@happyvertical/speech 0.97.0 → 0.98.0

package/dist/local.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"local.js","names":[],"sources":["../src/adapters/local/env.ts","../src/adapters/local/runtime.ts","../src/adapters/local/abort.ts","../src/adapters/local/audio.ts","../src/adapters/local/transcriber.ts","../src/adapters/local/worker.ts","../src/local.ts"],"sourcesContent":["/**\n * Environment configuration for the `local` transcriber. Explicit options win,\n * then the factory context, then `HAVE_SPEECH_TRANSCRIBER_*` variables (Node).\n */\n\nimport {\n  defaultEnv,\n  readEnv,\n  resolveTranscriberConfig,\n} from '../../shared/env.js';\nimport type { SpeechFactoryContext } from '../../shared/factory.js';\nimport type { LocalTranscriberOptions } from './types.js';\n\n/** Local-only environment variables, in priority order. */\nexport const LOCAL_TRANSCRIBER_ENV_KEYS = {\n  device: ['HAVE_SPEECH_TRANSCRIBER_DEVICE'],\n  dtype: ['HAVE_SPEECH_TRANSCRIBER_DTYPE'],\n  cacheDir: ['HAVE_SPEECH_TRANSCRIBER_CACHE_DIR'],\n  modelHost: ['HAVE_SPEECH_TRANSCRIBER_MODEL_HOST'],\n} as const;\n\n/**\n * Merges explicit options with the environment. `model` and `maxBytes` reuse\n * the shared `HAVE_SPEECH_TRANSCRIBER_MODEL`/`_MAX_BYTES` keys.\n */\nexport function resolveLocalTranscriberOptions(\n  options: LocalTranscriberOptions = {},\n  context: Pick<SpeechFactoryContext, 'env'> = {},\n): LocalTranscriberOptions {\n  const env = context.env ?? defaultEnv();\n  const shared = resolveTranscriberConfig(\n    { model: options.model, maxBytes: options.maxBytes },\n    { env },\n  );\n\n  return {\n    ...options,\n    type: 'local',\n    model: shared.model,\n    maxBytes: shared.maxBytes,\n    device:\n      options.device ?? readEnv(env, ...LOCAL_TRANSCRIBER_ENV_KEYS.device),\n    dtype: options.dtype ?? readEnv(env, ...LOCAL_TRANSCRIBER_ENV_KEYS.dtype),\n    cacheDir:\n      options.cacheDir ?? readEnv(env, ...LOCAL_TRANSCRIBER_ENV_KEYS.cacheDir),\n    modelHost:\n      options.modelHost ??\n      readEnv(env, ...LOCAL_TRANSCRIBER_ENV_KEYS.modelHost),\n  };\n}\n","// biome-ignore-all lint/style/useNamingConvention: transformers.js options and outputs use snake_case and PascalCase exports.\n/**\n * Loads the optional `@huggingface/transformers` peer dependency and builds\n * `automatic-speech-recognition` pipelines from it.\n *\n * This is the only module that names the peer. It is reachable only from the\n * `@happyvertical/speech/local` entry, never from the core entry.\n */\n\nimport { SpeechConfigurationError } from '../../shared/errors.js';\nimport type {\n  LocalTranscriberDevice,\n  LocalTranscriberDtype,\n  LocalTranscriberProgressCallback,\n  LocalTransformersRuntime,\n} from './types.js';\n\nexport const TRANSFORMERS_PACKAGE = '@huggingface/transformers';\n\nexport const TRANSFORMERS_INSTALL_HINT =\n  `Local transcription needs the optional peer dependency ${TRANSFORMERS_PACKAGE}. ` +\n  `Install it with \\`pnpm add ${TRANSFORMERS_PACKAGE}\\` (or npm/yarn equivalent).`;\n\n/** One ASR result as returned by transformers.js. */\nexport interface AsrOutput {\n  text: string;\n  chunks?: Array<{ text: string; timestamp: [number, number | null] }>;\n}\n\n/** Callable `automatic-speech-recognition` pipeline (structural subset). */\nexport interface AsrPipeline {\n  (\n    audio: Float32Array,\n    options?: Record<string, unknown>,\n  ): Promise<AsrOutput | AsrOutput[]>;\n  dispose?: () => Promise<void>;\n  model?: { config?: { model_type?: string } };\n}\n\n/** The parts of the transformers.js module this adapter uses. */\nexport interface TransformersModule {\n  pipeline: (\n    task: 'automatic-speech-recognition',\n    model: string,\n    options: Record<string, unknown>,\n  ) => Promise<AsrPipeline>;\n  env?: Record<string, unknown>;\n  InterruptableStoppingCriteria?: new () => { interrupt(): void };\n}\n\n/**\n * Resolves the transformers.js module: an injected module or loader, else a\n * dynamic import of the peer. A missing peer becomes a\n * `SpeechConfigurationError` with an install hint.\n */\nexport async function loadTransformers(\n  runtime: LocalTransformersRuntime | undefined,\n): Promise<TransformersModule> {\n  let loaded: unknown;\n  try {\n    loaded =\n      typeof runtime === 'function'\n        ? await runtime()\n        : (runtime ?? (await importTransformers()));\n  } catch (error) {\n    if (error instanceof SpeechConfigurationError) {\n      throw error;\n    }\n    const configurationError = new SpeechConfigurationError(\n      `${TRANSFORMERS_INSTALL_HINT} (${errorMessage(error)})`,\n      'local',\n    );\n    configurationError.cause = error;\n    throw configurationError;\n  }\n\n  // Accept both the ES namespace and a CommonJS-style `default` wrapper.\n  const candidate = loaded as\n    | (Partial<TransformersModule> & { default?: TransformersModule })\n    | undefined;\n  const module =\n    typeof candidate?.pipeline === 'function'\n      ? (candidate as TransformersModule)\n      : candidate?.default;\n\n  if (typeof module?.pipeline !== 'function') {\n    throw new SpeechConfigurationError(\n      `${TRANSFORMERS_PACKAGE} did not export pipeline(); expected v4 or later`,\n      'local',\n    );\n  }\n  return module;\n}\n\n/** Isolated so tests can observe whether the peer was ever imported. */\nexport function importTransformers(): Promise<unknown> {\n  return import('@huggingface/transformers');\n}\n\nexport interface CreatePipelineOptions {\n  model: string;\n  device: LocalTranscriberDevice;\n  dtype?: LocalTranscriberDtype;\n  revision?: string;\n  cacheDir?: string;\n  onProgress?: LocalTranscriberProgressCallback;\n}\n\n/**\n * Creates the ASR pipeline. `device: 'auto'` resolves to WebGPU when a\n * browser adapter is available (retrying on WASM if WebGPU fails to\n * initialise) and to CPU elsewhere.\n */\nexport async function createAsrPipeline(\n  transformers: TransformersModule,\n  options: CreatePipelineOptions,\n): Promise<{ pipeline: AsrPipeline; device: string }> {\n  const device =\n    options.device === 'auto' ? await detectAutoDevice() : options.device;\n  const build = (selected: string) =>\n    transformers.pipeline('automatic-speech-recognition', options.model, {\n      device: selected,\n      ...(options.dtype ? { dtype: options.dtype } : {}),\n      ...(options.revision ? { revision: options.revision } : {}),\n      ...(options.cacheDir ? { cache_dir: options.cacheDir } : {}),\n      ...(options.onProgress ? { progress_callback: options.onProgress } : {}),\n    });\n\n  try {\n    return { pipeline: await build(device), device };\n  } catch (error) {\n    if (options.device === 'auto' && device === 'webgpu') {\n      return { pipeline: await build('wasm'), device: 'wasm' };\n    }\n    throw error;\n  }\n}\n\n/** `webgpu` when a browser exposes a usable adapter, `wasm` in other browsers, `cpu` in Node. */\nexport async function detectAutoDevice(): Promise<string> {\n  if (isNodeRuntime()) {\n    return 'cpu';\n  }\n\n  const gpu = (\n    globalThis.navigator as\n      | { gpu?: { requestAdapter(): Promise<unknown> } }\n      | undefined\n  )?.gpu;\n  if (gpu) {\n    try {\n      if (await gpu.requestAdapter()) {\n        return 'webgpu';\n      }\n    } catch {\n      // No usable adapter: fall through to WASM.\n    }\n  }\n  return 'wasm';\n}\n\nexport function isNodeRuntime(): boolean {\n  return Boolean(\n    (globalThis as { process?: { versions?: { node?: string } } }).process\n      ?.versions?.node,\n  );\n}\n\nfunction errorMessage(error: unknown): string {\n  return error instanceof Error ? error.message : String(error);\n}\n","/**\n * Rejects with the signal's reason as soon as it aborts, without waiting for\n * `promise` (which keeps running; its eventual result is ignored).\n */\nexport function raceAbort<T>(\n  promise: Promise<T>,\n  signal: AbortSignal | undefined,\n): Promise<T> {\n  if (!signal) {\n    return promise;\n  }\n  if (signal.aborted) {\n    // Mark the abandoned promise handled so a later rejection is not reported.\n    promise.catch(() => undefined);\n    return Promise.reject(signal.reason);\n  }\n  return new Promise<T>((resolve, reject) => {\n    const onAbort = () => reject(signal.reason);\n    signal.addEventListener('abort', onAbort, { once: true });\n    promise.then(\n      (value) => {\n        signal.removeEventListener('abort', onAbort);\n        resolve(value);\n      },\n      (error: unknown) => {\n        signal.removeEventListener('abort', onAbort);\n        reject(error);\n      },\n    );\n  });\n}\n","/**\n * Audio decoding for on-device inference: turns encoded bytes into 16 kHz\n * mono Float32 PCM, the input Whisper and Moonshine expect.\n *\n * - WAV (PCM 8/16/24/32-bit, IEEE float 32/64, WAVE_FORMAT_EXTENSIBLE) and raw\n *   `audio/pcm` are decoded in-process in every runtime, including workers.\n * - Anything else goes to the caller's `decodeAudio` hook, or, in browsers,\n *   to `OfflineAudioContext.decodeAudioData` (MediaRecorder WebM/Opus, Safari\n *   `audio/mp4`, MP3, ...).\n */\n\nimport { normalizeMimeType } from '../../shared/audio.js';\nimport { SpeechConfigurationError } from '../../shared/errors.js';\nimport { raceAbort } from './abort.js';\nimport type { DecodedAudio, LocalAudioDecoder } from './types.js';\n\n/** Sample rate expected by Whisper and Moonshine feature extractors. */\nexport const LOCAL_SAMPLE_RATE = 16_000;\n\nconst ADAPTER = 'local';\n\nconst WAV_MIME_TYPES = new Set([\n  'audio/wav',\n  'audio/wave',\n  'audio/x-wav',\n  'audio/vnd.wave',\n]);\nconst PCM_MIME_TYPES = new Set(['audio/pcm', 'audio/x-pcm', 'audio/raw']);\n\nexport interface DecodeToPcmOptions {\n  /** Full MIME type including parameters (`audio/pcm;rate=16000`). */\n  mimeType: string;\n  /** Sample rate for raw PCM when the MIME type has no `rate` parameter. */\n  sampleRate?: number;\n  /** Channel count for raw PCM when the MIME type has no `channels` parameter. */\n  channels?: number;\n  decodeAudio?: LocalAudioDecoder;\n  signal?: AbortSignal;\n}\n\n/**\n * Decodes `bytes` and returns 16 kHz mono Float32 samples.\n *\n * Raw PCM (`audio/pcm`) is little-endian signed 16-bit by default; pass\n * `encoding=f32le` for Float32. The rate comes from the `rate` MIME parameter\n * or `sampleRate`, the channel count from `channels` (default 1).\n */\nexport async function decodeToPcm16k(\n  bytes: Uint8Array,\n  options: DecodeToPcmOptions,\n): Promise<Float32Array> {\n  // Decoders (caller hooks, decodeAudioData) cannot be cancelled: stop waiting on abort.\n  const decoded = await raceAbort(\n    decodeAudioBytes(bytes, options),\n    options.signal,\n  );\n  return toPcm16k(decoded);\n}\n\n/** Downmixes decoded audio to mono and resamples it to 16 kHz. */\nexport function toPcm16k(decoded: DecodedAudio): Float32Array {\n  return resample(\n    downmix(decoded.samples),\n    decoded.sampleRate,\n    LOCAL_SAMPLE_RATE,\n  );\n}\n\n/**\n * True for WAV and raw PCM, which this module decodes itself in every\n * runtime, including Web Workers.\n */\nexport function isInProcessDecodable(\n  bytes: Uint8Array,\n  mimeType: string,\n): boolean {\n  const essence = normalizeMimeType(mimeType);\n  return Boolean(\n    (essence && (WAV_MIME_TYPES.has(essence) || PCM_MIME_TYPES.has(essence))) ||\n      isRiffWave(bytes),\n  );\n}\n\n/**\n * Decodes a format without an in-process decoder through the caller's\n * `decodeAudio` hook or, in browsers, `OfflineAudioContext`. Resolves\n * `undefined` when neither is available.\n */\nexport async function decodeExternally(\n  bytes: Uint8Array,\n  options: Omit<DecodeToPcmOptions, 'sampleRate' | 'channels'>,\n): Promise<DecodedAudio | undefined> {\n  if (options.decodeAudio) {\n    const decoded = await options.decodeAudio({\n      bytes,\n      mimeType: options.mimeType,\n      signal: options.signal,\n    });\n    assertDecodedAudio(decoded);\n    return decoded;\n  }\n\n  if (hasOfflineAudioContext()) {\n    return decodeWithAudioContext(bytes);\n  }\n\n  return undefined;\n}\n\nasync function decodeAudioBytes(\n  bytes: Uint8Array,\n  options: DecodeToPcmOptions,\n): Promise<DecodedAudio> {\n  const essence = normalizeMimeType(options.mimeType);\n\n  if ((essence && WAV_MIME_TYPES.has(essence)) || isRiffWave(bytes)) {\n    return decodeWav(bytes);\n  }\n\n  if (essence && PCM_MIME_TYPES.has(essence)) {\n    return decodeRawPcm(bytes, options);\n  }\n\n  const decoded = await decodeExternally(bytes, options);\n  if (decoded) {\n    return decoded;\n  }\n\n  throw new SpeechConfigurationError(\n    `Cannot decode '${options.mimeType}' audio on this runtime: pass WAV or audio/pcm, ` +\n      'or supply a decodeAudio hook (e.g. ffmpeg) for compressed formats',\n    ADAPTER,\n  );\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\n/** Parses a RIFF/WAVE file into per-channel Float32 samples. */\nexport function decodeWav(bytes: Uint8Array): DecodedAudio {\n  if (!isRiffWave(bytes)) {\n    throw new SpeechConfigurationError(\n      'Invalid WAV data: missing RIFF/WAVE header',\n      ADAPTER,\n    );\n  }\n\n  const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);\n  let format: WavFormat | undefined;\n  let offset = 12;\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 ') {\n      format = readWavFormat(view, body, size);\n    } else if (id === 'data') {\n      if (!format) {\n        throw new SpeechConfigurationError(\n          'Invalid WAV data: data chunk before fmt chunk',\n          ADAPTER,\n        );\n      }\n      // Streamed WAVs may leave the size as 0 or 0xFFFFFFFF; clamp to the bytes we have.\n      const available = bytes.byteLength - body;\n      const length = size === 0 || size > available ? available : size;\n      return {\n        samples: deinterleave(\n          bytes.subarray(body, body + length),\n          format.channels,\n          format.encoding,\n        ),\n        sampleRate: format.sampleRate,\n      };\n    }\n\n    offset = body + size + (size % 2);\n  }\n\n  throw new SpeechConfigurationError(\n    'Invalid WAV data: no data chunk',\n    ADAPTER,\n  );\n}\n\ntype PcmEncoding = 'u8' | 's16le' | 's24le' | 's32le' | 'f32le' | 'f64le';\n\ninterface WavFormat {\n  channels: number;\n  sampleRate: number;\n  encoding: PcmEncoding;\n}\n\nconst WAVE_FORMAT_PCM = 1;\nconst WAVE_FORMAT_IEEE_FLOAT = 3;\nconst WAVE_FORMAT_EXTENSIBLE = 0xfffe;\n\nfunction readWavFormat(view: DataView, body: number, size: number): WavFormat {\n  if (size < 16) {\n    throw new SpeechConfigurationError(\n      'Invalid WAV data: fmt chunk too short',\n      ADAPTER,\n    );\n  }\n\n  let tag = 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\n  if (tag === WAVE_FORMAT_EXTENSIBLE && size >= 26) {\n    // The first two bytes of the SubFormat GUID carry the real format tag.\n    tag = view.getUint16(body + 24, true);\n  }\n\n  const encoding =\n    tag === WAVE_FORMAT_PCM\n      ? ({ 8: 'u8', 16: 's16le', 24: 's24le', 32: 's32le' } as const)[\n          bitsPerSample\n        ]\n      : tag === WAVE_FORMAT_IEEE_FLOAT\n        ? ({ 32: 'f32le', 64: 'f64le' } as const)[bitsPerSample]\n        : undefined;\n\n  if (!encoding || channels < 1 || sampleRate < 1) {\n    throw new SpeechConfigurationError(\n      `Unsupported WAV encoding (format ${tag}, ${bitsPerSample}-bit, ${channels} channels); ` +\n        'use PCM or IEEE float, or supply a decodeAudio hook',\n      ADAPTER,\n    );\n  }\n\n  return { channels, sampleRate, encoding };\n}\n\nfunction decodeRawPcm(\n  bytes: Uint8Array,\n  options: DecodeToPcmOptions,\n): DecodedAudio {\n  const params = mimeParameters(options.mimeType);\n  const sampleRate =\n    positiveInteger(params.rate) ??\n    explicitPositiveInteger(options.sampleRate, 'sampleRate');\n  const channels =\n    positiveInteger(params.channels) ??\n    explicitPositiveInteger(options.channels, 'channels') ??\n    1;\n  const encoding = (params.encoding ?? 's16le').toLowerCase();\n\n  if (!sampleRate) {\n    throw new SpeechConfigurationError(\n      \"Raw PCM audio needs a sample rate: use 'audio/pcm;rate=16000' or AudioInput.sampleRate\",\n      ADAPTER,\n    );\n  }\n  if (encoding !== 's16le' && encoding !== 'f32le') {\n    throw new SpeechConfigurationError(\n      `Unsupported raw PCM encoding '${encoding}' (expected s16le or f32le)`,\n      ADAPTER,\n    );\n  }\n\n  return {\n    samples: deinterleave(bytes, channels, encoding),\n    sampleRate,\n  };\n}\n\nconst BYTES_PER_SAMPLE: Record<PcmEncoding, number> = {\n  u8: 1,\n  s16le: 2,\n  s24le: 3,\n  s32le: 4,\n  f32le: 4,\n  f64le: 8,\n};\n\nfunction deinterleave(\n  bytes: Uint8Array,\n  channels: number,\n  encoding: PcmEncoding,\n): Float32Array[] {\n  const width = BYTES_PER_SAMPLE[encoding];\n  const frames = Math.floor(bytes.byteLength / (width * channels));\n  const view = new DataView(bytes.buffer, bytes.byteOffset, bytes.byteLength);\n  const output = Array.from(\n    { length: channels },\n    () => new Float32Array(frames),\n  );\n\n  for (let frame = 0; frame < frames; frame++) {\n    for (let channel = 0; channel < channels; channel++) {\n      const at = (frame * channels + channel) * width;\n      (output[channel] as Float32Array)[frame] = readSample(view, at, encoding);\n    }\n  }\n\n  return output;\n}\n\nfunction readSample(view: DataView, at: number, encoding: PcmEncoding): number {\n  switch (encoding) {\n    case 'u8':\n      return (view.getUint8(at) - 128) / 128;\n    case 's16le':\n      return view.getInt16(at, true) / 32_768;\n    case 's24le': {\n      const value =\n        view.getUint8(at) |\n        (view.getUint8(at + 1) << 8) |\n        (view.getInt8(at + 2) << 16);\n      return value / 8_388_608;\n    }\n    case 's32le':\n      return view.getInt32(at, true) / 2_147_483_648;\n    case 'f32le':\n      return view.getFloat32(at, true);\n    case 'f64le':\n      return view.getFloat64(at, true);\n  }\n}\n\n/** Averages channels into one mono track. */\nexport function downmix(samples: Float32Array | Float32Array[]): Float32Array {\n  if (samples instanceof Float32Array) {\n    return samples;\n  }\n  if (samples.length === 1) {\n    return samples[0] as Float32Array;\n  }\n\n  const length = Math.min(...samples.map((channel) => channel.length));\n  const mono = new Float32Array(length);\n  for (const channel of samples) {\n    for (let index = 0; index < length; index++) {\n      (mono as Float32Array)[index] =\n        (mono[index] as number) + (channel[index] as number) / samples.length;\n    }\n  }\n  return mono;\n}\n\n/**\n * Resamples mono audio with linear interpolation. When downsampling, a box\n * filter one source period wide is applied first to limit aliasing; this is\n * adequate for speech recognition, not for music playback.\n */\nexport function resample(\n  samples: Float32Array,\n  fromRate: number,\n  toRate: number,\n): Float32Array {\n  if (fromRate === toRate || samples.length === 0) {\n    return samples;\n  }\n\n  const ratio = fromRate / toRate;\n  const source = ratio > 1 ? boxFilter(samples, Math.round(ratio)) : samples;\n  const length = Math.max(1, Math.round(samples.length / ratio));\n  const output = new Float32Array(length);\n\n  for (let index = 0; index < length; index++) {\n    const position = index * ratio;\n    const left = Math.floor(position);\n    const right = Math.min(left + 1, source.length - 1);\n    const fraction = position - left;\n    output[index] =\n      (source[left] as number) * (1 - fraction) +\n      (source[right] as number) * fraction;\n  }\n\n  return output;\n}\n\nfunction boxFilter(samples: Float32Array, width: number): Float32Array {\n  if (width <= 1) {\n    return samples;\n  }\n\n  const output = new Float32Array(samples.length);\n  const half = Math.floor(width / 2);\n  let sum = 0;\n  let count = 0;\n  let start = 0;\n  let end = 0;\n\n  for (let index = 0; index < samples.length; index++) {\n    const windowStart = Math.max(0, index - half);\n    const windowEnd = Math.min(samples.length, index - half + width);\n    while (end < windowEnd) {\n      sum += samples[end] as number;\n      count++;\n      end++;\n    }\n    while (start < windowStart) {\n      sum -= samples[start] as number;\n      count--;\n      start++;\n    }\n    output[index] = sum / count;\n  }\n\n  return output;\n}\n\nfunction hasOfflineAudioContext(): boolean {\n  return typeof globalThis.OfflineAudioContext === 'function';\n}\n\n/**\n * Browser decode path. Decoding through a 16 kHz `OfflineAudioContext` makes\n * the browser resample with its own high-quality resampler. Not available in\n * Web Workers: decode on the main thread (see the worker client).\n */\nasync function decodeWithAudioContext(\n  bytes: Uint8Array,\n): Promise<DecodedAudio> {\n  const context = new OfflineAudioContext(1, 1, LOCAL_SAMPLE_RATE);\n  // decodeAudioData detaches its input, so hand it a private copy.\n  const copy = new Uint8Array(bytes.byteLength);\n  copy.set(bytes);\n\n  let buffer: AudioBuffer;\n  try {\n    buffer = await context.decodeAudioData(copy.buffer);\n  } catch (error) {\n    throw new SpeechConfigurationError(\n      `The browser could not decode the audio: ${errorMessage(error)}`,\n      ADAPTER,\n    );\n  }\n\n  const channels = Array.from({ length: buffer.numberOfChannels }, (_, index) =>\n    buffer.getChannelData(index),\n  );\n  return { samples: channels, sampleRate: buffer.sampleRate };\n}\n\nfunction assertDecodedAudio(decoded: DecodedAudio): void {\n  const channels =\n    decoded?.samples instanceof Float32Array\n      ? [decoded.samples]\n      : decoded?.samples;\n  if (\n    !Array.isArray(channels) ||\n    channels.length === 0 ||\n    !channels.every((channel) => channel instanceof Float32Array) ||\n    !(decoded.sampleRate > 0)\n  ) {\n    throw new SpeechConfigurationError(\n      'decodeAudio must return { samples: Float32Array | Float32Array[], sampleRate }',\n      ADAPTER,\n    );\n  }\n}\n\nfunction mimeParameters(mimeType: string): Record<string, string> {\n  const params: Record<string, string> = {};\n  for (const part of mimeType.split(';').slice(1)) {\n    const [key, value] = part.split('=');\n    if (key && value) {\n      params[key.trim().toLowerCase()] = value.trim().replace(/^\"|\"$/g, '');\n    }\n  }\n  return params;\n}\n\nfunction positiveInteger(value: string | undefined): number | undefined {\n  const parsed = value ? Number.parseInt(value, 10) : Number.NaN;\n  return Number.isFinite(parsed) && parsed > 0 ? parsed : undefined;\n}\n\n/** Validates a caller-supplied `AudioInput.sampleRate`/`channels`. */\nfunction explicitPositiveInteger(\n  value: number | undefined,\n  name: 'sampleRate' | 'channels',\n): number | undefined {\n  if (value === undefined || value === null) {\n    return undefined;\n  }\n  if (!Number.isInteger(value) || value < 1) {\n    throw new SpeechConfigurationError(\n      `AudioInput.${name} must be a positive integer, got ${String(value)}`,\n      ADAPTER,\n    );\n  }\n  return value;\n}\n\nfunction ascii(bytes: Uint8Array, offset: number, length: number): string {\n  return String.fromCharCode(...bytes.subarray(offset, offset + length));\n}\n\nfunction errorMessage(error: unknown): string {\n  return error instanceof Error ? error.message : String(error);\n}\n","import {\n  DEFAULT_MAX_AUDIO_BYTES,\n  normalizeAudioInput,\n} from '../../shared/audio.js';\nimport {\n  SpeechConfigurationError,\n  SpeechProviderError,\n} from '../../shared/errors.js';\nimport type {\n  AudioInput,\n  AudioSource,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n  TranscriptSegment,\n  WordTiming,\n} from '../../shared/types.js';\nimport {\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from '../../shared/usage.js';\nimport { raceAbort } from './abort.js';\nimport { decodeToPcm16k, LOCAL_SAMPLE_RATE } from './audio.js';\nimport {\n  type AsrOutput,\n  type AsrPipeline,\n  createAsrPipeline,\n  loadTransformers,\n  type TransformersModule,\n} from './runtime.js';\nimport type { LocalTranscriberOptions } from './types.js';\n\n/** Small multilingual Whisper (~80 MB at the default q8 quantisation). */\nexport const DEFAULT_LOCAL_MODEL = 'onnx-community/whisper-base';\n\nconst DEFAULT_CHUNK_LENGTH_SECONDS = 30;\nconst WHISPER_WINDOW_SECONDS = 30;\nconst WHISPER_MODEL_TYPES = new Set(['whisper', 'lite-whisper']);\n\ninterface LoadedPipeline {\n  pipeline: AsrPipeline;\n  device: string;\n}\n\n/**\n * On-device speech-to-text with transformers.js (Whisper or Moonshine ONNX).\n * Runs on WebGPU/WASM in browsers and onnxruntime-node in Node; no audio\n * leaves the device and no API key is involved.\n */\nexport class LocalTranscriber implements Transcriber {\n  readonly type = 'local' as const;\n\n  private readonly options: LocalTranscriberOptions;\n  private readonly model: string;\n  private readonly maxBytes: number;\n  private readonly onUsage?: SpeechUsageCallback;\n  private runtime?: Promise<TransformersModule>;\n  private readonly pipelines = new Map<string, Promise<LoadedPipeline>>();\n  /** Serialises inference: ONNX sessions (especially WebGPU) run one call at a time. */\n  private queue: Promise<unknown> = Promise.resolve();\n\n  constructor(options: LocalTranscriberOptions = {}) {\n    this.options = options;\n    this.model = options.model?.trim() || DEFAULT_LOCAL_MODEL;\n    this.maxBytes = options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES;\n    this.onUsage = options.onUsage;\n  }\n\n  /**\n   * Downloads (first time) and initialises the model so the first\n   * `transcribe()` is fast. Progress goes to `onProgress`.\n   */\n  async preload(model?: string, signal?: AbortSignal): Promise<void> {\n    await raceAbort(this.loadPipeline(model?.trim() || this.model), signal);\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    const { signal } = request;\n    const audio = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes: request.maxBytes ?? this.maxBytes,\n      adapter: this.type,\n      signal,\n    });\n    const format = pcmFormat(request.audio);\n    const samples = await decodeToPcm16k(\n      new Uint8Array(await audio.blob.arrayBuffer()),\n      {\n        mimeType: audio.mimeType,\n        sampleRate: format.sampleRate,\n        channels: format.channels,\n        decodeAudio: this.options.decodeAudio,\n        signal,\n      },\n    );\n    signal?.throwIfAborted();\n\n    const model = request.model?.trim() || this.model;\n    const loaded = await raceAbort(this.loadPipeline(model), signal);\n    const audioSeconds = samples.length / LOCAL_SAMPLE_RATE;\n    // The queue tracks the real inference; the caller stops waiting on abort.\n    const output = await raceAbort(\n      this.enqueue(() =>\n        this.run(loaded.pipeline, samples, audioSeconds, request),\n      ),\n      signal,\n    );\n\n    const usage: SpeechUsage = {\n      operation: 'transcription',\n      provider: this.type,\n      model,\n      audioSeconds,\n      bytes: audio.bytes,\n    };\n    const result: TranscriptResult = {\n      ...mapAsrOutput(\n        output,\n        request.timestampGranularities?.includes('word') ?? false,\n      ),\n      language: request.language,\n      durationSeconds: audioSeconds,\n      provider: this.type,\n      model,\n      raw: output,\n      usage,\n    };\n\n    await reportSpeechUsage(usage, this.onUsage, request.onUsage);\n    return result;\n  }\n\n  /** Waits for in-flight inference, then releases every loaded model session. */\n  async dispose(): Promise<void> {\n    await this.queue;\n    const loaded = [...this.pipelines.values()];\n    this.pipelines.clear();\n    await Promise.all(\n      loaded.map(async (entry) => {\n        const { pipeline } = await entry.catch(() => ({ pipeline: undefined }));\n        await pipeline?.dispose?.();\n      }),\n    );\n  }\n\n  private async run(\n    pipeline: AsrPipeline,\n    samples: Float32Array,\n    audioSeconds: number,\n    request: TranscriptionRequest,\n  ): Promise<AsrOutput> {\n    const { signal } = request;\n    signal?.throwIfAborted();\n\n    const transformers = await this.loadRuntime();\n    const options: Record<string, unknown> = {};\n    const isWhisper = WHISPER_MODEL_TYPES.has(\n      pipeline.model?.config?.model_type ?? 'whisper',\n    );\n\n    if (isWhisper) {\n      const granularities = request.timestampGranularities ?? [];\n      if (granularities.includes('word')) {\n        options.return_timestamps = 'word';\n      } else if (granularities.includes('segment')) {\n        options.return_timestamps = true;\n      }\n      const chunkLength =\n        this.options.chunkLengthSeconds ?? DEFAULT_CHUNK_LENGTH_SECONDS;\n      if (chunkLength > 0 && audioSeconds > WHISPER_WINDOW_SECONDS) {\n        options.chunk_length_s = chunkLength;\n      }\n      if (request.language) {\n        options.language = request.language;\n        options.task = 'transcribe';\n      }\n    }\n\n    // Stops generation between tokens when the caller aborts.\n    let stopping: { interrupt(): void } | undefined;\n    if (signal && transformers.InterruptableStoppingCriteria) {\n      stopping = new transformers.InterruptableStoppingCriteria();\n      options.stopping_criteria = stopping;\n    }\n    const onAbort = () => stopping?.interrupt();\n    signal?.addEventListener('abort', onAbort, { once: true });\n\n    try {\n      const output = await pipeline(samples, options);\n      return Array.isArray(output) ? (output[0] ?? { text: '' }) : output;\n    } catch (error) {\n      if (signal?.aborted) {\n        throw signal.reason;\n      }\n      const hint =\n        options.return_timestamps === 'word' &&\n        /cross attentions/i.test(errorMessage(error))\n          ? ' (word timestamps need a model exported with cross attentions, e.g. onnx-community/whisper-base_timestamped)'\n          : '';\n      throw new SpeechProviderError(\n        this.type,\n        `Local transcription failed: ${errorMessage(error)}${hint}`,\n        { cause: error },\n      );\n    } finally {\n      signal?.removeEventListener('abort', onAbort);\n    }\n  }\n\n  private enqueue<T>(task: () => Promise<T>): Promise<T> {\n    const next = this.queue.then(task, task);\n    this.queue = next.catch(() => undefined);\n    return next;\n  }\n\n  private loadRuntime(): Promise<TransformersModule> {\n    if (!this.runtime) {\n      const { transformers, configureEnv, modelHost } = this.options;\n      this.runtime = loadTransformers(transformers).then((module) => {\n        if (module.env) {\n          if (modelHost) {\n            module.env.remoteHost = modelHost;\n          }\n          configureEnv?.(module.env);\n        }\n        return module;\n      });\n      // A failed load (e.g. missing peer) is retried on the next call.\n      this.runtime.catch(() => {\n        this.runtime = undefined;\n      });\n    }\n    return this.runtime;\n  }\n\n  private loadPipeline(model: string): Promise<LoadedPipeline> {\n    let loaded = this.pipelines.get(model);\n    if (!loaded) {\n      loaded = this.loadRuntime().then((transformers) =>\n        createAsrPipeline(transformers, {\n          model,\n          device: this.options.device ?? 'auto',\n          dtype: this.options.dtype,\n          revision: this.options.revision,\n          cacheDir: this.options.cacheDir,\n          onProgress: this.options.onProgress,\n        }).catch((error: unknown) => {\n          if (error instanceof SpeechConfigurationError) {\n            throw error;\n          }\n          throw new SpeechProviderError(\n            this.type,\n            `Failed to load local model '${model}': ${errorMessage(error)}`,\n            { cause: error },\n          );\n        }),\n      );\n      this.pipelines.set(model, loaded);\n      loaded.catch(() => {\n        if (this.pipelines.get(model) === loaded) {\n          this.pipelines.delete(model);\n        }\n      });\n    }\n    return loaded;\n  }\n}\n\n/**\n * Maps transformers.js chunks onto `words` (when word timestamps were\n * requested) or `segments`.\n */\nexport function mapAsrOutput(\n  output: AsrOutput,\n  wordLevel: boolean,\n): Pick<TranscriptResult, 'text' | 'segments' | 'words'> {\n  const result: Pick<TranscriptResult, 'text' | 'segments' | 'words'> = {\n    text: (output.text ?? '').trim(),\n  };\n  const chunks = output.chunks;\n  if (!chunks?.length) {\n    return result;\n  }\n\n  if (wordLevel) {\n    result.words = chunks\n      .filter((chunk) => chunk.text.trim())\n      .map(\n        (chunk): WordTiming => ({\n          word: chunk.text.trim(),\n          startSeconds: chunk.timestamp[0],\n          endSeconds: chunk.timestamp[1] ?? chunk.timestamp[0],\n        }),\n      );\n  } else {\n    result.segments = chunks.map(\n      (chunk): TranscriptSegment => ({\n        text: chunk.text.trim(),\n        startSeconds: chunk.timestamp[0],\n        endSeconds: chunk.timestamp[1] ?? undefined,\n      }),\n    );\n  }\n  return result;\n}\n\nfunction pcmFormat(audio: AudioInput | AudioSource): {\n  sampleRate?: number;\n  channels?: number;\n} {\n  if (\n    audio &&\n    typeof audio === 'object' &&\n    'data' in audio &&\n    !(audio instanceof Uint8Array)\n  ) {\n    return { sampleRate: audio.sampleRate, channels: audio.channels };\n  }\n  return {};\n}\n\nfunction errorMessage(error: unknown): string {\n  return error instanceof Error ? error.message : String(error);\n}\n","/**\n * Web Worker support for the `local` transcriber.\n *\n * Inference runs in a worker (`serveLocalTranscriber`) so long recordings never\n * block the UI thread. The main-thread client (`LocalTranscriberWorkerClient`)\n * transfers WAV and raw PCM bytes to the worker, which parses, downmixes, and\n * resamples them there. Only formats that need `AudioContext` (which workers\n * lack) or the client's `decodeAudio` hook are decoded on the main thread;\n * their Float32 channels are then transferred and resampled in the worker.\n */\n\nimport {\n  DEFAULT_MAX_AUDIO_BYTES,\n  normalizeAudioInput,\n} from '../../shared/audio.js';\nimport {\n  SpeechConfigurationError,\n  SpeechError,\n  SpeechProviderError,\n} from '../../shared/errors.js';\nimport type {\n  AudioInput,\n  TimestampGranularity,\n  Transcriber,\n  TranscriptionRequest,\n  TranscriptResult,\n} from '../../shared/types.js';\nimport {\n  reportSpeechUsage,\n  type SpeechUsage,\n  type SpeechUsageCallback,\n} from '../../shared/usage.js';\nimport { raceAbort } from './abort.js';\nimport {\n  decodeExternally,\n  isInProcessDecodable,\n  LOCAL_SAMPLE_RATE,\n  toPcm16k,\n} from './audio.js';\nimport { resolveLocalTranscriberOptions } from './env.js';\nimport { LocalTranscriber } from './transcriber.js';\nimport type {\n  LocalAudioDecoder,\n  LocalTranscriberOptions,\n  LocalTranscriberProgress,\n  LocalTranscriberProgressCallback,\n} from './types.js';\n\n/** Minimal `postMessage` endpoint: a `Worker`, `DedicatedWorkerGlobalScope`, or `MessagePort`. */\nexport interface LocalWorkerEndpoint {\n  postMessage(message: unknown, transfer?: Transferable[]): void;\n  addEventListener(\n    type: 'message',\n    listener: (event: MessageEvent) => void,\n  ): void;\n  removeEventListener(\n    type: 'message',\n    listener: (event: MessageEvent) => void,\n  ): void;\n}\n\n/** Audio sent to the worker. Its buffers are transferred, not copied. */\nexport type LocalWorkerAudio =\n  | {\n      /** WAV or raw PCM bytes (or any format the worker's `decodeAudio` handles). */\n      kind: 'encoded';\n      bytes: Uint8Array;\n      mimeType: string;\n      sampleRate?: number;\n      channels?: number;\n    }\n  | {\n      /** Channels decoded on the main thread; the worker downmixes and resamples. */\n      kind: 'decoded';\n      samples: Float32Array[];\n      sampleRate: number;\n    };\n\n/** Messages from the client to the worker. */\nexport type LocalWorkerRequest =\n  | {\n      kind: 'transcribe';\n      id: number;\n      audio: LocalWorkerAudio;\n      model?: string;\n      language?: string;\n      timestampGranularities?: TimestampGranularity[];\n    }\n  | { kind: 'preload'; id: number; model?: string }\n  | { kind: 'abort'; id: number };\n\n/** Messages from the worker to the client. */\nexport type LocalWorkerResponse =\n  | { kind: 'result'; id: number; result?: TranscriptResult }\n  | {\n      kind: 'error';\n      id: number;\n      error: { name: string; message: string; code?: string };\n    }\n  | { kind: 'progress'; progress: LocalTranscriberProgress };\n\nconst PCM_MIME_TYPE = `audio/pcm;rate=${LOCAL_SAMPLE_RATE};encoding=f32le`;\n\n/**\n * Runs a {@link LocalTranscriber} inside a worker and answers client\n * requests. Call it from your worker module:\n *\n * ```ts\n * // transcriber.worker.ts\n * import { serveLocalTranscriber } from '@happyvertical/speech/local';\n * serveLocalTranscriber({ model: 'onnx-community/whisper-base' });\n * ```\n *\n * Returns a function that stops listening and disposes the model.\n */\nexport function serveLocalTranscriber(\n  options: LocalTranscriberOptions = {},\n  endpoint: LocalWorkerEndpoint = globalThis as unknown as LocalWorkerEndpoint,\n): () => Promise<void> {\n  const userProgress = options.onProgress;\n  const transcriber = new LocalTranscriber({\n    ...resolveLocalTranscriberOptions(options),\n    // The client already enforced its byte limit on the encoded input.\n    maxBytes: Number.POSITIVE_INFINITY,\n    onProgress: (progress) => {\n      userProgress?.(progress);\n      endpoint.postMessage({\n        kind: 'progress',\n        progress: toCloneable(progress),\n      } satisfies LocalWorkerResponse);\n    },\n  });\n  const pending = new Map<number, AbortController>();\n\n  const reply = (message: LocalWorkerResponse) => endpoint.postMessage(message);\n  const listener = (event: MessageEvent) => {\n    const message = event.data as LocalWorkerRequest;\n    if (!message || typeof message !== 'object') {\n      return;\n    }\n\n    if (message.kind === 'abort') {\n      pending.get(message.id)?.abort();\n      return;\n    }\n\n    const controller = new AbortController();\n    pending.set(message.id, controller);\n    const work =\n      message.kind === 'preload'\n        ? transcriber.preload(message.model, controller.signal).then(() => {\n            reply({ kind: 'result', id: message.id });\n          })\n        : Promise.resolve()\n            .then(() =>\n              transcriber.transcribe({\n                audio: toAudioInput(message.audio),\n                model: message.model,\n                language: message.language,\n                timestampGranularities: message.timestampGranularities,\n                signal: controller.signal,\n              }),\n            )\n            .then((result) => {\n              reply({ kind: 'result', id: message.id, result });\n            });\n\n    work\n      .catch((error: unknown) => {\n        reply({ kind: 'error', id: message.id, error: serializeError(error) });\n      })\n      .finally(() => pending.delete(message.id));\n  };\n\n  endpoint.addEventListener('message', listener);\n  return async () => {\n    endpoint.removeEventListener('message', listener);\n    for (const controller of pending.values()) {\n      controller.abort();\n    }\n    await transcriber.dispose();\n  };\n}\n\nexport interface LocalTranscriberWorkerClientOptions {\n  /** Encoded input byte limit, checked on the main thread. Default 25 MB. */\n  maxBytes?: number;\n  /**\n   * Main-thread decoder for formats other than WAV and raw PCM. Formats with\n   * neither this hook nor `AudioContext` go to the worker's own `decodeAudio`.\n   */\n  decodeAudio?: LocalAudioDecoder;\n  /** Model download/load progress forwarded from the worker. */\n  onProgress?: LocalTranscriberProgressCallback;\n  /** Called with usage after every successful transcription. */\n  onUsage?: SpeechUsageCallback;\n}\n\n/**\n * Main-thread {@link Transcriber} that delegates inference to a worker running\n * {@link serveLocalTranscriber}.\n *\n * ```ts\n * const worker = new Worker(new URL('./transcriber.worker.ts', import.meta.url), { type: 'module' });\n * const transcriber = new LocalTranscriberWorkerClient(worker, { onProgress });\n * const { text } = await transcriber.transcribe({ audio: recordingBlob });\n * ```\n */\nexport class LocalTranscriberWorkerClient implements Transcriber {\n  readonly type = 'local' as const;\n\n  private readonly endpoint: LocalWorkerEndpoint;\n  private readonly options: LocalTranscriberWorkerClientOptions;\n  private readonly calls = new Map<\n    number,\n    {\n      resolve: (result: TranscriptResult | undefined) => void;\n      reject: (error: unknown) => void;\n    }\n  >();\n  private nextId = 1;\n  private closed = false;\n  /** Aborted by `close()` to reject calls still buffering or decoding. */\n  private readonly closing = new AbortController();\n  private readonly listener = (event: MessageEvent) =>\n    this.onMessage(event.data as LocalWorkerResponse);\n\n  constructor(\n    endpoint: LocalWorkerEndpoint,\n    options: LocalTranscriberWorkerClientOptions = {},\n  ) {\n    this.endpoint = endpoint;\n    this.options = options;\n    endpoint.addEventListener('message', this.listener);\n  }\n\n  /** Loads the model in the worker ahead of the first transcription. */\n  async preload(model?: string, signal?: AbortSignal): Promise<void> {\n    await this.call({ kind: 'preload', model }, signal);\n  }\n\n  async transcribe(request: TranscriptionRequest): Promise<TranscriptResult> {\n    this.assertOpen();\n    const { signal } = request;\n    // Buffering and main-thread decoding count as in flight: close() rejects them.\n    const { audio, bytes } = await raceAbort(\n      this.prepare(request),\n      this.closing.signal,\n    );\n\n    const result = await this.call(\n      {\n        kind: 'transcribe',\n        audio,\n        model: request.model,\n        language: request.language,\n        timestampGranularities: request.timestampGranularities,\n      },\n      signal,\n      transferList(audio),\n    );\n    if (!result) {\n      throw new SpeechProviderError(this.type, 'Worker returned no result');\n    }\n\n    const usage: SpeechUsage = {\n      ...(result.usage ?? { operation: 'transcription', provider: this.type }),\n      bytes,\n    };\n    await reportSpeechUsage(usage, this.options.onUsage, request.onUsage);\n    return { ...result, usage };\n  }\n\n  /**\n   * Stops listening, rejects in-flight and later calls, and asks the worker to\n   * abort posted work. Does not terminate the worker.\n   */\n  close(): void {\n    if (this.closed) {\n      return;\n    }\n    this.closed = true;\n    this.endpoint.removeEventListener('message', this.listener);\n    this.closing.abort(closedError());\n    for (const [id, call] of this.calls) {\n      this.endpoint.postMessage({\n        kind: 'abort',\n        id,\n      } satisfies LocalWorkerRequest);\n      call.reject(closedError());\n    }\n    this.calls.clear();\n  }\n\n  /** Buffers the input and, only when the worker cannot, decodes it here. */\n  private async prepare(\n    request: TranscriptionRequest,\n  ): Promise<{ audio: LocalWorkerAudio; bytes: number }> {\n    const { signal } = request;\n    const normalized = await normalizeAudioInput(request.audio, {\n      mimeType: request.mimeType,\n      maxBytes:\n        request.maxBytes ?? this.options.maxBytes ?? DEFAULT_MAX_AUDIO_BYTES,\n      adapter: this.type,\n      signal,\n    });\n    const bytes = new Uint8Array(await normalized.blob.arrayBuffer());\n\n    if (!isInProcessDecodable(bytes, normalized.mimeType)) {\n      // Decoders (caller hooks, decodeAudioData) cannot be cancelled: stop waiting on abort.\n      const decoded = await raceAbort(\n        decodeExternally(bytes, {\n          mimeType: normalized.mimeType,\n          decodeAudio: this.options.decodeAudio,\n          signal,\n        }),\n        signal,\n      );\n      if (decoded) {\n        const channels =\n          decoded.samples instanceof Float32Array\n            ? [decoded.samples]\n            : decoded.samples;\n        return {\n          audio: {\n            kind: 'decoded',\n            samples: channels.map(ownedCopy),\n            sampleRate: decoded.sampleRate,\n          },\n          bytes: normalized.bytes,\n        };\n      }\n    }\n\n    const wrapped =\n      typeof request.audio === 'object' &&\n      'data' in request.audio &&\n      !(request.audio instanceof Uint8Array)\n        ? request.audio\n        : undefined;\n    return {\n      audio: {\n        kind: 'encoded',\n        bytes: ownedCopy(bytes),\n        mimeType: normalized.mimeType,\n        sampleRate: wrapped?.sampleRate,\n        channels: wrapped?.channels,\n      },\n      bytes: normalized.bytes,\n    };\n  }\n\n  private assertOpen(): void {\n    if (this.closed) {\n      throw closedError();\n    }\n  }\n\n  private call(\n    message:\n      | Omit<Extract<LocalWorkerRequest, { kind: 'transcribe' }>, 'id'>\n      | Omit<Extract<LocalWorkerRequest, { kind: 'preload' }>, 'id'>,\n    signal: AbortSignal | undefined,\n    transfer: Transferable[] = [],\n  ): Promise<TranscriptResult | undefined> {\n    this.assertOpen();\n    signal?.throwIfAborted();\n    const id = this.nextId++;\n\n    return new Promise((resolve, reject) => {\n      const onAbort = () => {\n        this.calls.delete(id);\n        this.endpoint.postMessage({ kind: 'abort', id });\n        reject(signal?.reason);\n      };\n      const settle = () => signal?.removeEventListener('abort', onAbort);\n      this.calls.set(id, {\n        resolve: (result) => {\n          settle();\n          resolve(result);\n        },\n        reject: (error) => {\n          settle();\n          reject(error);\n        },\n      });\n      signal?.addEventListener('abort', onAbort, { once: true });\n      this.endpoint.postMessage({ ...message, id }, transfer);\n    });\n  }\n\n  private onMessage(message: LocalWorkerResponse): void {\n    if (!message || typeof message !== 'object') {\n      return;\n    }\n    if (message.kind === 'progress') {\n      this.options.onProgress?.(message.progress);\n      return;\n    }\n\n    const call = this.calls.get(message.id);\n    if (!call) {\n      return;\n    }\n    this.calls.delete(message.id);\n    if (message.kind === 'result') {\n      call.resolve(message.result);\n    } else {\n      call.reject(deserializeError(message.error));\n    }\n  }\n}\n\nfunction closedError(): SpeechConfigurationError {\n  return new SpeechConfigurationError('Worker client closed', 'local');\n}\n\n/** Rebuilds the transcriber input from a worker message, resampling decoded channels here. */\nfunction toAudioInput(audio: LocalWorkerAudio): AudioInput {\n  if (audio.kind === 'decoded') {\n    const pcm = toPcm16k({\n      samples: audio.samples,\n      sampleRate: audio.sampleRate,\n    });\n    return {\n      data: new Uint8Array(pcm.buffer, pcm.byteOffset, pcm.byteLength),\n      mimeType: PCM_MIME_TYPE,\n    };\n  }\n  return {\n    data: audio.bytes,\n    mimeType: audio.mimeType,\n    sampleRate: audio.sampleRate,\n    channels: audio.channels,\n  };\n}\n\n/**\n * Returns `view` when it spans a whole ordinary `ArrayBuffer` (safe to\n * transfer), otherwise a copy. Views of a larger buffer would transfer\n * unrelated bytes, and a `SharedArrayBuffer` cannot be transferred at all.\n */\nfunction ownedCopy<T extends Uint8Array | Float32Array>(view: T): T {\n  const { buffer } = view;\n  return isPlainArrayBuffer(buffer) &&\n    view.byteOffset === 0 &&\n    view.byteLength === buffer.byteLength\n    ? view\n    : (view.slice() as T);\n}\n\nfunction isPlainArrayBuffer(buffer: ArrayBufferLike): buffer is ArrayBuffer {\n  // Tag check rather than instanceof: buffers may come from another realm.\n  return Object.prototype.toString.call(buffer) === '[object ArrayBuffer]';\n}\n\n/** Distinct buffers to transfer (two channels may share one). */\nfunction transferList(audio: LocalWorkerAudio): ArrayBuffer[] {\n  const views = audio.kind === 'decoded' ? audio.samples : [audio.bytes];\n  return [...new Set(views.map((view) => view.buffer as ArrayBuffer))];\n}\n\nfunction serializeError(error: unknown): {\n  name: string;\n  message: string;\n  code?: string;\n} {\n  if (error instanceof SpeechError) {\n    return { name: error.name, message: error.message, code: error.code };\n  }\n  if (error instanceof Error) {\n    return { name: error.name, message: error.message };\n  }\n  return { name: 'Error', message: String(error) };\n}\n\nfunction deserializeError(error: {\n  name: string;\n  message: string;\n  code?: string;\n}): Error {\n  switch (error.name) {\n    case 'SpeechConfigurationError':\n      return new SpeechConfigurationError(error.message, 'local');\n    case 'SpeechProviderError':\n      return new SpeechProviderError('local', error.message);\n    case 'AbortError':\n      return new DOMException(error.message, 'AbortError');\n    default: {\n      const restored = new Error(error.message);\n      restored.name = error.name;\n      return restored;\n    }\n  }\n}\n\n/** Progress objects cross `postMessage`, so drop anything not cloneable. */\nfunction toCloneable(\n  progress: LocalTranscriberProgress,\n): LocalTranscriberProgress {\n  const clone: LocalTranscriberProgress = { status: progress.status };\n  for (const [key, value] of Object.entries(progress)) {\n    if (\n      value === null ||\n      ['string', 'number', 'boolean'].includes(typeof value)\n    ) {\n      clone[key] = value;\n    }\n  }\n  return clone;\n}\n","/**\n * @happyvertical/speech/local\n *\n * On-device speech-to-text with transformers.js (`@huggingface/transformers`,\n * an optional peer dependency loaded on first use). Importing this entry\n * registers `type: 'local'` with `getTranscriber()` and `getSpeech()`.\n *\n * @example\n * ```typescript\n * import { getTranscriber } from '@happyvertical/speech';\n * import '@happyvertical/speech/local';\n *\n * const transcriber = await getTranscriber({ type: 'local' });\n * const { text } = await transcriber.transcribe({ audio: wavBytes, mimeType: 'audio/wav' });\n * ```\n *\n * @packageDocumentation\n */\n\nimport { resolveLocalTranscriberOptions } from './adapters/local/env.js';\nimport { loadTransformers } from './adapters/local/runtime.js';\nimport { LocalTranscriber } from './adapters/local/transcriber.js';\nimport type { LocalTranscriberOptions } from './adapters/local/types.js';\nimport type { SpeechFactoryContext } from './shared/factory.js';\nimport { registerOptionalTranscriber } from './shared/registry.js';\n\nexport {\n  decodeToPcm16k,\n  LOCAL_SAMPLE_RATE,\n} from './adapters/local/audio.js';\nexport { LOCAL_TRANSCRIBER_ENV_KEYS } from './adapters/local/env.js';\nexport {\n  TRANSFORMERS_INSTALL_HINT,\n  TRANSFORMERS_PACKAGE,\n} from './adapters/local/runtime.js';\nexport {\n  DEFAULT_LOCAL_MODEL,\n  LocalTranscriber,\n} from './adapters/local/transcriber.js';\nexport type {\n  DecodedAudio,\n  LocalAudioDecoder,\n  LocalTranscriberDevice,\n  LocalTranscriberDtype,\n  LocalTranscriberOptions,\n  LocalTranscriberProgress,\n  LocalTranscriberProgressCallback,\n  LocalTranscriberSettings,\n  LocalTransformersRuntime,\n} from './adapters/local/types.js';\nexport {\n  LocalTranscriberWorkerClient,\n  type LocalTranscriberWorkerClientOptions,\n  type LocalWorkerAudio,\n  type LocalWorkerEndpoint,\n  type LocalWorkerRequest,\n  type LocalWorkerResponse,\n  serveLocalTranscriber,\n} from './adapters/local/worker.js';\n\n/**\n * Creates an on-device transcriber. 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package/dist/shared/env.d.ts

@@ -67,4 +67,9 @@ export declare function readEnv(env: SpeechEnv, ...keys: string[]): string | und
 export declare function hasAnyEnv(env: SpeechEnv, ...keys: string[]): boolean;
 export declare function parseOptionalInteger(value: string | undefined): number | undefined;
 export declare function defaultEnv(): SpeechEnv;
+/**
+ * Parses a JSON object of header names to string values. The raw value is
+ * never echoed in errors because it may carry credentials.
+ */
+export declare function parseHeadersEnv(env: SpeechEnv, keys: readonly string[]): Record<string, string> | undefined;
 //# sourceMappingURL=env.d.ts.map

package/dist/shared/env.d.ts.map

@@ -1 +1 @@
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package/dist/shared/pcm.d.ts

@@ -0,0 +1,21 @@
+import { StreamingAudioChunk, StreamingAudioFormat } from './streaming-types.js';
+/** Returns a byte view over a chunk without copying. */
+export declare function chunkToBytes(chunk: StreamingAudioChunk, adapter?: string): Uint8Array;
+export declare function encodeBase64(bytes: Uint8Array): string;
+/** Bytes per second of audio in `format`. */
+export declare function bytesPerSecond(format: StreamingAudioFormat): number;
+/** Audio duration in seconds for `bytes` of `format`, rounded to milliseconds. */
+export declare function audioSecondsForBytes(bytes: number, format: StreamingAudioFormat): number;
+export interface RawAudio {
+    bytes: Uint8Array;
+    /** Format fields the input declared (WAV header, MIME parameters, or `AudioInput`). */
+    format: Partial<StreamingAudioFormat>;
+}
+/**
+ * Interprets a finished recording as raw audio for a streaming session.
+ * Accepts PCM/G.711 WAV files, `audio/pcm`/`audio/L16` (with optional `rate`
+ * and `channels` parameters), `audio/pcmu`/`audio/pcma`, and untyped bytes.
+ * Compressed containers are rejected with a `SpeechConfigurationError`.
+ */
+export declare function unwrapRawAudio(bytes: Uint8Array, mimeType: string, adapter: string): RawAudio;
+//# sourceMappingURL=pcm.d.ts.map

package/dist/shared/pcm.d.ts.map

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package/dist/shared/realtime-auth.d.ts

@@ -0,0 +1,34 @@
+import { RealtimeConnection } from './realtime-session.js';
+import { StreamingClientSecret } from './streaming-types.js';
+/** Subprotocol prefix that carries a client secret (OpenAI's convention). */
+export declare const CLIENT_SECRET_SUBPROTOCOL_PREFIX = "openai-insecure-api-key.";
+/** Throws when an adapter is given a long-lived `apiKey` in a browser. */
+export declare function assertNoBrowserApiKey(adapter: string, apiKey: string | undefined): void;
+/** Resolves a static or per-session client secret and validates its shape. */
+export declare function resolveClientSecret(adapter: string, secret: StreamingClientSecret | undefined): Promise<string | undefined>;
+export interface RealtimeConnectorOptions {
+    adapter: string;
+    url: string;
+    apiKey?: string;
+    headers: Record<string, string>;
+    clientSecret?: StreamingClientSecret;
+    /** Subprotocols to offer, given the resolved client secret (if any). */
+    protocols: (secret: string | undefined) => string[];
+}
+/**
+ * Builds the per-session `connect` callback: refuses handshake headers in a
+ * browser, resolves the client secret, and sets `Authorization` from
+ * `apiKey` unless a header already supplies it.
+ */
+export declare function realtimeConnector(options: RealtimeConnectorOptions): () => Promise<RealtimeConnection>;
+/**
+ * Values of caller-supplied headers to redact from provider error text.
+ * Credential-named headers are redacted whatever their length; for
+ * `Authorization`, the credential after the scheme is included too. Other
+ * headers may also carry tenant secrets, so their values are redacted when
+ * they are at least 8 characters long, which keeps short values such as
+ * `1` from mangling messages.
+ */
+export declare function headerSecrets(headers: HeadersInit | undefined): string[];
+export declare function mergeHeaderRecords(...sets: Array<HeadersInit | undefined>): Record<string, string>;
+//# sourceMappingURL=realtime-auth.d.ts.map

package/dist/shared/realtime-auth.d.ts.map

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package/dist/shared/realtime-session.d.ts

@@ -0,0 +1,240 @@
+import { StreamingAudioChunk, StreamingAudioFormat, StreamingSession, StreamingSessionEventName, StreamingSessionListener, StreamingSessionState, StreamingTurnDetection } from './streaming-types.js';
+import { TurnLimit } from './turn-limit.js';
+import { TranscriptResult } from './types.js';
+import { SpeechUsageCallback } from './usage.js';
+import { SpeechWebSocketFactory } from './websocket.js';
+export declare const DEFAULT_STREAMING_TIMEOUT_MS = 30000;
+export declare const DEFAULT_STREAMING_CONNECT_TIMEOUT_MS = 10000;
+export declare const DEFAULT_STREAMING_HIGH_WATER_MARK: number;
+export declare const DEFAULT_STREAMING_MAX_BUFFERED_BYTES: number;
+/** Normalised events a protocol parser returns for one server message. */
+export type RealtimeProtocolEvent = {
+    type: 'session';
+    sessionId?: string;
+}
+/** The provider committed a turn (from a client commit or server VAD). */
+ | {
+    type: 'committed';
+    itemId?: string;
+}
+/** The provider rejected a commit because its buffer was empty. Not fatal. */
+ | {
+    type: 'commit_empty';
+} | {
+    type: 'partial';
+    itemId?: string;
+    delta: string;
+} | {
+    type: 'final';
+    itemId?: string;
+    text: string;
+    providerUsage?: Record<string, unknown>;
+    raw?: unknown;
+    /**
+     * `false` when the provider reports that the turn consumed no audio
+     * (vLLM: `usage.prompt_tokens <= 1`). A turn that was sent audio but
+     * consumed none means the provider's turn state is out of sync, and the
+     * session fails instead of silently dropping that audio.
+     */
+    audioConsumed?: boolean;
+} | {
+    type: 'speech_started';
+    itemId?: string;
+    audioMs?: number;
+} | {
+    type: 'speech_stopped';
+    itemId?: string;
+    audioMs?: number;
+}
+/** A fatal provider error. */
+ | {
+    type: 'error';
+    message: string;
+    code?: string;
+    raw?: unknown;
+};
+/** Resolved per-session settings handed to the protocol. */
+export interface RealtimeSessionConfig {
+    model: string;
+    format: StreamingAudioFormat;
+    turnDetection: StreamingTurnDetection;
+    language?: string;
+    prompt?: string;
+}
+/** Wire format of one realtime provider. */
+export interface RealtimeProtocol {
+    /** Adapter name used for errors and usage, e.g. `openai-realtime`. */
+    readonly provider: string;
+    /**
+     * How a client commit is acknowledged: by a `committed` event (OpenAI), or
+     * only by the turn's `final` event (servers without a commit ack).
+     */
+    readonly commitAck: 'committed' | 'final';
+    /**
+     * Whether `end()` commits only when uncommitted audio remains
+     * (`if-audio`), or always sends a final commit (`always`, for servers that
+     * need an explicit end-of-stream marker).
+     */
+    readonly endCommit: 'if-audio' | 'always';
+    /**
+     * Default per-turn audio cap in seconds, for providers whose model context
+     * covers one turn (vLLM resets it after each final commit). Adapters pass
+     * it to `resolveTurnLimit()`; a session with a turn limit rolls over to a
+     * new turn before the cap, or rejects writes past it. Unset: no default.
+     */
+    readonly maxTurnSeconds?: number;
+    /**
+     * Shortest audio, in seconds, the provider accepts in one commit (OpenAI:
+     * 100 ms; shorter commits are rejected as empty). A session with a turn
+     * limit pads a shorter, non-empty turn with silence before committing it,
+     * so the tail left after a rollover is transcribed instead of dropped.
+     */
+    readonly minCommitSeconds?: number;
+    /** Messages sent once the socket opens (session configuration). */
+    sessionMessages(config: RealtimeSessionConfig): unknown[];
+    /** One audio chunk, already base64-encoded. */
+    appendMessage(base64Audio: string): unknown;
+    /** Ends the current turn. `final` is true for the commit sent by `end()`. */
+    commitMessage(options: {
+        final: boolean;
+    }): unknown;
+    /**
+     * Opens a turn before its first audio, for servers that only start
+     * generating after an explicit start signal (vLLM's non-final
+     * `input_audio_buffer.commit`). When set, the session sends it before the
+     * first append of every turn, and holds audio written after a commit until
+     * every outstanding turn has been acknowledged, so a server that resets its
+     * buffer between turns never drops audio.
+     */
+    turnStartMessage?(): unknown;
+    /** Maps one parsed JSON server message to zero or more events. */
+    parse(message: unknown): RealtimeProtocolEvent[];
+}
+/** Where and how to open the socket; resolved per session. */
+export interface RealtimeConnection {
+    url: string;
+    protocols: string[];
+    headers: Record<string, string>;
+    /** Credentials to redact from surfaced provider text. */
+    secrets: string[];
+}
+export interface RealtimeSessionInit {
+    protocol: RealtimeProtocol;
+    config: RealtimeSessionConfig;
+    /** Resolves the connection (may mint a short-lived token). */
+    connect: () => Promise<RealtimeConnection>;
+    createWebSocket: SpeechWebSocketFactory;
+    timeoutMs: number;
+    connectTimeoutMs: number;
+    highWaterMark: number;
+    maxBufferedBytes: number;
+    signal?: AbortSignal;
+    /** Called in order with the aggregated usage when `end()` succeeds. */
+    onUsage: Array<SpeechUsageCallback | undefined>;
+    /** Per-turn audio limit; unset means turns are unbounded. */
+    turnLimit?: TurnLimit;
+}
+export declare class RealtimeTranscriptionSession implements StreamingSession {
+    state: StreamingSessionState;
+    sessionId?: string;
+    readonly ready: Promise<void>;
+    private readonly init;
+    private readonly provider;
+    private resolveReady;
+    private rejectReady;
+    private socket?;
+    private opened;
+    private secrets;
+    private queue;
+    private queued;
+    private pumping;
+    private totalBytes;
+    private uncommittedBytes;
+    private outstandingCommits;
+    /** Per outstanding commit, in order: whether it carried audio. */
+    private readonly commitHadAudio;
+    private turnOpen;
+    /** `Date.now()` of the last server message; bounds inactivity waits. */
+    private lastServerMessageAt;
+    private readonly itemOrder;
+    private readonly pendingItems;
+    private readonly vadStoppedItems;
+    private readonly partials;
+    private readonly finals;
+    private readonly listeners;
+    private error?;
+    private endPromise?;
+    private endWaiter?;
+    private connectTimer?;
+    private endTimer?;
+    private closeEmitted;
+    /** Measures the current turn in write order when a turn limit applies. */
+    private readonly splitter?;
+    private readonly onAbort;
+    constructor(init: RealtimeSessionInit);
+    get queuedBytes(): number;
+    write(chunk: StreamingAudioChunk): Promise<void>;
+    commit(): void;
+    end(): Promise<TranscriptResult>;
+    /**
+     * (Re)starts the end timeout. It measures provider inactivity: every server
+     * message restarts it, so a long recording that is still being transcribed
+     * is not cut off, while a silent provider still fails after `timeoutMs`.
+     */
+    private armEndTimer;
+    abort(reason?: unknown): void;
+    on<E extends StreamingSessionEventName>(event: E, listener: StreamingSessionListener<E>): () => void;
+    off<E extends StreamingSessionEventName>(event: E, listener: StreamingSessionListener<E>): void;
+    private enqueueAudio;
+    private connect;
+    private onOpen;
+    private onMessage;
+    private handle;
+    /** Acknowledges the oldest outstanding commit; returns whether it carried audio. */
+    private ackCommit;
+    private pump;
+    private sendCommit;
+    /**
+     * In a session with a turn limit, tops up a non-empty turn shorter than the
+     * protocol's `minCommitSeconds` with silence (not counted as streamed
+     * audio). Rollover can leave such a tail, which the provider would
+     * otherwise reject as empty and drop. Returns false when the send failed.
+     */
+    private padShortTurn;
+    /**
+     * For protocols with an explicit turn start: before the next audio, waits
+     * for every outstanding turn to be acknowledged, then opens a new turn. Like
+     * the end timeout, the wait is bounded by provider inactivity: it fails only
+     * after `timeoutMs` without any server message. Returns false when the
+     * session has failed.
+     */
+    private openTurnIfNeeded;
+    /**
+     * Waits until the socket's send buffer is at or below `highWaterMark`. A
+     * socket that stays congested for `timeoutMs` fails the session, so a
+     * stalled connection can never leave `write()` pending forever.
+     */
+    private waitForDrain;
+    private send;
+    private checkEnded;
+    private finish;
+    private rank;
+    private onClose;
+    private fail;
+    private cleanup;
+    private closeSocket;
+    private emit;
+    private isTerminal;
+    private redact;
+}
+/**
+ * Throws when a turn cap is below the provider's minimum commit length:
+ * padding a short turn up to that minimum would then exceed the cap.
+ */
+export declare function assertTurnLimitFitsProtocol(protocol: Pick<RealtimeProtocol, 'provider' | 'minCommitSeconds'>, limit: Pick<TurnLimit, 'maxTurnSeconds'> | undefined): void;
+/**
+ * Sums numeric fields of per-turn provider usage blocks (recursively), keeping
+ * the first value of non-numeric fields such as `type`.
+ */
+export declare function sumUsage(blocks: Array<Record<string, unknown> | undefined>): Record<string, unknown> | undefined;
+//# sourceMappingURL=realtime-session.d.ts.map

package/dist/shared/realtime-session.d.ts.map

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package/dist/shared/registry.d.ts

@@ -0,0 +1,9 @@
+import { SpeechFactoryContext } from './factory.js';
+import { GetTranscriberOptions, Transcriber, TranscriberType } from './types.js';
+export type OptionalTranscriberFactory = (options: GetTranscriberOptions, context: SpeechFactoryContext) => Transcriber | Promise<Transcriber>;
+export declare function registerOptionalTranscriber(type: TranscriberType, factory: OptionalTranscriberFactory): void;
+export declare function getOptionalTranscriberFactory(type: string): OptionalTranscriberFactory | undefined;
+/** Subpath that registers `type`, when it is an opt-in adapter. */
+export declare function optionalTranscriberEntry(type: string): string | undefined;
+export declare function registeredOptionalTranscribers(): TranscriberType[];
+//# sourceMappingURL=registry.d.ts.map

package/dist/shared/registry.d.ts.map

@@ -0,0 +1 @@
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package/dist/shared/streaming-client-secret.d.ts

@@ -0,0 +1,90 @@
+import { SpeechEnv } from './env.js';
+import { OpenAIRealtimeTranscriberOptions, StreamingAudioFormat, StreamingTranscriberType, StreamingTurnDetection } from './streaming-types.js';
+import { SpeechFetch } from './types.js';
+/** Default client-secret lifetime: long enough to connect, short if leaked. */
+export declare const DEFAULT_CLIENT_SECRET_TTL_SECONDS = 60;
+/** OpenAI's accepted `expires_after.seconds` range. */
+export declare const CLIENT_SECRET_TTL_RANGE: {
+    readonly min: 10;
+    readonly max: 7200;
+};
+export interface CreateStreamingClientSecretOptions {
+    /** Streaming adapter. Default `HAVE_SPEECH_STREAMING_TYPE`, then `openai-realtime`. */
+    type?: StreamingTranscriberType;
+    /** Long-lived server key. Default `HAVE_SPEECH_STREAMING_API_KEY`. */
+    apiKey?: string;
+    /**
+     * API base. Accepts the same values as the streaming adapter
+     * (`https://api.openai.com`, `…/v1`, or `wss://…/v1/realtime`). Default
+     * `HAVE_SPEECH_STREAMING_BASE_URL`, then `https://api.openai.com/v1`.
+     */
+    baseUrl?: string;
+    /** Secret lifetime in seconds, 10–7200. Default 60. */
+    ttlSeconds?: number;
+    /**
+     * Tenant the secret is minted for. Echoed in the result for your usage
+     * ledger; OpenAI's transcription client secrets carry no attribution field.
+     */
+    tenantId?: string;
+    /** Your session id (e.g. the dictation session), echoed in the result. */
+    sessionId?: string;
+    /** Extra attribution echoed in the result. Never sent to the provider. */
+    metadata?: Record<string, unknown>;
+    /** Transcription model the secret is bound to. Default `HAVE_SPEECH_STREAMING_MODEL`, then `gpt-4o-transcribe`. */
+    model?: string;
+    language?: string;
+    prompt?: string;
+    /** Default `HAVE_SPEECH_STREAMING_TURN_DETECTION`, then `server_vad`. */
+    turnDetection?: StreamingTurnDetection;
+    /** Default PCM16 24 kHz mono. */
+    format?: Partial<StreamingAudioFormat>;
+    noiseReduction?: OpenAIRealtimeTranscriberOptions['noiseReduction'];
+    include?: string[];
+    /**
+     * Extra `transcription` fields. The typed `model`, `language`, and `prompt`
+     * take precedence, so the secret is bound to the model the result reports.
+     */
+    transcriptionOptions?: Record<string, unknown>;
+    /**
+     * Headers for the mint request, e.g. a per-tenant gateway virtual key
+     * (`x-bf-vk`) for provider-side attribution. Merged over
+     * `HAVE_SPEECH_STREAMING_HEADERS`.
+     */
+    headers?: HeadersInit;
+    fetch?: SpeechFetch;
+    signal?: AbortSignal;
+    /** Request timeout. Default `HAVE_SPEECH_STREAMING_TIMEOUT[_MS]`, then 10000. */
+    timeoutMs?: number;
+}
+/** A minted browser credential plus the attribution it was minted for. */
+export interface StreamingClientSecretResult {
+    type: StreamingTranscriberType;
+    /** The short-lived secret. Send it to the browser as `clientSecret`. */
+    value: string;
+    /** Expiry as epoch milliseconds. */
+    expiresAt: number;
+    ttlSeconds: number;
+    model: string;
+    tenantId?: string;
+    sessionId?: string;
+    metadata?: Record<string, unknown>;
+    /** The provider's echoed session config. */
+    session?: Record<string, unknown>;
+}
+export interface StreamingClientSecretContext {
+    env?: SpeechEnv;
+}
+/**
+ * Mints a short-lived, session-scoped client secret for browser streaming.
+ * Server-side only: throws in a browser runtime, and for adapters without an
+ * ephemeral-token mechanism (`voxtral-realtime`).
+ */
+export declare function createStreamingClientSecret(options?: CreateStreamingClientSecretOptions, context?: StreamingClientSecretContext): Promise<StreamingClientSecretResult>;
+/**
+ * Resolves the client-secret endpoint from any accepted realtime base:
+ * `ws(s)` becomes `http(s)`, a trailing `/realtime` and query are dropped,
+ * then `/v1/realtime/client_secrets` (or `/realtime/client_secrets` under a
+ * versioned root) is appended.
+ */
+export declare function resolveClientSecretUrl(baseUrl: string): string;
+//# sourceMappingURL=streaming-client-secret.d.ts.map