const form = new FormData();\n    form.append('audio', audio.blob, audio.filename);\n    appendOptionalFormValue(form, 'language', request.language);\n\n    return this.post(\n      this.type,\n      this.transcribePath,\n      {\n        body: form,\n        headers: request.headers,\n        signal: request.signal,\n      },\n      (response) => readTranscriptResponse(response, this.type),\n    );\n  }\n}\n\nexport class StudioServerSpeechSynthesizer\n  extends HttpSpeechAdapter\n  implements SpeechSynthesizer\n{\n  readonly type = 'studio-server' as const;\n  private readonly synthesizePath: string;\n  private readonly defaultVoice?: string;\n\n  constructor(options: StudioServerSpeechSynthesizerOptions) {\n    super(options);\n    this.synthesizePath = options.synthesizePath ?? '/v1/tts/synthesize';\n    this.defaultVoice = options.defaultVoice;\n  }\n\n  async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n    const form = createHappyVerticalSynthesisForm(request, this.defaultVoice);\n\n    return this.post(\n      this.type,\n      this.synthesizePath,\n      {\n        body: form,\n        signal: request.signal,\n      },\n      (response) => readSynthesizedSpeechResponse(response, this.type),\n    );\n  }\n}\n","/**\n * Streaming transcriber factory, environment configuration, and the\n * record-then-send `Transcriber` wrapper.\n */\n\nimport { OpenAIRealtimeTranscriber } from '../adapters/openai-realtime.js';\nimport { VoxtralRealtimeTranscriber } from '../adapters/voxtral-realtime.js';\nimport { DEFAULT_MAX_AUDIO_BYTES, normalizeAudioInput } from './audio.js';\nimport {\n  defaultEnv,\n  mergeHeaderInits,\n  parseHeadersEnv,\n  parseOptionalInteger,\n  readEnv,\n  type SpeechEnv,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n  SpeechError,\n} from './errors.js';\nimport { unwrapRawAudio } from './pcm.js';\nimport type {\n  GetStreamingTranscriberOptions,\n  StreamingSessionOptions,\n  StreamingTranscriber,\n  StreamingTranscriberType,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport type {\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from './types.js';\n\n/** Streaming adapters `getStreamingTranscriber()` can build. */\nexport const STREAMING_TRANSCRIBER_TYPES: readonly StreamingTranscriberType[] =\n  ['openai-realtime', 'voxtral-realtime'];\n\n/** Environment variable names read for each streaming setting, in priority order. */\nexport const STREAMING_TRANSCRIBER_ENV_KEYS = {\n  type: ['HAVE_SPEECH_STREAMING_TYPE'],\n  baseUrl: ['HAVE_SPEECH_STREAMING_BASE_URL'],\n  model: ['HAVE_SPEECH_STREAMING_MODEL'],\n  apiKey: ['HAVE_SPEECH_STREAMING_API_KEY'],\n  language: ['HAVE_SPEECH_STREAMING_LANGUAGE'],\n  turnDetection: ['HAVE_SPEECH_STREAMING_TURN_DETECTION'],\n  timeoutMs: [\n    'HAVE_SPEECH_STREAMING_TIMEOUT',\n    'HAVE_SPEECH_STREAMING_TIMEOUT_MS',\n  ],\n  connectTimeoutMs: ['HAVE_SPEECH_STREAMING_CONNECT_TIMEOUT_MS'],\n  headers: ['HAVE_SPEECH_STREAMING_HEADERS'],\n  maxTurnSeconds: ['HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS'],\n} as const;\n\nexport interface StreamingFactoryContext {\n  env?: SpeechEnv;\n  /** Headers merged under explicit `headers` (env → context → options). */\n  headers?: HeadersInit;\n}\n\nexport function isStreamingTranscriberType(\n  type: string | undefined,\n): type is StreamingTranscriberType {\n  return (\n    type !== undefined &&\n    (STREAMING_TRANSCRIBER_TYPES as readonly string[]).includes(type)\n  );\n}\n\n/**\n * Creates a realtime streaming transcriber. Explicit options win over\n * `HAVE_SPEECH_STREAMING_*` environment variables. The type defaults to\n * `openai-realtime`.\n */\nexport function getStreamingTranscriber(\n  options: GetStreamingTranscriberOptions = {},\n  context: StreamingFactoryContext = {},\n): StreamingTranscriber {\n  const env = context.env ?? defaultEnv();\n  const keys = STREAMING_TRANSCRIBER_ENV_KEYS;\n  const type = options.type ?? readEnv(env, ...keys.type) ?? 'openai-realtime';\n\n  if (!isStreamingTranscriberType(type)) {\n    throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n\n  const headers = mergeHeaderInits(\n    parseHeadersEnv(env, keys.headers),\n    context.headers,\n    options.headers,\n  );\n  const resolved = {\n    ...options,\n    type,\n    baseUrl: (options.baseUrl ?? readEnv(env, ...keys.baseUrl))?.trim(),\n    model: options.model ?? readEnv(env, ...keys.model),\n    apiKey: options.apiKey ?? readEnv(env, ...keys.apiKey),\n    language: options.language ?? readEnv(env, ...keys.language),\n    turnDetection:\n      options.turnDetection ??\n      parseTurnDetection(readEnv(env, ...keys.turnDetection)),\n    timeoutMs:\n      options.timeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.timeoutMs)),\n    connectTimeoutMs:\n      options.connectTimeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.connectTimeoutMs)),\n    maxTurnSeconds:\n      options.maxTurnSeconds ??\n      parseMaxTurnSeconds(readEnv(env, ...keys.maxTurnSeconds)),\n    headers,\n  };\n\n  switch (type) {\n    case 'openai-realtime':\n      return new OpenAIRealtimeTranscriber({ ...resolved, type });\n    case 'voxtral-realtime':\n      return new VoxtralRealtimeTranscriber({ ...resolved, type });\n    default:\n      throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n}\n\nexport function parseTurnDetection(\n  value: string | undefined,\n): StreamingTurnDetection | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const normalized = value.toLowerCase();\n  if (normalized === 'server_vad' || normalized === 'semantic_vad') {\n    return { type: normalized };\n  }\n  if (normalized === 'manual' || normalized === 'none') {\n    return { type: 'manual' };\n  }\n  throw new SpeechConfigurationError(\n    'HAVE_SPEECH_STREAMING_TURN_DETECTION must be server_vad, semantic_vad, or manual',\n  );\n}\n\n/**\n * Parses `HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS`: a positive number of\n * seconds, or `Infinity` for no limit.\n */\nexport function parseMaxTurnSeconds(\n  value: string | undefined,\n): number | undefined {\n  if (!value) {\n    return undefined;\n  }\n  const parsed = Number(value);\n  if (Number.isNaN(parsed) || parsed <= 0) {\n    throw new SpeechConfigurationError(\n      'HAVE_SPEECH_STREAMING_MAX_TURN_SECONDS must be a positive number of seconds or Infinity',\n    );\n  }\n  return parsed;\n}\n\nexport interface StreamingTranscriberWrapperOptions {\n  /** Bytes per `write()`. Default 64 KiB. */\n  chunkBytes?: number;\n  /** Upload limit in bytes. Default 25 MB; `Infinity` disables it. */\n  maxBytes?: number;\n  /**\n   * Turn detection for wrapped sessions. Default `manual`: the whole clip is\n   * one turn, committed by `end()`.\n   */\n  turnDetection?: StreamingTurnDetection;\n}\n\n/**\n * Adapts a streaming transcriber to the request/response `Transcriber`\n * interface for record-then-send callers. Each `transcribe()` opens one\n * session, streams the recording, and returns `session.end()`.\n *\n * Input must be raw audio: a PCM16/G.711 WAV file, `audio/pcm` or\n * `audio/L16` (optionally with `rate`/`channels` parameters), `audio/pcmu`,\n * `audio/pcma`, or untyped bytes in the adapter's default format. WAV headers\n * and MIME parameters set the session format; compressed recordings are\n * rejected, so use an HTTP transcriber for WebM/MP4/Ogg.\n *\n * A clip longer than the adapter's per-turn limit (`maxTurnSeconds`) is\n * never sent as one turn: with rollover (the default) the session splits it\n * into several turns at quiet boundaries, and with `rollover: false` the clip\n * is rejected with `SPEECH_TURN_TOO_LONG` before connecting.\n */\nexport function wrapStreamingTranscriber(\n  streaming: StreamingTranscriber,\n  options: StreamingTranscriberWrapperOptions = {},\n): Transcriber {\n  const chunkBytes = options.chunkBytes ?? 64 * 1024;\n  if (!Number.isInteger(chunkBytes) || chunkBytes <= 0) {\n    throw new SpeechConfigurationError(\n      'chunkBytes must be a positive integer',\n      streaming.type,\n    );\n  }\n\n  return {\n    type: streaming.type,\n    async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n      const audio = await normalizeAudioInput(request.audio, {\n        mimeType: request.mimeType,\n        maxBytes:\n          request.maxBytes ?? options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES,\n        adapter: streaming.type,\n        signal: request.signal,\n        deriveExtension: false,\n      });\n      const bytes = new Uint8Array(await audio.blob.arrayBuffer());\n      const raw = unwrapRawAudio(bytes, audio.mimeType, streaming.type);\n      const declared =\n        request.audio &&\n        typeof request.audio === 'object' &&\n        'data' in request.audio\n          ? request.audio\n          : undefined;\n      const format = {\n        ...raw.format,\n        sampleRate: raw.format.sampleRate ?? declared?.sampleRate,\n        channels: raw.format.channels ?? declared?.channels,\n      };\n\n      const sessionOptions: StreamingSessionOptions = {\n        format: Object.fromEntries(\n          Object.entries(format).filter(([, value]) => value !== undefined),\n        ),\n        language: request.language,\n        model: request.model,\n        prompt: request.prompt,\n        headers: request.headers,\n        signal: request.signal,\n        onUsage: request.onUsage,\n        metadata: request.metadata,\n        turnDetection: options.turnDetection ?? { type: 'manual' },\n      };\n      const limit = streaming.turnLimit?.(sessionOptions);\n      if (\n        limit &&\n        !limit.rollover &&\n        raw.bytes.byteLength > limit.maxTurnBytes\n      ) {\n        throw new SpeechError(\n          `The recording is longer than one turn may be (maxTurnSeconds ${limit.maxTurnSeconds} s) and rollover is disabled; enable rollover or split the recording`,\n          'SPEECH_TURN_TOO_LONG',\n          streaming.type,\n        );\n      }\n\n      const session = streaming.start(sessionOptions);\n\n      try {\n        for (\n          let offset = 0;\n          offset < raw.bytes.byteLength;\n          offset += chunkBytes\n        ) {\n          await session.write(raw.bytes.subarray(offset, offset + chunkBytes));\n        }\n      } catch (error) {\n        session.abort(error);\n        throw error;\n      }\n      return session.end();\n    },\n  };\n}\n","import { OpenAICompatibleSpeechSynthesizer } from '../adapters/openai-compatible.js';\nimport { OpenAICompatibleTranscriber } from '../adapters/openai-compatible-transcriber.js';\nimport { Qwen3SpeechSynthesizer } from '../adapters/qwen3.js';\nimport {\n  StudioServerSpeechSynthesizer,\n  StudioServerTranscriber,\n} from '../adapters/studio-server.js';\nimport {\n  defaultEnv,\n  hasAnyEnv,\n  hasTranscriberEnv,\n  parseOptionalInteger,\n  readEnv,\n  resolveTranscriberConfig,\n  type SpeechEnv,\n  TRANSCRIBER_ENV_KEYS,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n} from './errors.js';\nimport { compactJson } from './http.js';\nimport {\n  getStreamingTranscriber,\n  isStreamingTranscriberType,\n  STREAMING_TRANSCRIBER_TYPES,\n  wrapStreamingTranscriber,\n} from './streaming-factory.js';\nimport type {\n  GetStreamingTranscriberOptions,\n  StreamingTranscriberType,\n} from './streaming-types.js';\nimport {\n  getOptionalTranscriberFactory,\n  optionalTranscriberEntry,\n  registeredOptionalTranscribers,\n} from './registry.js';\nimport type {\n  GetSpeechOptions,\n  GetSpeechSynthesizerOptions,\n  GetTranscriberOptions,\n  OpenAICompatibleSpeechSynthesizerOptions,\n  OpenAICompatibleTranscriberOptions,\n  Qwen3SpeechSynthesizerOptions,\n  Speech,\n  SpeechAdapterAvailability,\n  SpeechFetch,\n  SpeechSynthesizer,\n  StudioServerSpeechSynthesizerOptions,\n  StudioServerTranscriberOptions,\n  SynthesisRequest,\n  SynthesizedSpeech,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from './types.js';\n\nexport interface SpeechFactoryContext {\n  env?: SpeechEnv;\n  fetch?: SpeechFetch;\n  headers?: HeadersInit;\n}\n\n/**\n * Creates a speech service from explicit options and/or environment variables.\n *\n * Explicit options win over environment defaults. If no STT or TTS config is\n * available, the returned service is still usable for dependency injection, but\n * calling the missing operation throws a configuration error.\n */\nexport async function getSpeech(\n  options: GetSpeechOptions = {},\n  context: SpeechFactoryContext = {},\n): Promise<Speech> {\n  const transcriber =\n    options.transcriber === false\n      ? undefined\n      : await getOptionalTranscriber(options.transcriber, context);\n  const synthesizer =\n    options.synthesizer === false\n      ? undefined\n      : await getOptionalSpeechSynthesizer(options.synthesizer, context);\n\n  return {\n    transcriber,\n    synthesizer,\n    async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n      if (!transcriber) {\n        throw new SpeechConfigurationError('No STT provider configured');\n      }\n      return transcriber.transcribe(request);\n    },\n    async synthesize(request: SynthesisRequest): Promise<SynthesizedSpeech> {\n      if (!synthesizer) {\n        throw new SpeechConfigurationError('No TTS provider configured');\n      }\n      return synthesizer.synthesize(request);\n    },\n  };\n}\n\nexport async function getTranscriber(\n  options: GetTranscriberOptions = {},\n  context: SpeechFactoryContext = {},\n): Promise<Transcriber> {\n  const streamingType =\n    options.type ??\n    readEnv(context.env ?? defaultEnv(), ...TRANSCRIBER_ENV_KEYS.type);\n  if (isStreamingTranscriberType(streamingType)) {\n    return getWrappedStreamingTranscriber(streamingType, options, context);\n  }\n\n  const optional = await getRegisteredTranscriber(options, context);\n  if (optional) {\n    return optional;\n  }\n\n  const resolved = normalizeTranscriberOptions(options, context);\n\n  switch (resolved.type) {\n    case 'studio-server':\n      return new StudioServerTranscriber(resolved);\n    case 'openai-compatible':\n      return new OpenAICompatibleTranscriber(resolved);\n    default:\n      throw new InvalidSpeechAdapterError(\n        (resolved as { type?: string }).type ?? 'unknown',\n        'STT',\n      );\n  }\n}\n\nexport async function getSpeechSynthesizer(\n  options?: GetSpeechSynthesizerOptions,\n  context: SpeechFactoryContext = {},\n): Promise<SpeechSynthesizer> {\n  const resolved = normalizeSpeechSynthesizerOptions(options, context);\n\n  switch (resolved.type) {\n    case 'studio-server':\n      return new StudioServerSpeechSynthesizer(resolved);\n    case 'qwen3-tts':\n      return new Qwen3SpeechSynthesizer(resolved);\n    case 'openai-compatible':\n      return new OpenAICompatibleSpeechSynthesizer(resolved);\n    default:\n      throw new InvalidSpeechAdapterError(\n        (resolved as { type?: string }).type ?? 'unknown',\n        'TTS',\n      );\n  }\n}\n\nexport function getAvailableSpeechAdapters(): SpeechAdapterAvailability {\n  return {\n    transcribers: [\n      'studio-server',\n      'openai-compatible',\n      ...registeredOptionalTranscribers(),\n      ...STREAMING_TRANSCRIBER_TYPES,\n    ],\n    streamingTranscribers: [...STREAMING_TRANSCRIBER_TYPES],\n    synthesizers: ['studio-server', 'qwen3-tts', 'openai-compatible'],\n  };\n}\n\n/**\n * Builds a streaming adapter from `getTranscriber()` options and wraps it as\n * a record-then-send `Transcriber`. Settings come from explicit options, then\n * `options.streaming`, then `HAVE_SPEECH_STREAMING_*`.\n */\nfunction getWrappedStreamingTranscriber(\n  type: StreamingTranscriberType,\n  options: GetTranscriberOptions,\n  context: SpeechFactoryContext,\n): Transcriber {\n  const streaming = getStreamingTranscriber(\n    {\n      ...options.streaming,\n      ...(compactJson({\n        type,\n        baseUrl: options.baseUrl,\n        apiKey: options.apiKey,\n        model: options.model,\n        headers: options.headers,\n        timeoutMs: options.timeoutMs,\n        onUsage: options.onUsage,\n      }) as GetStreamingTranscriberOptions),\n    },\n    { env: context.env, headers: context.headers },\n  );\n  return wrapStreamingTranscriber(streaming, { maxBytes: options.maxBytes });\n}\n\n/**\n * Builds opt-in adapters (e.g. `local`) registered by their subpath entry.\n * Returns `undefined` for built-in types.\n */\nasync function getRegisteredTranscriber(\n  options: GetTranscriberOptions,\n  context: SpeechFactoryContext,\n): Promise<Transcriber | undefined> {\n  const type = resolveTranscriberConfig(options, context).type;\n  const entry = type ? optionalTranscriberEntry(type) : undefined;\n  if (!type || !entry) {\n    return undefined;\n  }\n\n  const factory = getOptionalTranscriberFactory(type);\n  if (!factory) {\n    throw new SpeechConfigurationError(\n      `The '${type}' transcriber is opt-in: import '${entry}' before calling getTranscriber()`,\n      type,\n    );\n  }\n  return factory({ ...options, type: type as Transcriber['type'] }, context);\n}\n\nasync function getOptionalTranscriber(\n  options: GetTranscriberOptions | undefined,\n  context: SpeechFactoryContext,\n): Promise<Transcriber | undefined> {\n  if (!options && !hasTranscriberEnv(context.env ?? defaultEnv())) {\n    return undefined;\n  }\n\n  return getTranscriber(options, context);\n}\n\nasync function getOptionalSpeechSynthesizer(\n  options: GetSpeechSynthesizerOptions | undefined,\n  context: SpeechFactoryContext,\n): Promise<SpeechSynthesizer | undefined> {\n  if (!options && !hasSynthesizerEnv(context.env ?? defaultEnv())) {\n    return undefined;\n  }\n\n  return getSpeechSynthesizer(options, context);\n}\n\nfunction normalizeTranscriberOptions(\n  options: GetTranscriberOptions = {},\n  context: SpeechFactoryContext,\n): StudioServerTranscriberOptions | OpenAICompatibleTranscriberOptions {\n  const resolved = resolveTranscriberConfig(options, context);\n  const type = resolved.type ?? 'studio-server';\n\n  if (type !== 'studio-server' && type !== 'openai-compatible') {\n    throw new InvalidSpeechAdapterError(type, 'STT');\n  }\n\n  if (!resolved.baseUrl) {\n    throw new SpeechConfigurationError('STT baseUrl is required', type);\n  }\n\n  const shared = {\n    baseUrl: resolved.baseUrl,\n    fetch: resolved.fetch,\n    headers: resolved.headers,\n    apiKey: resolved.apiKey,\n    timeoutMs: resolved.timeoutMs,\n  };\n\n  if (type === 'openai-compatible') {\n    return {\n      ...shared,\n      type,\n      model: resolved.model,\n      responseFormat: options.responseFormat,\n      maxBytes: resolved.maxBytes,\n      retry: options.retry,\n      onUsage: options.onUsage,\n    };\n  }\n\n  return {\n    ...shared,\n    type,\n    transcribePath: options.transcribePath ?? resolved.path,\n  };\n}\n\nfunction normalizeSpeechSynthesizerOptions(\n  options: GetSpeechSynthesizerOptions | undefined,\n  context: SpeechFactoryContext,\n):\n  | StudioServerSpeechSynthesizerOptions\n  | Qwen3SpeechSynthesizerOptions\n  | OpenAICompatibleSpeechSynthesizerOptions {\n  const env = context.env ?? defaultEnv();\n  const type =\n    options?.type ??\n    (readEnv(\n      env,\n      'HAVE_SPEECH_TTS_TYPE',\n      'HAVE_SPEECH_TTS_ADAPTER',\n      'TTS_ADAPTER',\n    ) as GetSpeechSynthesizerOptions['type'] | undefined);\n  const baseUrl =\n    options?.baseUrl ??\n    readEnv(env, 'HAVE_SPEECH_TTS_BASE_URL', 'TTS_BASE_URL');\n\n  if (!type) {\n    throw new SpeechConfigurationError('TTS provider type is required');\n  }\n\n  if (\n    type !== 'studio-server' &&\n    type !== 'qwen3-tts' &&\n    type !== 'openai-compatible'\n  ) {\n    throw new InvalidSpeechAdapterError(type, 'TTS');\n  }\n\n  if (!baseUrl?.trim()) {\n    throw new SpeechConfigurationError('TTS baseUrl is required', type);\n  }\n\n  const shared = {\n    type,\n    baseUrl: baseUrl.trim(),\n    fetch: options?.fetch ?? context.fetch,\n    headers: options?.headers ?? context.headers,\n    apiKey:\n      options?.apiKey ??\n      readEnv(env, 'HAVE_SPEECH_TTS_API_KEY', 'TTS_API_KEY', 'SPEECH_API_KEY'),\n    timeoutMs:\n      options?.timeoutMs ??\n      parseOptionalInteger(\n        readEnv(env, 'HAVE_SPEECH_TTS_TIMEOUT_MS', 'TTS_TIMEOUT_MS'),\n      ),\n  };\n\n  if (type === 'studio-server') {\n    return {\n      ...shared,\n      type,\n      synthesizePath:\n        options?.synthesizePath ??\n        readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  if (type === 'qwen3-tts') {\n    return {\n      ...shared,\n      type,\n      speechPath:\n        options?.speechPath ?? readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultModel:\n        options?.defaultModel ??\n        readEnv(env, 'HAVE_SPEECH_TTS_MODEL', 'TTS_MODEL'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  if (type === 'openai-compatible') {\n    return {\n      ...shared,\n      type,\n      speechPath:\n        options?.speechPath ?? readEnv(env, 'HAVE_SPEECH_TTS_PATH', 'TTS_PATH'),\n      defaultModel:\n        options?.defaultModel ??\n        readEnv(env, 'HAVE_SPEECH_TTS_MODEL', 'TTS_MODEL'),\n      defaultVoice:\n        options?.defaultVoice ??\n        readEnv(env, 'HAVE_SPEECH_TTS_VOICE', 'TTS_VOICE'),\n    };\n  }\n\n  throw new InvalidSpeechAdapterError(type, 'TTS');\n}\n\nfunction hasSynthesizerEnv(env: SpeechEnv): boolean {\n  return hasAnyEnv(\n    env,\n    'HAVE_SPEECH_TTS_TYPE',\n    'HAVE_SPEECH_TTS_ADAPTER',\n    'HAVE_SPEECH_TTS_BASE_URL',\n    'TTS_ADAPTER',\n    'TTS_BASE_URL',\n  );\n}\n","import { VOXTRAL_REALTIME_AUDIO_FORMAT } from '../adapters/voxtral-realtime.js';\nimport type { TranscriberType } from './types.js';\n\n/** What audio a transcriber type accepts from a caller. */\nexport type TranscriberInputFormat =\n  | { kind: 'compressed' }\n  | { kind: 'pcm16'; sampleRate: number; channels: 1 };\n\n/** The `openai-realtime` default wire format: `audio/pcm`, 24 kHz mono. */\nconst OPENAI_REALTIME_PCM_RATE = 24_000;\n\n/**\n * Says what to record for a transcriber type. `pcm16` types reject compressed\n * containers (webm, ogg, mp3) and need 16-bit little-endian mono PCM, raw or in\n * a WAV, at exactly `sampleRate`; use `@happyvertical/speech/browser` or\n * `/pcm` to produce it. `compressed` types decode containers themselves\n * (HTTP APIs, and the on-device transcriber, which decodes with the platform).\n * For `openai-realtime` this is the default format; G.711 is an explicit\n * adapter option and not a capture format.\n */\nexport function transcriberInputFormat(\n  type: TranscriberType,\n): TranscriberInputFormat {\n  switch (type) {\n    case 'voxtral-realtime':\n      return {\n        kind: 'pcm16',\n        sampleRate: VOXTRAL_REALTIME_AUDIO_FORMAT.sampleRate ?? 16_000,\n        channels: 1,\n      };\n    case 'openai-realtime':\n      return {\n        kind: 'pcm16',\n        sampleRate: OPENAI_REALTIME_PCM_RATE,\n        channels: 1,\n      };\n    case 'studio-server':\n    case 'openai-compatible':\n    case 'local':\n      return { kind: 'compressed' };\n    default: {\n      const unknown: never = type;\n      throw new Error(`Unknown transcriber type: ${String(unknown)}`);\n    }\n  }\n}\n","/**\n * Server-side minting of short-lived browser credentials for realtime\n * transcription.\n *\n * A browser must never hold the long-lived provider key. Your server calls\n * {@link createStreamingClientSecret} once per tenant session (after its own\n * auth check), records the returned tenant/session attribution, and hands the\n * browser only `value` and `expiresAt`. The browser passes `value` as the\n * streaming adapter's `clientSecret`.\n *\n * - `openai-realtime`: `POST /v1/realtime/client_secrets` with\n *   `expires_after: { anchor: 'created_at', seconds }` and a\n *   `session.type: \"transcription\"` config, returning an `ek_…` secret\n *   (https://developers.openai.com/api/reference/resources/realtime/subresources/client_secrets/methods/create).\n * - `voxtral-realtime`: vLLM has no ephemeral-token mechanism, so this throws\n *   `SpeechConfigurationError`. Browser Voxtral needs your own proxy or token\n *   gateway in front of vLLM.\n */\n\nimport {\n  OPENAI_REALTIME_DEFAULT_MODEL,\n  OPENAI_REALTIME_DEFAULT_URL,\n  openAIRealtimeTranscriptionSession,\n  resolveOpenAIRealtimeFormat,\n} from '../adapters/openai-realtime.js';\nimport {\n  defaultEnv,\n  mergeHeaderInits,\n  parseHeadersEnv,\n  parseOptionalInteger,\n  readEnv,\n  type SpeechEnv,\n} from './env.js';\nimport {\n  InvalidSpeechAdapterError,\n  SpeechConfigurationError,\n  SpeechProviderError,\n} from './errors.js';\nimport {\n  HttpSpeechAdapter,\n  redactSecret,\n  resolveOpenAICompatibleUrl,\n} from './http.js';\nimport { headerSecrets } from './realtime-auth.js';\nimport {\n  isStreamingTranscriberType,\n  parseTurnDetection,\n  STREAMING_TRANSCRIBER_ENV_KEYS,\n} from './streaming-factory.js';\nimport type {\n  OpenAIRealtimeTranscriberOptions,\n  StreamingAudioFormat,\n  StreamingTranscriberType,\n  StreamingTurnDetection,\n} from './streaming-types.js';\nimport type { SpeechFetch } from './types.js';\nimport { isBrowserRuntime } from './websocket.js';\n\n/** Default client-secret lifetime: long enough to connect, short if leaked. */\nexport const DEFAULT_CLIENT_SECRET_TTL_SECONDS = 60;\n/** OpenAI's accepted `expires_after.seconds` range. */\nexport const CLIENT_SECRET_TTL_RANGE = { min: 10, max: 7200 } as const;\n\nexport interface CreateStreamingClientSecretOptions {\n  /** Streaming adapter. Default `HAVE_SPEECH_STREAMING_TYPE`, then `openai-realtime`. */\n  type?: StreamingTranscriberType;\n  /** Long-lived server key. Default `HAVE_SPEECH_STREAMING_API_KEY`. */\n  apiKey?: string;\n  /**\n   * API base. Accepts the same values as the streaming adapter\n   * (`https://api.openai.com`, `…/v1`, or `wss://…/v1/realtime`). Default\n   * `HAVE_SPEECH_STREAMING_BASE_URL`, then `https://api.openai.com/v1`.\n   */\n  baseUrl?: string;\n  /** Secret lifetime in seconds, 10–7200. Default 60. */\n  ttlSeconds?: number;\n  /**\n   * Tenant the secret is minted for. Echoed in the result for your usage\n   * ledger; OpenAI's transcription client secrets carry no attribution field.\n   */\n  tenantId?: string;\n  /** Your session id (e.g. the dictation session), echoed in the result. */\n  sessionId?: string;\n  /** Extra attribution echoed in the result. Never sent to the provider. */\n  metadata?: Record<string, unknown>;\n  /** Transcription model the secret is bound to. Default `HAVE_SPEECH_STREAMING_MODEL`, then `gpt-4o-transcribe`. */\n  model?: string;\n  language?: string;\n  prompt?: string;\n  /** Default `HAVE_SPEECH_STREAMING_TURN_DETECTION`, then `server_vad`. */\n  turnDetection?: StreamingTurnDetection;\n  /** Default PCM16 24 kHz mono. */\n  format?: Partial<StreamingAudioFormat>;\n  noiseReduction?: OpenAIRealtimeTranscriberOptions['noiseReduction'];\n  include?: string[];\n  /**\n   * Extra `transcription` fields. The typed `model`, `language`, and `prompt`\n   * take precedence, so the secret is bound to the model the result reports.\n   */\n  transcriptionOptions?: Record<string, unknown>;\n  /**\n   * Headers for the mint request, e.g. a per-tenant gateway virtual key\n   * (`x-bf-vk`) for provider-side attribution. Merged over\n   * `HAVE_SPEECH_STREAMING_HEADERS`.\n   */\n  headers?: HeadersInit;\n  fetch?: SpeechFetch;\n  signal?: AbortSignal;\n  /** Request timeout. Default `HAVE_SPEECH_STREAMING_TIMEOUT[_MS]`, then 10000. */\n  timeoutMs?: number;\n}\n\n/** A minted browser credential plus the attribution it was minted for. */\nexport interface StreamingClientSecretResult {\n  type: StreamingTranscriberType;\n  /** The short-lived secret. Send it to the browser as `clientSecret`. */\n  value: string;\n  /** Expiry as epoch milliseconds. */\n  expiresAt: number;\n  ttlSeconds: number;\n  model: string;\n  tenantId?: string;\n  sessionId?: string;\n  metadata?: Record<string, unknown>;\n  /** The provider's echoed session config. */\n  session?: Record<string, unknown>;\n}\n\nexport interface StreamingClientSecretContext {\n  env?: SpeechEnv;\n}\n\nconst OPENAI_CLIENT_SECRET_ENDPOINT = 'realtime/client_secrets';\n\n/**\n * Mints a short-lived, session-scoped client secret for browser streaming.\n * Server-side only: throws in a browser runtime, and for adapters without an\n * ephemeral-token mechanism (`voxtral-realtime`).\n */\nexport async function createStreamingClientSecret(\n  options: CreateStreamingClientSecretOptions = {},\n  context: StreamingClientSecretContext = {},\n): Promise<StreamingClientSecretResult> {\n  const env = context.env ?? defaultEnv();\n  const keys = STREAMING_TRANSCRIBER_ENV_KEYS;\n  const type = options.type ?? readEnv(env, ...keys.type) ?? 'openai-realtime';\n  if (!isStreamingTranscriberType(type)) {\n    throw new InvalidSpeechAdapterError(type, 'streaming STT');\n  }\n\n  if (isBrowserRuntime()) {\n    throw new SpeechConfigurationError(\n      'createStreamingClientSecret runs on your server: it needs the long-lived API key, which must never reach a browser',\n      type,\n    );\n  }\n\n  if (type === 'voxtral-realtime') {\n    throw new SpeechConfigurationError(\n      'voxtral-realtime cannot mint browser credentials: vLLM has no ephemeral-token mechanism. Put your own proxy or token gateway in front of vLLM, mint its short-lived tokens yourself, and never send the vLLM API key to a browser',\n      type,\n    );\n  }\n\n  const ttlSeconds = options.ttlSeconds ?? DEFAULT_CLIENT_SECRET_TTL_SECONDS;\n  if (\n    !Number.isInteger(ttlSeconds) ||\n    ttlSeconds < CLIENT_SECRET_TTL_RANGE.min ||\n    ttlSeconds > CLIENT_SECRET_TTL_RANGE.max\n  ) {\n    throw new SpeechConfigurationError(\n      `ttlSeconds must be an integer from ${CLIENT_SECRET_TTL_RANGE.min} to ${CLIENT_SECRET_TTL_RANGE.max}`,\n      type,\n    );\n  }\n\n  const apiKey = options.apiKey ?? readEnv(env, ...keys.apiKey);\n  if (!apiKey) {\n    throw new SpeechConfigurationError(\n      'createStreamingClientSecret requires apiKey (or HAVE_SPEECH_STREAMING_API_KEY)',\n      type,\n    );\n  }\n\n  const model =\n    options.model?.trim() ||\n    readEnv(env, ...keys.model)?.trim() ||\n    OPENAI_REALTIME_DEFAULT_MODEL;\n  const session = openAIRealtimeTranscriptionSession(\n    {\n      model,\n      format: resolveOpenAIRealtimeFormat(options.format),\n      turnDetection: options.turnDetection ??\n        parseTurnDetection(readEnv(env, ...keys.turnDetection)) ?? {\n          type: 'server_vad',\n        },\n      language: options.language ?? readEnv(env, ...keys.language),\n      prompt: options.prompt,\n    },\n    options,\n  );\n\n  const endpoint = resolveClientSecretUrl(\n    options.baseUrl?.trim() ||\n      readEnv(env, ...keys.baseUrl)?.trim() ||\n      OPENAI_REALTIME_DEFAULT_URL,\n  );\n  const headers = mergeHeaderInits(\n    parseHeadersEnv(env, keys.headers),\n    options.headers,\n  );\n  const client = new ClientSecretClient({\n    baseUrl: endpoint,\n    apiKey,\n    fetch: options.fetch,\n    headers,\n    timeoutMs:\n      options.timeoutMs ??\n      parseOptionalInteger(readEnv(env, ...keys.timeoutMs)) ??\n      10_000,\n  });\n\n  const body = await client\n    .mint(\n      endpoint,\n      {\n        expires_after: { anchor: 'created_at', seconds: ttlSeconds },\n        session,\n      },\n      options.signal,\n    )\n    .catch((error: unknown) => {\n      throw redactProviderError(error, headerSecrets(headers));\n    });\n\n  const value = body.value;\n  if (typeof value !== 'string' || !value) {\n    throw new SpeechProviderError(\n      type,\n      'openai-realtime client secret response did not include a value',\n    );\n  }\n  const expiresAtSeconds = body.expires_at;\n\n  return {\n    type,\n    value,\n    expiresAt:\n      typeof expiresAtSeconds === 'number' && Number.isFinite(expiresAtSeconds)\n        ? expiresAtSeconds * 1000\n        : Date.now() + ttlSeconds * 1000,\n    ttlSeconds,\n    model,\n    tenantId: options.tenantId,\n    sessionId: options.sessionId,\n    metadata: options.metadata,\n    session:\n      body.session && typeof body.session === 'object'\n        ? (body.session as Record<string, unknown>)\n        : undefined,\n  };\n}\n\n/**\n * Resolves the client-secret endpoint from any accepted realtime base:\n * `ws(s)` becomes `http(s)`, a trailing `/realtime` and query are dropped,\n * then `/v1/realtime/client_secrets` (or `/realtime/client_secrets` under a\n * versioned root) is appended.\n */\nexport function resolveClientSecretUrl(baseUrl: string): string {\n  const url = new URL(baseUrl);\n  if (url.protocol === 'ws:') {\n    url.protocol = 'http:';\n  } else if (url.protocol === 'wss:') {\n    url.protocol = 'https:';\n  }\n  url.search = '';\n  url.hash = '';\n  url.pathname = url.pathname.replace(/\\/+$/, '').replace(/\\/realtime$/, '');\n  return resolveOpenAICompatibleUrl(\n    url.toString(),\n    OPENAI_CLIENT_SECRET_ENDPOINT,\n  );\n}\n\n/**\n * The HTTP layer redacts the API key from provider error bodies; this also\n * removes gateway credentials sent in `headers` (e.g. `x-bf-vk`).\n */\nfunction redactProviderError(error: unknown, secrets: string[]): unknown {\n  if (!(error instanceof SpeechProviderError) || secrets.length === 0) {\n    return error;\n  }\n  let responseBody = error.responseBody;\n  let message = error.message;\n  for (const secret of secrets) {\n    responseBody = redactSecret(responseBody, secret);\n    message = redactSecret(message, secret) ?? message;\n  }\n  if (responseBody === error.responseBody && message === error.message) {\n    return error;\n  }\n  return new SpeechProviderError(error.adapter ?? 'openai-realtime', message, {\n    status: error.status,\n    responseBody,\n    retryAfterMs: error.retryAfterMs,\n  });\n}\n\nclass ClientSecretClient extends HttpSpeechAdapter {\n  mint(\n    endpoint: string,\n    payload: Record<string, unknown>,\n    signal: AbortSignal | undefined,\n  ): Promise<Record<string, unknown>> {\n    return this.post(\n      'openai-realtime',\n      endpoint,\n      {\n        body: JSON.stringify(payload),\n        headers: { 'content-type': 'application/json' },\n        signal,\n      },\n      async (response) => {\n        const json = (await response.json().catch(() => undefined)) as unknown;\n        if (!json || typeof json !== 'object' || Array.isArray(json)) {\n          throw new SpeechProviderError(\n            'openai-realtime',\n            'openai-realtime client secret response was not a JSON object',\n          );\n        }\n        return json as Record<string, unknown>;\n      },\n    );\n  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