@livekit/agents 1.0.36 → 1.0.38

package/dist/inference/stt.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/inference/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame } from '@livekit/rtc-node';\nimport type { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport {\n  STT as BaseSTT,\n  SpeechStream as BaseSpeechStream,\n  type SpeechData,\n  type SpeechEvent,\n  SpeechEventType,\n} from '../stt/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { type AudioBuffer, Event, Task, cancelAndWait, shortuuid, waitForAbort } from '../utils.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type DeepgramModels =\n  | 'deepgram'\n  | 'deepgram/nova-3'\n  | 'deepgram/nova-3-general'\n  | 'deepgram/nova-3-medical'\n  | 'deepgram/nova-2-conversationalai'\n  | 'deepgram/nova-2'\n  | 'deepgram/nova-2-general'\n  | 'deepgram/nova-2-medical'\n  | 'deepgram/nova-2-phonecall';\n\nexport type CartesiaModels = 'cartesia' | 'cartesia/ink-whisper';\n\nexport type AssemblyaiModels = 'assemblyai' | 'assemblyai/universal-streaming';\n\nexport interface CartesiaOptions {\n  min_volume?: number; // default: not specified\n  max_silence_duration_secs?: number; // default: not specified\n}\n\nexport interface DeepgramOptions {\n  filler_words?: boolean; // default: true\n  interim_results?: boolean; // default: true\n  endpointing?: number; // default: 25 (ms)\n  punctuate?: boolean; // default: false\n  smart_format?: boolean;\n  keywords?: Array<[string, number]>;\n  keyterms?: string[];\n  profanity_filter?: boolean;\n  numerals?: boolean;\n  mip_opt_out?: boolean;\n}\n\nexport interface AssemblyAIOptions {\n  format_turns?: boolean; // default: false\n  end_of_turn_confidence_threshold?: number; // default: 0.01\n  min_end_of_turn_silence_when_confident?: number; // default: 0\n  max_turn_silence?: number; // default: not specified\n  keyterms_prompt?: string[]; // default: not specified\n}\n\nexport type STTLanguages =\n  | 'multi'\n  | 'en'\n  | 'de'\n  | 'es'\n  | 'fr'\n  | 'ja'\n  | 'pt'\n  | 'zh'\n  | 'hi'\n  | AnyString;\n\ntype _STTModels = DeepgramModels | CartesiaModels | AssemblyaiModels;\n\nexport type STTModels = _STTModels | 'auto' | AnyString;\n\nexport type ModelWithLanguage = `${_STTModels}:${STTLanguages}` | STTModels;\n\nexport type STTOptions<TModel extends STTModels> = TModel extends DeepgramModels\n  ? DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? AssemblyAIOptions\n      : Record<string, unknown>;\n\nexport type STTEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: STTEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'wss://agent-gateway.livekit.cloud/v1';\nconst DEFAULT_CANCEL_TIMEOUT = 5000;\n\nexport interface InferenceSTTOptions<TModel extends STTModels> {\n  model?: TModel;\n  language?: STTLanguages;\n  encoding: STTEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: STTOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference STT\n */\nexport class STT<TModel extends STTModels> extends BaseSTT {\n  private opts: InferenceSTTOptions<TModel>;\n  private streams: Set<SpeechStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts?: {\n    model?: TModel;\n    language?: STTLanguages;\n    baseURL?: string;\n    encoding?: STTEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label(): string {\n    return 'inference.STT';\n  }\n\n  static fromModelString(modelString: string): STT<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, language] = modelString.split(':') as [AnyString, STTLanguages];\n      return new STT({ model, language });\n    }\n    return new STT({ model: modelString });\n  }\n\n  protected async _recognize(_: AudioBuffer): Promise<SpeechEvent> {\n    throw new Error('LiveKit STT does not support batch recognition, use stream() instead');\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  stream(options?: {\n    language?: STTLanguages | string;\n    connOptions?: APIConnectOptions;\n  }): SpeechStream<TModel> {\n    const { language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    const params = {\n      settings: {\n        sample_rate: String(this.opts.sampleRate),\n        encoding: this.opts.encoding,\n        extra: this.opts.modelOptions,\n      },\n    } as Record<string, unknown>;\n\n    if (this.opts.model && this.opts.model !== 'auto') {\n      params.model = this.opts.model;\n    }\n\n    if (this.opts.language) {\n      (params.settings as Record<string, unknown>).language = this.opts.language;\n    }\n\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/stt`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const socket = await connectWs(url, headers, timeout);\n    const msg = { ...params, type: 'session.create' };\n    socket.send(JSON.stringify(msg));\n\n    return socket;\n  }\n}\n\nexport class SpeechStream<TModel extends STTModels> extends BaseSpeechStream {\n  private opts: InferenceSTTOptions<TModel>;\n  private requestId = shortuuid('stt_request_');\n  private speaking = false;\n  private speechDuration = 0;\n  private reconnectEvent = new Event();\n  private stt: STT<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(\n    sttImpl: STT<TModel>,\n    opts: InferenceSTTOptions<TModel>,\n    connOptions: APIConnectOptions,\n  ) {\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.opts = opts;\n    this.stt = sttImpl;\n    this.connOptions = connOptions;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n    this.reconnectEvent.set();\n  }\n\n  protected async run(): Promise<void> {\n    while (true) {\n      // Create fresh resources for each connection attempt\n      let ws: WebSocket | null = null;\n      let closing = false;\n      let finalReceived = false;\n\n      type SttServerEvent = Record<string, any>;\n      const eventChannel = createStreamChannel<SttServerEvent>();\n\n      const resourceCleanup = () => {\n        if (closing) return;\n        closing = true;\n        eventChannel.close();\n        ws?.removeAllListeners();\n        ws?.close();\n      };\n\n      const createWsListener = async (ws: WebSocket, signal: AbortSignal) => {\n        return new Promise<void>((resolve, reject) => {\n          const onAbort = () => {\n            resourceCleanup();\n            reject(new Error('WebSocket connection aborted'));\n          };\n\n          signal.addEventListener('abort', onAbort, { once: true });\n\n          ws.on('message', (data) => {\n            const json = JSON.parse(data.toString()) as SttServerEvent;\n            eventChannel.write(json);\n          });\n\n          ws.on('error', (e) => {\n            this.#logger.error({ error: e }, 'WebSocket error');\n            resourceCleanup();\n            reject(e);\n          });\n\n          ws.on('close', (code: number) => {\n            resourceCleanup();\n\n            if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n            if (finalReceived) return resolve();\n\n            reject(\n              new APIStatusError({\n                message: 'LiveKit STT connection closed unexpectedly',\n                options: { statusCode: code },\n              }),\n            );\n          });\n        });\n      };\n\n      const send = async (socket: WebSocket, signal: AbortSignal) => {\n        const audioStream = new AudioByteStream(\n          this.opts.sampleRate,\n          1,\n          Math.floor(this.opts.sampleRate / 20), // 50ms\n        );\n\n        // Create abort promise once to avoid memory leak\n        const abortPromise = new Promise<never>((_, reject) => {\n          if (signal.aborted) {\n            return reject(new Error('Send aborted'));\n          }\n          const onAbort = () => reject(new Error('Send aborted'));\n          signal.addEventListener('abort', onAbort, { once: true });\n        });\n\n        // Manual iteration to support cancellation\n        const iterator = this.input[Symbol.asyncIterator]();\n        try {\n          while (true) {\n            const result = await Promise.race([iterator.next(), abortPromise]);\n\n            if (result.done) break;\n            const ev = result.value;\n\n            let frames: AudioFrame[];\n            if (ev === SpeechStream.FLUSH_SENTINEL) {\n              frames = audioStream.flush();\n            } else {\n              const frame = ev as AudioFrame;\n              frames = audioStream.write(new Int16Array(frame.data).buffer);\n            }\n\n            for (const frame of frames) {\n              this.speechDuration += frame.samplesPerChannel / frame.sampleRate;\n              const base64 = Buffer.from(frame.data.buffer).toString('base64');\n              const msg = { type: 'input_audio', audio: base64 };\n              socket.send(JSON.stringify(msg));\n            }\n          }\n\n          closing = true;\n          socket.send(JSON.stringify({ type: 'session.finalize' }));\n        } catch (e) {\n          if ((e as Error).message === 'Send aborted') {\n            // Expected abort, don't log\n            return;\n          }\n          throw e;\n        }\n      };\n\n      const recv = async (signal: AbortSignal) => {\n        const serverEventStream = eventChannel.stream();\n        const reader = serverEventStream.getReader();\n\n        try {\n          while (!this.closed && !signal.aborted) {\n            const result = await reader.read();\n            if (signal.aborted) return;\n            if (result.done) return;\n\n            const json = result.value;\n            const type = json.type as string | undefined;\n\n            switch (type) {\n              case 'session.created':\n              case 'session.finalized':\n                break;\n              case 'session.closed':\n                finalReceived = true;\n                resourceCleanup();\n                break;\n              case 'interim_transcript':\n                this.processTranscript(json, false);\n                break;\n              case 'final_transcript':\n                this.processTranscript(json, true);\n                break;\n              case 'error':\n                this.#logger.error({ error: json }, 'Received error from LiveKit STT');\n                resourceCleanup();\n                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);\n              default:\n                this.#logger.warn(\n                  { message: json },\n                  'Received unexpected message from LiveKit STT',\n                );\n                break;\n            }\n          }\n        } finally {\n          reader.releaseLock();\n          try {\n            await serverEventStream.cancel();\n          } catch (e) {\n            this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n          }\n        }\n      };\n\n      try {\n        ws = await this.stt.connectWs(this.connOptions.timeoutMs);\n\n        const controller = this.abortController; // Use base class abortController for proper cancellation\n        const sendTask = Task.from(({ signal }) => send(ws!, signal), controller);\n        const wsListenerTask = Task.from(({ signal }) => createWsListener(ws!, signal), controller);\n        const recvTask = Task.from(({ signal }) => recv(signal), controller);\n        const waitReconnectTask = Task.from(\n          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),\n          controller,\n        );\n\n        try {\n          await Promise.race([\n            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),\n            waitReconnectTask.result,\n          ]);\n\n          // If reconnect didn't trigger, tasks finished - exit loop\n          if (!waitReconnectTask.done) break;\n\n          // Reconnect triggered - clear event and continue loop\n          this.reconnectEvent.clear();\n        } finally {\n          // Cancel all tasks to ensure cleanup\n          await cancelAndWait(\n            [sendTask, wsListenerTask, recvTask, waitReconnectTask],\n            DEFAULT_CANCEL_TIMEOUT,\n          );\n          resourceCleanup();\n        }\n      } finally {\n        // Ensure cleanup even if connectWs throws\n        resourceCleanup();\n      }\n    }\n  }\n\n  private processTranscript(data: Record<string, any>, isFinal: boolean) {\n    // Check if queue is closed to avoid race condition during disconnect\n    if (this.queue.closed) return;\n\n    const requestId = data.request_id ?? this.requestId;\n    const text = data.transcript ?? '';\n    const language = data.language ?? this.opts.language ?? 'en';\n\n    if (!text && !isFinal) return;\n\n    try {\n      // We'll have a more accurate way of detecting when speech started when we have VAD\n      if (!this.speaking) {\n        this.speaking = true;\n        this.queue.put({ type: SpeechEventType.START_OF_SPEECH });\n      }\n\n      const speechData: SpeechData = {\n        language,\n        startTime: data.start ?? 0,\n        endTime: data.duration ?? 0,\n        confidence: data.confidence ?? 1.0,\n        text,\n      };\n\n      if (isFinal) {\n        if (this.speechDuration > 0) {\n          this.queue.put({\n            type: SpeechEventType.RECOGNITION_USAGE,\n            requestId,\n            recognitionUsage: { audioDuration: this.speechDuration },\n          });\n          this.speechDuration = 0;\n        }\n\n        this.queue.put({\n          type: SpeechEventType.FINAL_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n\n        if (this.speaking) {\n          this.speaking = false;\n          this.queue.put({ type: SpeechEventType.END_OF_SPEECH });\n        }\n      } else {\n        this.queue.put({\n          type: SpeechEventType.INTERIM_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n      }\n    } catch (e) {\n      if (e instanceof Error && e.message.includes('Queue is closed')) {\n        // Expected behavior on disconnect, log as warning\n        this.#logger.warn(\n          { err: e },\n          'Queue closed during transcript processing (expected during disconnect)',\n        );\n      } else {\n        this.#logger.error({ err: e }, 'Error putting transcript to queue');\n      }\n    }\n  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DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? 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modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true, alignedTranscript: 'word' });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label(): string {\n    return 'inference.STT';\n  }\n\n  static fromModelString(modelString: string): STT<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, language] = modelString.split(':') as [AnyString, STTLanguages];\n      return new STT({ model, language });\n    }\n    return new STT({ model: modelString });\n  }\n\n  protected async _recognize(_: AudioBuffer): Promise<SpeechEvent> {\n    throw new Error('LiveKit STT does not support batch recognition, use stream() instead');\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  stream(options?: {\n    language?: STTLanguages | string;\n    connOptions?: APIConnectOptions;\n  }): SpeechStream<TModel> {\n    const { language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    const params = {\n      settings: {\n        sample_rate: String(this.opts.sampleRate),\n        encoding: this.opts.encoding,\n        extra: this.opts.modelOptions,\n      },\n    } as Record<string, unknown>;\n\n    if (this.opts.model && this.opts.model !== 'auto') {\n      params.model = this.opts.model;\n    }\n\n    if (this.opts.language) {\n      (params.settings as Record<string, unknown>).language = this.opts.language;\n    }\n\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/stt`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const socket = await connectWs(url, headers, timeout);\n    const msg = { ...params, type: 'session.create' };\n    socket.send(JSON.stringify(msg));\n\n    return socket;\n  }\n}\n\nexport class SpeechStream<TModel extends STTModels> extends BaseSpeechStream {\n  private opts: InferenceSTTOptions<TModel>;\n  private requestId = shortuuid('stt_request_');\n  private speaking = false;\n  private speechDuration = 0;\n  private reconnectEvent = new Event();\n  private stt: STT<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(\n    sttImpl: STT<TModel>,\n    opts: InferenceSTTOptions<TModel>,\n    connOptions: APIConnectOptions,\n  ) {\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.opts = opts;\n    this.stt = sttImpl;\n    this.connOptions = connOptions;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n    this.reconnectEvent.set();\n  }\n\n  protected async run(): Promise<void> {\n    while (true) {\n      // Create fresh resources for each connection attempt\n      let ws: WebSocket | null = null;\n      let closing = false;\n      let finalReceived = false;\n\n      const eventChannel = createStreamChannel<SttServerEvent>();\n\n      const resourceCleanup = () => {\n        if (closing) return;\n        closing = true;\n        eventChannel.close();\n        ws?.removeAllListeners();\n        ws?.close();\n      };\n\n      const createWsListener = async (ws: WebSocket, signal: AbortSignal) => {\n        return new Promise<void>((resolve, reject) => {\n          const onAbort = () => {\n            resourceCleanup();\n            reject(new Error('WebSocket connection aborted'));\n          };\n\n          signal.addEventListener('abort', onAbort, { once: true });\n\n          ws.on('message', (data) => {\n            const json = JSON.parse(data.toString()) as SttServerEvent;\n            eventChannel.write(json);\n          });\n\n          ws.on('error', (e) => {\n            this.#logger.error({ error: e }, 'WebSocket error');\n            resourceCleanup();\n            reject(e);\n          });\n\n          ws.on('close', (code: number) => {\n            resourceCleanup();\n\n            if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n            if (finalReceived) return resolve();\n\n            reject(\n              new APIStatusError({\n                message: 'LiveKit STT connection closed unexpectedly',\n                options: { statusCode: code },\n              }),\n            );\n          });\n        });\n      };\n\n      const send = async (socket: WebSocket, signal: AbortSignal) => {\n        const audioStream = new AudioByteStream(\n          this.opts.sampleRate,\n          1,\n          Math.floor(this.opts.sampleRate / 20), // 50ms\n        );\n\n        // Create abort promise once to avoid memory leak\n        const abortPromise = new Promise<never>((_, reject) => {\n          if (signal.aborted) {\n            return reject(new Error('Send aborted'));\n          }\n          const onAbort = () => reject(new Error('Send aborted'));\n          signal.addEventListener('abort', onAbort, { once: true });\n        });\n\n        // Manual iteration to support cancellation\n        const iterator = this.input[Symbol.asyncIterator]();\n        try {\n          while (true) {\n            const result = await Promise.race([iterator.next(), abortPromise]);\n\n            if (result.done) break;\n            const ev = result.value;\n\n            let frames: AudioFrame[];\n            if (ev === SpeechStream.FLUSH_SENTINEL) {\n              frames = audioStream.flush();\n            } else {\n              const frame = ev as AudioFrame;\n              frames = audioStream.write(new Int16Array(frame.data).buffer);\n            }\n\n            for (const frame of frames) {\n              this.speechDuration += frame.samplesPerChannel / frame.sampleRate;\n              const base64 = Buffer.from(frame.data.buffer).toString('base64');\n              const msg = { type: 'input_audio', audio: base64 };\n              socket.send(JSON.stringify(msg));\n            }\n          }\n\n          closing = true;\n          socket.send(JSON.stringify({ type: 'session.finalize' }));\n        } catch (e) {\n          if ((e as Error).message === 'Send aborted') {\n            // Expected abort, don't log\n            return;\n          }\n          throw e;\n        }\n      };\n\n      const recv = async (signal: AbortSignal) => {\n        const serverEventStream = eventChannel.stream();\n        const reader = serverEventStream.getReader();\n\n        try {\n          while (!this.closed && !signal.aborted) {\n            const result = await reader.read();\n            if (signal.aborted) return;\n            if (result.done) return;\n\n            // Parse and validate with Zod schema\n            const parseResult = await sttServerEventSchema.safeParseAsync(result.value);\n            if (!parseResult.success) {\n              this.#logger.warn(\n                { error: parseResult.error, rawData: result.value },\n                'Failed to parse STT server event',\n              );\n              continue;\n            }\n\n            const event: SttServerEvent = parseResult.data;\n\n            switch (event.type) {\n              case 'session.created':\n              case 'session.finalized':\n                break;\n              case 'session.closed':\n                finalReceived = true;\n                resourceCleanup();\n                break;\n              case 'interim_transcript':\n                this.processTranscript(event, false);\n                break;\n              case 'final_transcript':\n                this.processTranscript(event, true);\n                break;\n              case 'error':\n                this.#logger.error({ error: event }, 'Received error from LiveKit STT');\n                resourceCleanup();\n                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(event)}`);\n            }\n          }\n        } finally {\n          reader.releaseLock();\n          try {\n            await serverEventStream.cancel();\n          } catch (e) {\n            this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n          }\n        }\n      };\n\n      try {\n        ws = await this.stt.connectWs(this.connOptions.timeoutMs);\n\n        const controller = this.abortController; // Use base class abortController for proper cancellation\n        const sendTask = Task.from(({ signal }) => send(ws!, signal), controller);\n        const wsListenerTask = Task.from(({ signal }) => createWsListener(ws!, signal), controller);\n        const recvTask = Task.from(({ signal }) => recv(signal), controller);\n        const waitReconnectTask = Task.from(\n          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),\n          controller,\n        );\n\n        try {\n          await Promise.race([\n            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),\n            waitReconnectTask.result,\n          ]);\n\n          // If reconnect didn't trigger, tasks finished - exit loop\n          if (!waitReconnectTask.done) break;\n\n          // Reconnect triggered - clear event and continue loop\n          this.reconnectEvent.clear();\n        } finally {\n          // Cancel all tasks to ensure cleanup\n          await cancelAndWait(\n            [sendTask, wsListenerTask, recvTask, waitReconnectTask],\n            DEFAULT_CANCEL_TIMEOUT,\n          );\n          resourceCleanup();\n        }\n      } finally {\n        // Ensure cleanup even if connectWs throws\n        resourceCleanup();\n      }\n    }\n  }\n\n  private processTranscript(data: SttTranscriptEvent, isFinal: boolean) {\n    // Check if queue is closed to avoid race condition during disconnect\n    if (this.queue.closed) return;\n\n    const requestId = data.session_id || this.requestId;\n    const text = data.transcript;\n    const language = data.language || this.opts.language || 'en';\n\n    if (!text && !isFinal) return;\n\n    try {\n      // We'll have a more accurate way of detecting when speech started when we have VAD\n      if (!this.speaking) {\n        this.speaking = true;\n        this.queue.put({ type: SpeechEventType.START_OF_SPEECH });\n      }\n\n      const speechData: SpeechData = {\n        language,\n        startTime: this.startTimeOffset + data.start,\n        endTime: this.startTimeOffset + data.start + data.duration,\n        confidence: data.confidence,\n        text,\n        words: data.words.map(\n          (word): TimedString => ({\n            text: word.word,\n            startTime: word.start + this.startTimeOffset,\n            endTime: word.end + this.startTimeOffset,\n            startTimeOffset: this.startTimeOffset,\n            confidence: word.confidence,\n          }),\n        ),\n      };\n\n      if (isFinal) {\n        if (this.speechDuration > 0) {\n          this.queue.put({\n            type: SpeechEventType.RECOGNITION_USAGE,\n            requestId,\n            recognitionUsage: { audioDuration: this.speechDuration },\n          });\n          this.speechDuration = 0;\n        }\n\n        this.queue.put({\n          type: SpeechEventType.FINAL_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n\n        if (this.speaking) {\n          this.speaking = false;\n          this.queue.put({ type: SpeechEventType.END_OF_SPEECH });\n        }\n      } else {\n        this.queue.put({\n          type: SpeechEventType.INTERIM_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n      }\n    } catch (e) {\n      if (e instanceof Error && e.message.includes('Queue is closed')) {\n        // Expected behavior on disconnect, log as warning\n        this.#logger.warn(\n          { err: e },\n          'Queue closed during transcript processing (expected during disconnect)',\n        );\n      } else {\n        this.#logger.error({ err: e }, 'Error putting transcript to queue');\n      }\n    }\n  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package/dist/inference/stt.d.ts.map

@@ -1 +1 @@
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package/dist/inference/stt.js

@@ -10,6 +10,9 @@ import {
 } from "../stt/index.js";
 import { DEFAULT_API_CONNECT_OPTIONS } from "../types.js";
 import { Event, Task, cancelAndWait, shortuuid, waitForAbort } from "../utils.js";
+import {
+  sttServerEventSchema
+} from "./api_protos.js";
 import { connectWs, createAccessToken } from "./utils.js";
 const DEFAULT_ENCODING = "pcm_s16le";
 const DEFAULT_SAMPLE_RATE = 16e3;
@@ -20,7 +23,7 @@ class STT extends BaseSTT {
   streams = /* @__PURE__ */ new Set();
   #logger = log();
   constructor(opts) {
-    super({ streaming: true, interimResults: true });
+    super({ streaming: true, interimResults: true, alignedTranscript: "word" });
     const {
       model,
       language,
@@ -222,9 +225,16 @@ class SpeechStream extends BaseSpeechStream {
             const result = await reader.read();
             if (signal.aborted) return;
             if (result.done) return;
-            const json = result.value;
-            const type = json.type;
-            switch (type) {
+            const parseResult = await sttServerEventSchema.safeParseAsync(result.value);
+            if (!parseResult.success) {
+              this.#logger.warn(
+                { error: parseResult.error, rawData: result.value },
+                "Failed to parse STT server event"
+              );
+              continue;
+            }
+            const event = parseResult.data;
+            switch (event.type) {
               case "session.created":
               case "session.finalized":
                 break;
@@ -233,21 +243,15 @@ class SpeechStream extends BaseSpeechStream {
                 resourceCleanup();
                 break;
               case "interim_transcript":
-                this.processTranscript(json, false);
+                this.processTranscript(event, false);
                 break;
               case "final_transcript":
-                this.processTranscript(json, true);
+                this.processTranscript(event, true);
                 break;
               case "error":
-                this.#logger.error({ error: json }, "Received error from LiveKit STT");
+                this.#logger.error({ error: event }, "Received error from LiveKit STT");
                 resourceCleanup();
-                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);
-              default:
-                this.#logger.warn(
-                  { message: json },
-                  "Received unexpected message from LiveKit STT"
-                );
-                break;
+                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(event)}`);
             }
           }
         } finally {
@@ -290,9 +294,9 @@ class SpeechStream extends BaseSpeechStream {
   }
   processTranscript(data, isFinal) {
     if (this.queue.closed) return;
-    const requestId = data.request_id ?? this.requestId;
-    const text = data.transcript ?? "";
-    const language = data.language ?? this.opts.language ?? "en";
+    const requestId = data.session_id || this.requestId;
+    const text = data.transcript;
+    const language = data.language || this.opts.language || "en";
     if (!text && !isFinal) return;
     try {
       if (!this.speaking) {
@@ -301,10 +305,19 @@ class SpeechStream extends BaseSpeechStream {
       }
       const speechData = {
         language,
-        startTime: data.start ?? 0,
-        endTime: data.duration ?? 0,
-        confidence: data.confidence ?? 1,
-        text
+        startTime: this.startTimeOffset + data.start,
+        endTime: this.startTimeOffset + data.start + data.duration,
+        confidence: data.confidence,
+        text,
+        words: data.words.map(
+          (word) => ({
+            text: word.word,
+            startTime: word.start + this.startTimeOffset,
+            endTime: word.end + this.startTimeOffset,
+            startTimeOffset: this.startTimeOffset,
+            confidence: word.confidence
+          })
+        )
       };
       if (isFinal) {
         if (this.speechDuration > 0) {

package/dist/inference/stt.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/inference/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame } from '@livekit/rtc-node';\nimport type { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport {\n  STT as BaseSTT,\n  SpeechStream as BaseSpeechStream,\n  type SpeechData,\n  type SpeechEvent,\n  SpeechEventType,\n} from '../stt/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { type AudioBuffer, Event, Task, cancelAndWait, shortuuid, waitForAbort } from '../utils.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type DeepgramModels =\n  | 'deepgram'\n  | 'deepgram/nova-3'\n  | 'deepgram/nova-3-general'\n  | 'deepgram/nova-3-medical'\n  | 'deepgram/nova-2-conversationalai'\n  | 'deepgram/nova-2'\n  | 'deepgram/nova-2-general'\n  | 'deepgram/nova-2-medical'\n  | 'deepgram/nova-2-phonecall';\n\nexport type CartesiaModels = 'cartesia' | 'cartesia/ink-whisper';\n\nexport type AssemblyaiModels = 'assemblyai' | 'assemblyai/universal-streaming';\n\nexport interface CartesiaOptions {\n  min_volume?: number; // default: not specified\n  max_silence_duration_secs?: number; // default: not specified\n}\n\nexport interface DeepgramOptions {\n  filler_words?: boolean; // default: true\n  interim_results?: boolean; // default: true\n  endpointing?: number; // default: 25 (ms)\n  punctuate?: boolean; // default: false\n  smart_format?: boolean;\n  keywords?: Array<[string, number]>;\n  keyterms?: string[];\n  profanity_filter?: boolean;\n  numerals?: boolean;\n  mip_opt_out?: boolean;\n}\n\nexport interface AssemblyAIOptions {\n  format_turns?: boolean; // default: false\n  end_of_turn_confidence_threshold?: number; // default: 0.01\n  min_end_of_turn_silence_when_confident?: number; // default: 0\n  max_turn_silence?: number; // default: not specified\n  keyterms_prompt?: string[]; // default: not specified\n}\n\nexport type STTLanguages =\n  | 'multi'\n  | 'en'\n  | 'de'\n  | 'es'\n  | 'fr'\n  | 'ja'\n  | 'pt'\n  | 'zh'\n  | 'hi'\n  | AnyString;\n\ntype _STTModels = DeepgramModels | CartesiaModels | AssemblyaiModels;\n\nexport type STTModels = _STTModels | 'auto' | AnyString;\n\nexport type ModelWithLanguage = `${_STTModels}:${STTLanguages}` | STTModels;\n\nexport type STTOptions<TModel extends STTModels> = TModel extends DeepgramModels\n  ? DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? AssemblyAIOptions\n      : Record<string, unknown>;\n\nexport type STTEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: STTEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'wss://agent-gateway.livekit.cloud/v1';\nconst DEFAULT_CANCEL_TIMEOUT = 5000;\n\nexport interface InferenceSTTOptions<TModel extends STTModels> {\n  model?: TModel;\n  language?: STTLanguages;\n  encoding: STTEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: STTOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference STT\n */\nexport class STT<TModel extends STTModels> extends BaseSTT {\n  private opts: InferenceSTTOptions<TModel>;\n  private streams: Set<SpeechStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts?: {\n    model?: TModel;\n    language?: STTLanguages;\n    baseURL?: string;\n    encoding?: STTEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label(): string {\n    return 'inference.STT';\n  }\n\n  static fromModelString(modelString: string): STT<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, language] = modelString.split(':') as [AnyString, STTLanguages];\n      return new STT({ model, language });\n    }\n    return new STT({ model: modelString });\n  }\n\n  protected async _recognize(_: AudioBuffer): Promise<SpeechEvent> {\n    throw new Error('LiveKit STT does not support batch recognition, use stream() instead');\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  stream(options?: {\n    language?: STTLanguages | string;\n    connOptions?: APIConnectOptions;\n  }): SpeechStream<TModel> {\n    const { language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    const params = {\n      settings: {\n        sample_rate: String(this.opts.sampleRate),\n        encoding: this.opts.encoding,\n        extra: this.opts.modelOptions,\n      },\n    } as Record<string, unknown>;\n\n    if (this.opts.model && this.opts.model !== 'auto') {\n      params.model = this.opts.model;\n    }\n\n    if (this.opts.language) {\n      (params.settings as Record<string, unknown>).language = this.opts.language;\n    }\n\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/stt`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const socket = await connectWs(url, headers, timeout);\n    const msg = { ...params, type: 'session.create' };\n    socket.send(JSON.stringify(msg));\n\n    return socket;\n  }\n}\n\nexport class SpeechStream<TModel extends STTModels> extends BaseSpeechStream {\n  private opts: InferenceSTTOptions<TModel>;\n  private requestId = shortuuid('stt_request_');\n  private speaking = false;\n  private speechDuration = 0;\n  private reconnectEvent = new Event();\n  private stt: STT<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(\n    sttImpl: STT<TModel>,\n    opts: InferenceSTTOptions<TModel>,\n    connOptions: APIConnectOptions,\n  ) {\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.opts = opts;\n    this.stt = sttImpl;\n    this.connOptions = connOptions;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n    this.reconnectEvent.set();\n  }\n\n  protected async run(): Promise<void> {\n    while (true) {\n      // Create fresh resources for each connection attempt\n      let ws: WebSocket | null = null;\n      let closing = false;\n      let finalReceived = false;\n\n      type SttServerEvent = Record<string, any>;\n      const eventChannel = createStreamChannel<SttServerEvent>();\n\n      const resourceCleanup = () => {\n        if (closing) return;\n        closing = true;\n        eventChannel.close();\n        ws?.removeAllListeners();\n        ws?.close();\n      };\n\n      const createWsListener = async (ws: WebSocket, signal: AbortSignal) => {\n        return new Promise<void>((resolve, reject) => {\n          const onAbort = () => {\n            resourceCleanup();\n            reject(new Error('WebSocket connection aborted'));\n          };\n\n          signal.addEventListener('abort', onAbort, { once: true });\n\n          ws.on('message', (data) => {\n            const json = JSON.parse(data.toString()) as SttServerEvent;\n            eventChannel.write(json);\n          });\n\n          ws.on('error', (e) => {\n            this.#logger.error({ error: e }, 'WebSocket error');\n            resourceCleanup();\n            reject(e);\n          });\n\n          ws.on('close', (code: number) => {\n            resourceCleanup();\n\n            if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n            if (finalReceived) return resolve();\n\n            reject(\n              new APIStatusError({\n                message: 'LiveKit STT connection closed unexpectedly',\n                options: { statusCode: code },\n              }),\n            );\n          });\n        });\n      };\n\n      const send = async (socket: WebSocket, signal: AbortSignal) => {\n        const audioStream = new AudioByteStream(\n          this.opts.sampleRate,\n          1,\n          Math.floor(this.opts.sampleRate / 20), // 50ms\n        );\n\n        // Create abort promise once to avoid memory leak\n        const abortPromise = new Promise<never>((_, reject) => {\n          if (signal.aborted) {\n            return reject(new Error('Send aborted'));\n          }\n          const onAbort = () => reject(new Error('Send aborted'));\n          signal.addEventListener('abort', onAbort, { once: true });\n        });\n\n        // Manual iteration to support cancellation\n        const iterator = this.input[Symbol.asyncIterator]();\n        try {\n          while (true) {\n            const result = await Promise.race([iterator.next(), abortPromise]);\n\n            if (result.done) break;\n            const ev = result.value;\n\n            let frames: AudioFrame[];\n            if (ev === SpeechStream.FLUSH_SENTINEL) {\n              frames = audioStream.flush();\n            } else {\n              const frame = ev as AudioFrame;\n              frames = audioStream.write(new Int16Array(frame.data).buffer);\n            }\n\n            for (const frame of frames) {\n              this.speechDuration += frame.samplesPerChannel / frame.sampleRate;\n              const base64 = Buffer.from(frame.data.buffer).toString('base64');\n              const msg = { type: 'input_audio', audio: base64 };\n              socket.send(JSON.stringify(msg));\n            }\n          }\n\n          closing = true;\n          socket.send(JSON.stringify({ type: 'session.finalize' }));\n        } catch (e) {\n          if ((e as Error).message === 'Send aborted') {\n            // Expected abort, don't log\n            return;\n          }\n          throw e;\n        }\n      };\n\n      const recv = async (signal: AbortSignal) => {\n        const serverEventStream = eventChannel.stream();\n        const reader = serverEventStream.getReader();\n\n        try {\n          while (!this.closed && !signal.aborted) {\n            const result = await reader.read();\n            if (signal.aborted) return;\n            if (result.done) return;\n\n            const json = result.value;\n            const type = json.type as string | undefined;\n\n            switch (type) {\n              case 'session.created':\n              case 'session.finalized':\n                break;\n              case 'session.closed':\n                finalReceived = true;\n                resourceCleanup();\n                break;\n              case 'interim_transcript':\n                this.processTranscript(json, false);\n                break;\n              case 'final_transcript':\n                this.processTranscript(json, true);\n                break;\n              case 'error':\n                this.#logger.error({ error: json }, 'Received error from LiveKit STT');\n                resourceCleanup();\n                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);\n              default:\n                this.#logger.warn(\n                  { message: json },\n                  'Received unexpected message from LiveKit STT',\n                );\n                break;\n            }\n          }\n        } finally {\n          reader.releaseLock();\n          try {\n            await serverEventStream.cancel();\n          } catch (e) {\n            this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n          }\n        }\n      };\n\n      try {\n        ws = await this.stt.connectWs(this.connOptions.timeoutMs);\n\n        const controller = this.abortController; // Use base class abortController for proper cancellation\n        const sendTask = Task.from(({ signal }) => send(ws!, signal), controller);\n        const wsListenerTask = Task.from(({ signal }) => createWsListener(ws!, signal), controller);\n        const recvTask = Task.from(({ signal }) => recv(signal), controller);\n        const waitReconnectTask = Task.from(\n          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),\n          controller,\n        );\n\n        try {\n          await Promise.race([\n            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),\n            waitReconnectTask.result,\n          ]);\n\n          // If reconnect didn't trigger, tasks finished - exit loop\n          if (!waitReconnectTask.done) break;\n\n          // Reconnect triggered - clear event and continue loop\n          this.reconnectEvent.clear();\n        } finally {\n          // Cancel all tasks to ensure cleanup\n          await cancelAndWait(\n            [sendTask, wsListenerTask, recvTask, waitReconnectTask],\n            DEFAULT_CANCEL_TIMEOUT,\n          );\n          resourceCleanup();\n        }\n      } finally {\n        // Ensure cleanup even if connectWs throws\n        resourceCleanup();\n      }\n    }\n  }\n\n  private processTranscript(data: Record<string, any>, isFinal: boolean) {\n    // Check if queue is closed to avoid race condition during disconnect\n    if (this.queue.closed) return;\n\n    const requestId = data.request_id ?? this.requestId;\n    const text = data.transcript ?? '';\n    const language = data.language ?? this.opts.language ?? 'en';\n\n    if (!text && !isFinal) return;\n\n    try {\n      // We'll have a more accurate way of detecting when speech started when we have VAD\n      if (!this.speaking) {\n        this.speaking = true;\n        this.queue.put({ type: SpeechEventType.START_OF_SPEECH });\n      }\n\n      const speechData: SpeechData = {\n        language,\n        startTime: data.start ?? 0,\n        endTime: data.duration ?? 0,\n        confidence: data.confidence ?? 1.0,\n        text,\n      };\n\n      if (isFinal) {\n        if (this.speechDuration > 0) {\n          this.queue.put({\n            type: SpeechEventType.RECOGNITION_USAGE,\n            requestId,\n            recognitionUsage: { audioDuration: this.speechDuration },\n          });\n          this.speechDuration = 0;\n        }\n\n        this.queue.put({\n          type: SpeechEventType.FINAL_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n\n        if (this.speaking) {\n          this.speaking = false;\n          this.queue.put({ type: SpeechEventType.END_OF_SPEECH });\n        }\n      } else {\n        this.queue.put({\n          type: SpeechEventType.INTERIM_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n      }\n    } catch (e) {\n      if (e instanceof Error && e.message.includes('Queue is closed')) {\n        // Expected behavior on disconnect, log as warning\n        this.#logger.warn(\n          { err: e },\n          'Queue closed during transcript processing (expected during disconnect)',\n        );\n      } else {\n        this.#logger.error({ err: e }, 'Error putting transcript to queue');\n      }\n    }\n  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+{"version":3,"sources":["../../src/inference/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame } from '@livekit/rtc-node';\nimport type { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport {\n  STT as BaseSTT,\n  SpeechStream as BaseSpeechStream,\n  type SpeechData,\n  type SpeechEvent,\n  SpeechEventType,\n} from '../stt/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { type AudioBuffer, Event, Task, cancelAndWait, shortuuid, waitForAbort } from '../utils.js';\nimport type { TimedString } from '../voice/io.js';\nimport {\n  type SttServerEvent,\n  type SttTranscriptEvent,\n  sttServerEventSchema,\n} from './api_protos.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type DeepgramModels =\n  | 'deepgram'\n  | 'deepgram/nova-3'\n  | 'deepgram/nova-3-general'\n  | 'deepgram/nova-3-medical'\n  | 'deepgram/nova-2-conversationalai'\n  | 'deepgram/nova-2'\n  | 'deepgram/nova-2-general'\n  | 'deepgram/nova-2-medical'\n  | 'deepgram/nova-2-phonecall';\n\nexport type CartesiaModels = 'cartesia' | 'cartesia/ink-whisper';\n\nexport type AssemblyaiModels = 'assemblyai' | 'assemblyai/universal-streaming';\n\nexport interface CartesiaOptions {\n  min_volume?: number; // default: not specified\n  max_silence_duration_secs?: number; // default: not specified\n}\n\nexport interface DeepgramOptions {\n  filler_words?: boolean; // default: true\n  interim_results?: boolean; // default: true\n  endpointing?: number; // default: 25 (ms)\n  punctuate?: boolean; // default: false\n  smart_format?: boolean;\n  keywords?: Array<[string, number]>;\n  keyterms?: string[];\n  profanity_filter?: boolean;\n  numerals?: boolean;\n  mip_opt_out?: boolean;\n}\n\nexport interface AssemblyAIOptions {\n  format_turns?: boolean; // default: false\n  end_of_turn_confidence_threshold?: number; // default: 0.01\n  min_end_of_turn_silence_when_confident?: number; // default: 0\n  max_turn_silence?: number; // default: not specified\n  keyterms_prompt?: string[]; // default: not specified\n}\n\nexport type STTLanguages =\n  | 'multi'\n  | 'en'\n  | 'de'\n  | 'es'\n  | 'fr'\n  | 'ja'\n  | 'pt'\n  | 'zh'\n  | 'hi'\n  | AnyString;\n\ntype _STTModels = DeepgramModels | CartesiaModels | AssemblyaiModels;\n\nexport type STTModels = _STTModels | 'auto' | AnyString;\n\nexport type ModelWithLanguage = `${_STTModels}:${STTLanguages}` | STTModels;\n\nexport type STTOptions<TModel extends STTModels> = TModel extends DeepgramModels\n  ? DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? AssemblyAIOptions\n      : Record<string, unknown>;\n\nexport type STTEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: STTEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'wss://agent-gateway.livekit.cloud/v1';\nconst DEFAULT_CANCEL_TIMEOUT = 5000;\n\nexport interface InferenceSTTOptions<TModel extends STTModels> {\n  model?: TModel;\n  language?: STTLanguages;\n  encoding: STTEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: STTOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference STT\n */\nexport class STT<TModel extends STTModels> extends BaseSTT {\n  private opts: InferenceSTTOptions<TModel>;\n  private streams: Set<SpeechStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts?: {\n    model?: TModel;\n    language?: STTLanguages;\n    baseURL?: string;\n    encoding?: STTEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true, alignedTranscript: 'word' });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label(): string {\n    return 'inference.STT';\n  }\n\n  static fromModelString(modelString: string): STT<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, language] = modelString.split(':') as [AnyString, STTLanguages];\n      return new STT({ model, language });\n    }\n    return new STT({ model: modelString });\n  }\n\n  protected async _recognize(_: AudioBuffer): Promise<SpeechEvent> {\n    throw new Error('LiveKit STT does not support batch recognition, use stream() instead');\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  stream(options?: {\n    language?: STTLanguages | string;\n    connOptions?: APIConnectOptions;\n  }): SpeechStream<TModel> {\n    const { language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    const params = {\n      settings: {\n        sample_rate: String(this.opts.sampleRate),\n        encoding: this.opts.encoding,\n        extra: this.opts.modelOptions,\n      },\n    } as Record<string, unknown>;\n\n    if (this.opts.model && this.opts.model !== 'auto') {\n      params.model = this.opts.model;\n    }\n\n    if (this.opts.language) {\n      (params.settings as Record<string, unknown>).language = this.opts.language;\n    }\n\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/stt`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const socket = await connectWs(url, headers, timeout);\n    const msg = { ...params, type: 'session.create' };\n    socket.send(JSON.stringify(msg));\n\n    return socket;\n  }\n}\n\nexport class SpeechStream<TModel extends STTModels> extends BaseSpeechStream {\n  private opts: InferenceSTTOptions<TModel>;\n  private requestId = shortuuid('stt_request_');\n  private speaking = false;\n  private speechDuration = 0;\n  private reconnectEvent = new Event();\n  private stt: STT<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(\n    sttImpl: STT<TModel>,\n    opts: InferenceSTTOptions<TModel>,\n    connOptions: APIConnectOptions,\n  ) {\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.opts = opts;\n    this.stt = sttImpl;\n    this.connOptions = connOptions;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n    this.reconnectEvent.set();\n  }\n\n  protected async run(): Promise<void> {\n    while (true) {\n      // Create fresh resources for each connection attempt\n      let ws: WebSocket | null = null;\n      let closing = false;\n      let finalReceived = false;\n\n      const eventChannel = createStreamChannel<SttServerEvent>();\n\n      const resourceCleanup = () => {\n        if (closing) return;\n        closing = true;\n        eventChannel.close();\n        ws?.removeAllListeners();\n        ws?.close();\n      };\n\n      const createWsListener = async (ws: WebSocket, signal: AbortSignal) => {\n        return new Promise<void>((resolve, reject) => {\n          const onAbort = () => {\n            resourceCleanup();\n            reject(new Error('WebSocket connection aborted'));\n          };\n\n          signal.addEventListener('abort', onAbort, { once: true });\n\n          ws.on('message', (data) => {\n            const json = JSON.parse(data.toString()) as SttServerEvent;\n            eventChannel.write(json);\n          });\n\n          ws.on('error', (e) => {\n            this.#logger.error({ error: e }, 'WebSocket error');\n            resourceCleanup();\n            reject(e);\n          });\n\n          ws.on('close', (code: number) => {\n            resourceCleanup();\n\n            if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n            if (finalReceived) return resolve();\n\n            reject(\n              new APIStatusError({\n                message: 'LiveKit STT connection closed unexpectedly',\n                options: { statusCode: code },\n              }),\n            );\n          });\n        });\n      };\n\n      const send = async (socket: WebSocket, signal: AbortSignal) => {\n        const audioStream = new AudioByteStream(\n          this.opts.sampleRate,\n          1,\n          Math.floor(this.opts.sampleRate / 20), // 50ms\n        );\n\n        // Create abort promise once to avoid memory leak\n        const abortPromise = new Promise<never>((_, reject) => {\n          if (signal.aborted) {\n            return reject(new Error('Send aborted'));\n          }\n          const onAbort = () => reject(new Error('Send aborted'));\n          signal.addEventListener('abort', onAbort, { once: true });\n        });\n\n        // Manual iteration to support cancellation\n        const iterator = this.input[Symbol.asyncIterator]();\n        try {\n          while (true) {\n            const result = await Promise.race([iterator.next(), abortPromise]);\n\n            if (result.done) break;\n            const ev = result.value;\n\n            let frames: AudioFrame[];\n            if (ev === SpeechStream.FLUSH_SENTINEL) {\n              frames = audioStream.flush();\n            } else {\n              const frame = ev as AudioFrame;\n              frames = audioStream.write(new Int16Array(frame.data).buffer);\n            }\n\n            for (const frame of frames) {\n              this.speechDuration += frame.samplesPerChannel / frame.sampleRate;\n              const base64 = Buffer.from(frame.data.buffer).toString('base64');\n              const msg = { type: 'input_audio', audio: base64 };\n              socket.send(JSON.stringify(msg));\n            }\n          }\n\n          closing = true;\n          socket.send(JSON.stringify({ type: 'session.finalize' }));\n        } catch (e) {\n          if ((e as Error).message === 'Send aborted') {\n            // Expected abort, don't log\n            return;\n          }\n          throw e;\n        }\n      };\n\n      const recv = async (signal: AbortSignal) => {\n        const serverEventStream = eventChannel.stream();\n        const reader = serverEventStream.getReader();\n\n        try {\n          while (!this.closed && !signal.aborted) {\n            const result = await reader.read();\n            if (signal.aborted) return;\n            if (result.done) return;\n\n            // Parse and validate with Zod schema\n            const parseResult = await sttServerEventSchema.safeParseAsync(result.value);\n            if (!parseResult.success) {\n              this.#logger.warn(\n                { error: parseResult.error, rawData: result.value },\n                'Failed to parse STT server event',\n              );\n              continue;\n            }\n\n            const event: SttServerEvent = parseResult.data;\n\n            switch (event.type) {\n              case 'session.created':\n              case 'session.finalized':\n                break;\n              case 'session.closed':\n                finalReceived = true;\n                resourceCleanup();\n                break;\n              case 'interim_transcript':\n                this.processTranscript(event, false);\n                break;\n              case 'final_transcript':\n                this.processTranscript(event, true);\n                break;\n              case 'error':\n                this.#logger.error({ error: event }, 'Received error from LiveKit STT');\n                resourceCleanup();\n                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(event)}`);\n            }\n          }\n        } finally {\n          reader.releaseLock();\n          try {\n            await serverEventStream.cancel();\n          } catch (e) {\n            this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n          }\n        }\n      };\n\n      try {\n        ws = await this.stt.connectWs(this.connOptions.timeoutMs);\n\n        const controller = this.abortController; // Use base class abortController for proper cancellation\n        const sendTask = Task.from(({ signal }) => send(ws!, signal), controller);\n        const wsListenerTask = Task.from(({ signal }) => createWsListener(ws!, signal), controller);\n        const recvTask = Task.from(({ signal }) => recv(signal), controller);\n        const waitReconnectTask = Task.from(\n          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),\n          controller,\n        );\n\n        try {\n          await Promise.race([\n            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),\n            waitReconnectTask.result,\n          ]);\n\n          // If reconnect didn't trigger, tasks finished - exit loop\n          if (!waitReconnectTask.done) break;\n\n          // Reconnect triggered - clear event and continue loop\n          this.reconnectEvent.clear();\n        } finally {\n          // Cancel all tasks to ensure cleanup\n          await cancelAndWait(\n            [sendTask, wsListenerTask, recvTask, waitReconnectTask],\n            DEFAULT_CANCEL_TIMEOUT,\n          );\n          resourceCleanup();\n        }\n      } finally {\n        // Ensure cleanup even if connectWs throws\n        resourceCleanup();\n      }\n    }\n  }\n\n  private processTranscript(data: SttTranscriptEvent, isFinal: boolean) {\n    // Check if queue is closed to avoid race condition during disconnect\n    if (this.queue.closed) return;\n\n    const requestId = data.session_id || this.requestId;\n    const text = data.transcript;\n    const language = data.language || this.opts.language || 'en';\n\n    if (!text && !isFinal) return;\n\n    try {\n      // We'll have a more accurate way of detecting when speech started when we have VAD\n      if (!this.speaking) {\n        this.speaking = true;\n        this.queue.put({ type: SpeechEventType.START_OF_SPEECH });\n      }\n\n      const speechData: SpeechData = {\n        language,\n        startTime: this.startTimeOffset + data.start,\n        endTime: this.startTimeOffset + data.start + data.duration,\n        confidence: data.confidence,\n        text,\n        words: data.words.map(\n          (word): TimedString => ({\n            text: word.word,\n            startTime: word.start + this.startTimeOffset,\n            endTime: word.end + this.startTimeOffset,\n            startTimeOffset: this.startTimeOffset,\n            confidence: word.confidence,\n          }),\n        ),\n      };\n\n      if (isFinal) {\n        if (this.speechDuration > 0) {\n          this.queue.put({\n            type: SpeechEventType.RECOGNITION_USAGE,\n            requestId,\n            recognitionUsage: { audioDuration: this.speechDuration },\n          });\n          this.speechDuration = 0;\n        }\n\n        this.queue.put({\n          type: SpeechEventType.FINAL_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n\n        if (this.speaking) {\n          this.speaking = false;\n          this.queue.put({ type: SpeechEventType.END_OF_SPEECH });\n        }\n      } else {\n        this.queue.put({\n          type: SpeechEventType.INTERIM_TRANSCRIPT,\n          requestId,\n          alternatives: [speechData],\n        });\n      }\n    } catch (e) {\n      if (e instanceof Error && e.message.includes('Queue is closed')) {\n        // Expected behavior on disconnect, log as warning\n        this.#logger.warn(\n          { err: e },\n          'Queue closed during transcript processing (expected during disconnect)',\n        );\n      } else {\n        this.#logger.error({ err: e }, 'Error putting transcript to queue');\n      }\n    }\n  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package/dist/ipc/inference_proc_executor.cjs.map

@@ -1 +1 @@
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{ execArgv: process.execArgv } : undefined;\n\n    return fork(forkUrl, [this.#agent], forkOptions);\n  }\n\n  async mainTask(proc: ChildProcess) {\n    proc.on('message', (msg: IPCMessage) => {\n      switch (msg.case) {\n        case 'inferenceRequest':\n          this.#inferenceTasks.push(this.#doInferenceTask(proc, msg.value));\n      }\n    });\n  }\n\n  async #doInferenceTask(\n    proc: ChildProcess,\n    req: { method: string; requestId: string; data: unknown },\n  ) {\n    if (!this.#inferenceExecutor) {\n      this.#logger.warn('inference request received but no inference executor');\n      proc.send({\n        case: 'inferenceResponse',\n        value: { requestId: req.requestId, error: new Error('no inference executor') },\n      });\n      return;\n    }\n\n    try {\n      const data = await this.#inferenceExecutor.doInference(req.method, req.data);\n      proc.send({ case: 'inferenceResponse', value: { requestId: req.requestId, data } });\n    } catch (error) {\n      proc.send({ case: 'inferenceResponse', value: { requestId: req.requestId, error } });\n    }\n  }\n\n  async launchJob(info: RunningJobInfo) {\n    if (this.#runningJob) {\n      throw Error('process already has a running job');\n    }\n    if (!this.init.done) {\n      throw Error('process not initialized');\n    }\n    this.#jobStatus = JobStatus.RUNNING;\n    this.#runningJob = info;\n\n    this.proc!.send({ case: 'startJobRequest', value: { runningJob: info } });\n  }\n}\n","// Shim globals in cjs bundle\n// There's a weird bug that esbuild will always inject importMetaUrl\n// if we export it as `const importMetaUrl = ... __filename ...`\n// But using a function will not cause this issue\n\nconst getImportMetaUrl = () =>\n  typeof document === 'undefined'\n    ? 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package/dist/stt/stt.cjs

@@ -82,6 +82,7 @@ class SpeechStream {
   deferredInputStream;
   logger = (0, import_log.log)();
   _connOptions;
+  _startTimeOffset = 0;
   abortController = new AbortController();
   constructor(stt, sampleRate, connectionOptions = import_types.DEFAULT_API_CONNECT_OPTIONS) {
     this.#stt = stt;
@@ -181,6 +182,15 @@ class SpeechStream {
   get abortSignal() {
     return this.abortController.signal;
   }
+  get startTimeOffset() {
+    return this._startTimeOffset;
+  }
+  set startTimeOffset(value) {
+    if (value < 0) {
+      throw new Error("startTimeOffset must be non-negative");
+    }
+    this._startTimeOffset = value;
+  }
   updateInputStream(audioStream) {
     this.deferredInputStream.setSource(audioStream);
   }

package/dist/stt/stt.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): SpeechStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n\n  protected abortController = new AbortController();\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().finally(() => this.queue.close()));\n  }\n\n  private async mainTask() {\n    for (let i = 0; i < this._connOptions.maxRetry + 1; i++) {\n      try {\n        return await this.run();\n      } catch (error) {\n        if (error instanceof APIError) {\n          const retryInterval = intervalForRetry(this._connOptions, i);\n\n          if (this._connOptions.maxRetry === 0 || !error.retryable) {\n            this.emitError({ error, recoverable: false });\n            throw error;\n          } else if (i === this._connOptions.maxRetry) {\n            this.emitError({ error, recoverable: false });\n            throw new APIConnectionError({\n              message: `failed to recognize speech after ${this._connOptions.maxRetry + 1} attempts`,\n              options: { retryable: false },\n            });\n          } else {\n            // Don't emit error event for recoverable errors during retry loop\n            // to avoid ERR_UNHANDLED_ERROR or premature session termination\n            this.logger.warn(\n              { tts: this.#stt.label, attempt: i + 1, error },\n              `failed to recognize speech, retrying in ${retryInterval}s`,\n            );\n          }\n\n          if (retryInterval > 0) {\n            await delay(retryInterval);\n          }\n        } else {\n          this.emitError({ error: toError(error), recoverable: false });\n          throw error;\n        }\n      }\n    }\n  }\n\n  private emitError({ error, recoverable }: { error: Error; recoverable: boolean }) {\n    this.#stt.emit('error', {\n      type: 'stt_error',\n      timestamp: Date.now(),\n      label: this.#stt.label,\n      error,\n      recoverable,\n    });\n  }\n\n  protected async pumpInput() {\n    // TODO(AJS-35): Implement STT with webstreams API\n    const inputStream = this.deferredInputStream.stream;\n    const reader = inputStream.getReader();\n\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        this.pushFrame(value);\n      }\n    } catch (error) {\n      this.logger.error('Error in STTStream mainTask:', error);\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    for await (const event of this.queue) {\n      if (!this.output.closed) {\n        try {\n          this.output.put(event);\n        } catch (e) {\n          if (e instanceof Error && e.message.includes('Queue is closed')) {\n            this.logger.warn(\n              { err: e },\n              'Queue closed during transcript processing (expected during disconnect)',\n            );\n          }\n        }\n      }\n      if (event.type !== SpeechEventType.RECOGNITION_USAGE) continue;\n      const metrics: STTMetrics = {\n        type: 'stt_metrics',\n        timestamp: Date.now(),\n        requestId: event.requestId!,\n        durationMs: 0,\n        label: this.#stt.label,\n        audioDurationMs: Math.round(event.recognitionUsage!.audioDuration * 1000),\n        streamed: true,\n      };\n      this.#stt.emit('metrics_collected', metrics);\n    }\n    if (!this.output.closed) {\n      this.output.close();\n    }\n  }\n\n  protected abstract run(): Promise<void>;\n\n  protected get abortSignal(): AbortSignal {\n    return this.abortController.signal;\n  }\n\n  updateInputStream(audioStream: ReadableStream<AudioFrame>) {\n    this.deferredInputStream.setSource(audioStream);\n  }\n\n  detachInputStream() {\n    this.deferredInputStream.detachSource();\n  }\n\n  /** Push an audio frame to the STT */\n  pushFrame(frame: AudioFrame) {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n\n    if (this.neededSampleRate && frame.sampleRate !== this.neededSampleRate) {\n      if 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+{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\nimport type { TimedString } from '../voice/index.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n  words?: TimedString[];\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n  /**\n   * Whether this STT supports aligned transcripts with word/chunk timestamps.\n   * - 'word': Provider returns word-level timestamps\n   * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)\n   * - false: Provider does not support aligned transcripts\n   */\n  alignedTranscript?: 'word' | 'chunk' | false;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): SpeechStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n  private _startTimeOffset: number = 0;\n\n  protected abortController = new AbortController();\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().finally(() => this.queue.close()));\n  }\n\n  private async mainTask() {\n    for (let i = 0; i < this._connOptions.maxRetry + 1; i++) {\n      try {\n        return await this.run();\n      } catch (error) {\n        if (error instanceof APIError) {\n          const retryInterval = intervalForRetry(this._connOptions, i);\n\n          if (this._connOptions.maxRetry === 0 || !error.retryable) {\n            this.emitError({ error, recoverable: false });\n            throw error;\n          } else if (i === this._connOptions.maxRetry) {\n            this.emitError({ error, recoverable: false });\n            throw new APIConnectionError({\n              message: `failed to recognize speech after ${this._connOptions.maxRetry + 1} attempts`,\n              options: { retryable: false },\n            });\n          } else {\n            // Don't emit 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   if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n\n    if (this.neededSampleRate && frame.sampleRate !== this.neededSampleRate) {\n      if (!this.resampler) {\n        this.resampler = new AudioResampler(frame.sampleRate, this.neededSampleRate);\n      }\n    }\n\n    if (this.resampler) {\n      const frames = this.resampler.push(frame);\n      for (const frame of frames) {\n        this.input.put(frame);\n      }\n    } else {\n      this.input.put(frame);\n    }\n  }\n\n  /** Flush the STT, causing it to process all pending text */\n  flush() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.put(SpeechStream.FLUSH_SENTINEL);\n  }\n\n  /** Mark the input as ended and forbid additional pushes */\n  endInput() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.close();\n  }\n\n  next(): Promise<IteratorResult<SpeechEvent>> {\n    return this.output.next();\n  }\n\n  /** Close both the input and output of the STT stream */\n  close() {\n    if (!this.input.closed) this.input.close();\n    if (!this.queue.closed) this.queue.close();\n    if (!this.output.closed) this.output.close();\n    if (!this.abortController.signal.aborted) this.abortController.abort();\n    this.closed = true;\n  }\n\n  [Symbol.asyncIterator](): SpeechStream {\n    return this;\n  }\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAGA,sBAAgD;AAEhD,yBAA6B;AAE7B,wBAA6C;AAC7C,mBAA8C;AAC9C,iBAAoB;AAEpB,6BAAuC;AACvC,mBAAsF;AAEtF,mBAA8D;AAIvD,IAAK,kBAAL,kBAAKA,qBAAL;AAML,EAAAA,kCAAA,qBAAkB,KAAlB;AAIA,EAAAA,kCAAA,wBAAqB,KAArB;AAKA,EAAAA,kCAAA,sBAAmB,KAAnB;AAKA,EAAAA,kCAAA,mBAAgB,KAAhB;AAEA,EAAAA,kCAAA,uBAAoB,KAApB;AAMA,EAAAA,kCAAA,0BAAuB,KAAvB;AA5BU,SAAAA;AAAA,GAAA;AA2FL,MAAe,YAAa,gCAAsD;AAAA,EAEvF;AAAA,EAEA,YAAY,cAA+B;AACzC,UAAM;AACN,SAAK,gBAAgB;AAAA,EACvB;AAAA;AAAA,EAGA,IAAI,eAAgC;AAClC,WAAO,KAAK;AAAA,EACd;AAAA;AAAA,EAGA,MAAM,UAAU,OAAoB,aAAiD;AACnF,UAAM,YAAY,QAAQ,OAAO,OAAO;AACxC,UAAM,QAAQ,MAAM,KAAK,WAAW,OAAO,WAAW;AACtD,UAAM,aAAa,QAAQ,QAAQ,OAAO,OAAO,IAAI,aAAa,OAAO,GAAO,CAAC;AACjF,SAAK,KAAK,qBAAqB;AAAA,MAC7B,MAAM;AAAA,MACN,WAAW,MAAM,aAAa;AAAA,MAC9B,WAAW,KAAK,IAAI;AAAA,MACpB;AAAA,MACA,OAAO,KAAK;AAAA,MACZ,iBAAiB,KAAK,UAAM,4CAA8B,KAAK,IAAI,GAAI;AAAA,MACvE,UAAU;AAAA,IACZ,CAAC;AACD,WAAO;AAAA,EACT;AAAA,EAeA,MAAM,QAAuB;AAC3B;AAAA,EACF;AACF;AAkBO,MAAe,aAA2D;AAAA,EAC/E,OAA0B,iBAAiB,OAAO,gBAAgB;AAAA,EACxD,QAAQ,IAAI,gCAAoE;AAAA,EAChF,SAAS,IAAI,gCAAgC;AAAA,EAC7C,QAAQ,IAAI,gCAAgC;AAAA,EAC5C;AAAA,EACA;AAAA,EAEA,SAAS;AAAA,EACnB;AAAA,EACQ;AAAA,EACA,aAAS,gBAAI;AAAA,EACb;AAAA,EACA,mBAA2B;AAAA,EAEzB,kBAAkB,IAAI,gBAAgB;AAAA,EAEhD,YACE,KACA,YACA,oBAAuC,0CACvC;AACA,SAAK,OAAO;AACZ,SAAK,eAAe;AACpB,SAAK,sBAAsB,IAAI,8CAAmC;AAClE,SAAK,mBAAmB;AACxB,SAAK,eAAe;AACpB,SAAK,UAAU;AAMf,gCAAU,MAAM,KAAK,SAAS,EAAE,QAAQ,MAAM,KAAK,MAAM,MAAM,CAAC,CAAC;AAAA,EACnE;AAAA,EAEA,MAAc,WAAW;AACvB,aAAS,IAAI,GAAG,IAAI,KAAK,aAAa,WAAW,GAAG,KAAK;AACvD,UAAI;AACF,eAAO,MAAM,KAAK,IAAI;AAAA,MACxB,SAAS,OAAO;AACd,YAAI,iBAAiB,4BAAU;AAC7B,gBAAM,oBAAgB,+BAAiB,KAAK,cAAc,CAAC;AAE3D,cAAI,KAAK,aAAa,aAAa,KAAK,CAAC,MAAM,WAAW;AACxD,iBAAK,UAAU,EAAE,OAAO,aAAa,MAAM,CAAC;AAC5C,kBAAM;AAAA,UACR,WAAW,MAAM,KAAK,aAAa,UAAU;AAC3C,iBAAK,UAAU,EAAE,OAAO,aAAa,MAAM,CAAC;AAC5C,kBAAM,IAAI,qCAAmB;AAAA,cAC3B,SAAS,oCAAoC,KAAK,aAAa,WAAW,CAAC;AAAA,cAC3E,SAAS,EAAE,WAAW,MAAM;AAAA,YAC9B,CAAC;AAAA,UACH,OAAO;AAGL,iBAAK,OAAO;AAAA,cACV,EAAE,KAAK,KAAK,KAAK,OAAO,SAAS,IAAI,GAAG,MAAM;AAAA,cAC9C,2CAA2C,aAAa;AAAA,YAC1D;AAAA,UACF;AAEA,cAAI,gBAAgB,GAAG;AACrB,sBAAM,oBAAM,aAAa;AAAA,UAC3B;AAAA,QACF,OAAO;AACL,eAAK,UAAU,EAAE,WAAO,sBAAQ,KAAK,GAAG,aAAa,MAAM,CAAC;AAC5D,gBAAM;AAAA,QACR;AAAA,MACF;AAAA,IACF;AAAA,EACF;AAAA,EAEQ,UAAU,EAAE,OAAO,YAAY,GAA2C;AAChF,SAAK,KAAK,KAAK,SAAS;AAAA,MACtB,MAAM;AAAA,MACN,WAAW,KAAK,IAAI;AAAA,MACpB,OAAO,KAAK,KAAK;AAAA,MACjB;AAAA,MACA;AAAA,IACF,CAAC;AAAA,EACH;AAAA,EAEA,MAAgB,YAAY;AAE1B,UAAM,cAAc,KAAK,oBAAoB;AAC7C,UAAM,SAAS,YAAY,UAAU;AAErC,QAAI;AACF,aAAO,MAAM;AACX,cAAM,EAAE,MAAM,MAAM,IAAI,MAAM,OAAO,KAAK;AAC1C,YAAI,KAAM;AACV,aAAK,UAAU,KAAK;AAAA,MACtB;AAAA,IACF,SAAS,OAAO;AACd,WAAK,OAAO,MAAM,gCAAgC,KAAK;AAAA,IACzD,UAAE;AACA,aAAO,YAAY;AAAA,IACrB;AAAA,EACF;AAAA,EAEA,MAAgB,iBAAiB;AAC/B,qBAAiB,SAAS,KAAK,OAAO;AACpC,UAAI,CAAC,KAAK,OAAO,QAAQ;AACvB,YAAI;AACF,eAAK,OAAO,IAAI,KAAK;AAAA,QACvB,SAAS,GAAG;AACV,cAAI,aAAa,SAAS,EAAE,QAAQ,SAAS,iBAAiB,GAAG;AAC/D,iBAAK,OAAO;AAAA,cACV,EAAE,KAAK,EAAE;AAAA,cACT;AAAA,YACF;AAAA,UACF;AAAA,QACF;AAAA,MACF;AACA,UAAI,MAAM,SAAS,0BAAmC;AACtD,YAAM,UAAsB;AAAA,QAC1B,MAAM;AAAA,QACN,WAAW,KAAK,IAAI;AAAA,QACpB,WAAW,MAAM;AAAA,QACjB,YAAY;AAAA,QACZ,OAAO,KAAK,KAAK;AAAA,QACjB,iBAAiB,KAAK,MAAM,MAAM,iBAAkB,gBAAgB,GAAI;AAAA,QACxE,UAAU;AAAA,MACZ;AACA,WAAK,KAAK,KAAK,qBAAqB,OAAO;AAAA,IAC7C;AACA,QAAI,CAAC,KAAK,OAAO,QAAQ;AACvB,WAAK,OAAO,MAAM;AAAA,IACpB;AAAA,EACF;AAAA,EAIA,IAAc,cAA2B;AACvC,WAAO,KAAK,gBAAgB;AAAA,EAC9B;AAAA,EAEA,IAAI,kBAA0B;AAC5B,WAAO,KAAK;AAAA,EACd;AAAA,EAEA,IAAI,gBAAgB,OAAe;AACjC,QAAI,QAAQ,GAAG;AACb,YAAM,IAAI,MAAM,sCAAsC;AAAA,IACxD;AACA,SAAK,mBAAmB;AAAA,EAC1B;AAAA,EAEA,kBAAkB,aAAyC;AACzD,SAAK,oBAAoB,UAAU,WAAW;AAAA,EAChD;AAAA,EAEA,oBAAoB;AAClB,SAAK,oBAAoB,aAAa;AAAA,EACxC;AAAA;AAAA,EAGA,UAAU,OAAmB;AAC3B,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AAEA,QAAI,KAAK,oBAAoB,MAAM,eAAe,KAAK,kBAAkB;AACvE,UAAI,CAAC,KAAK,WAAW;AACnB,aAAK,YAAY,IAAI,+BAAe,MAAM,YAAY,KAAK,gBAAgB;AAAA,MAC7E;AAAA,IACF;AAEA,QAAI,KAAK,WAAW;AAClB,YAAM,SAAS,KAAK,UAAU,KAAK,KAAK;AACxC,iBAAWC,UAAS,QAAQ;AAC1B,aAAK,MAAM,IAAIA,MAAK;AAAA,MACtB;AAAA,IACF,OAAO;AACL,WAAK,MAAM,IAAI,KAAK;AAAA,IACtB;AAAA,EACF;AAAA;AAAA,EAGA,QAAQ;AACN,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AACA,SAAK,MAAM,IAAI,aAAa,cAAc;AAAA,EAC5C;AAAA;AAAA,EAGA,WAAW;AACT,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AACA,SAAK,MAAM,MAAM;AAAA,EACnB;AAAA,EAEA,OAA6C;AAC3C,WAAO,KAAK,OAAO,KAAK;AAAA,EAC1B;AAAA;AAAA,EAGA,QAAQ;AACN,QAAI,CAAC,KAAK,MAAM,OAAQ,MAAK,MAAM,MAAM;AACzC,QAAI,CAAC,KAAK,MAAM,OAAQ,MAAK,MAAM,MAAM;AACzC,QAAI,CAAC,KAAK,OAAO,OAAQ,MAAK,OAAO,MAAM;AAC3C,QAAI,CAAC,KAAK,gBAAgB,OAAO,QAAS,MAAK,gBAAgB,MAAM;AACrE,SAAK,SAAS;AAAA,EAChB;AAAA,EAEA,CAAC,OAAO,aAAa,IAAkB;AACrC,WAAO;AAAA,EACT;AACF;","names":["SpeechEventType","frame"]}

package/dist/stt/stt.d.cts

@@ -6,6 +6,7 @@ import type { STTMetrics } from '../metrics/base.js';
 import { type APIConnectOptions } from '../types.js';
 import type { AudioBuffer } from '../utils.js';
 import { AsyncIterableQueue } from '../utils.js';
+import type { TimedString } from '../voice/index.js';
 /** Indicates start/middle/end of speech */
 export declare enum SpeechEventType {
     /**
@@ -44,6 +45,7 @@ export interface SpeechData {
     startTime: number;
     endTime: number;
     confidence: number;
+    words?: TimedString[];
 }
 export interface RecognitionUsage {
     audioDuration: number;
@@ -64,6 +66,13 @@ export interface SpeechEvent {
 export interface STTCapabilities {
     streaming: boolean;
     interimResults: boolean;
+    /**
+     * Whether this STT supports aligned transcripts with word/chunk timestamps.
+     * - 'word': Provider returns word-level timestamps
+     * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)
+     * - false: Provider does not support aligned transcripts
+     */
+    alignedTranscript?: 'word' | 'chunk' | false;
 }
 export interface STTError {
     type: 'stt_error';
@@ -133,6 +142,7 @@ export declare abstract class SpeechStream implements AsyncIterableIterator<Spee
     private deferredInputStream;
     private logger;
     private _connOptions;
+    private _startTimeOffset;
     protected abortController: AbortController;
     constructor(stt: STT, sampleRate?: number, connectionOptions?: APIConnectOptions);
     private mainTask;
@@ -141,6 +151,8 @@ export declare abstract class SpeechStream implements AsyncIterableIterator<Spee
     protected monitorMetrics(): Promise<void>;
     protected abstract run(): Promise<void>;
     protected get abortSignal(): AbortSignal;
+    get startTimeOffset(): number;
+    set startTimeOffset(value: number);
     updateInputStream(audioStream: ReadableStream<AudioFrame>): void;
     detachInputStream(): void;
     /** Push an audio frame to the STT */