@livekit/agents-plugin-assemblyai 0.0.0 → 1.2.7

package/dist/stt.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioBuffer, AudioByteStream, log, stt } from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { AssemblyAI } from 'assemblyai';\nimport type { RealtimeTranscriber, RealtimeTranscript } from 'assemblyai';\n\nexport interface STTOptions {\n  apiKey?: string;\n  interimResults: boolean;\n  sampleRate: number;\n  keywords: [string, number][];\n  endUtteranceSilenceThreshold?: number;\n}\n\nconst defaultSTTOptions: STTOptions = {\n  apiKey: process.env.ASSEMBLY_AI_KEY,\n  interimResults: true,\n  sampleRate: 16000,\n  keywords: [],\n  // NOTE:\n  // The default is 700ms from AssemblyAI.\n  // We use a low default of 300ms here because we also use\n  // the new end-of-utterance model from LiveKit to handle\n  // turn detection in my agent. Which means that even though\n  // this will quickly return a final transcript EVEN THOUGH\n  // USER IS NOT DONE SPEAKING, the EOU model from LiveKit\n  // DOES properly differentiate and doesn't interrupt (magically!)\n  // Ref: https://blog.livekit.io/using-a-transformer-to-improve-end-of-turn-detection/\n  endUtteranceSilenceThreshold: 200,\n};\n\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n  #logger = log();\n  label = 'assemblyai.STT';\n\n  constructor(opts: Partial<STTOptions> = defaultSTTOptions) {\n    super({\n      streaming: true,\n      interimResults: opts.interimResults ?? defaultSTTOptions.interimResults,\n    });\n    if (opts.apiKey === undefined && defaultSTTOptions.apiKey === undefined) {\n      throw new Error(\n        'AssemblyAI API key is required, whether as an argument or as $ASSEMBLY_AI_KEY',\n      );\n    }\n\n    this.#opts = { ...defaultSTTOptions, ...opts };\n  }\n\n  // eslint-disable-next-line @typescript-eslint/no-unused-vars\n  async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n    throw new Error('Recognize is not supported on AssemblyAI STT');\n  }\n\n  stream(): stt.SpeechStream {\n    return new SpeechStream(this, this.#opts);\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #speaking = false;\n  #client: AssemblyAI;\n  #transcriber?: RealtimeTranscriber;\n  label = 'assemblyai.SpeechStream';\n\n  constructor(stt: STT, opts: STTOptions) {\n    super(stt);\n    this.#opts = opts;\n    this.closed = false;\n    this.#client = new AssemblyAI({\n      apiKey: this.#opts.apiKey || '',\n    });\n\n    this.#run();\n  }\n\n  async #run() {\n    try {\n      this.#transcriber = this.#client.realtime.transcriber({\n        sampleRate: this.#opts.sampleRate,\n        wordBoost: this.#opts.keywords.map((k) => k[0]),\n        endUtteranceSilenceThreshold: this.#opts.endUtteranceSilenceThreshold,\n      });\n\n      this.#transcriber.on('open', (data) => {\n        this.#logger\n          .child({ sessionId: data.sessionId, expiresAt: data.expiresAt })\n          .debug(`AssemblyAI session opened`);\n      });\n\n      this.#transcriber.on('close', (code, reason) => {\n        this.#logger.child({ code, reason }).debug(`AssemblyAI session closed`);\n        if (!this.closed) {\n          this.#run();\n        }\n      });\n\n      this.#transcriber.on('error', (error) => {\n        this.#logger.child({ error: error.message }).error(`AssemblyAI error`);\n      });\n\n      this.#transcriber.on('transcript', (transcript) => {\n        if (this.closed) return;\n\n        if (!transcript.text || transcript.text.trim() === '') {\n          return;\n        }\n\n        if (!this.#speaking) {\n          this.#speaking = true;\n          this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n        }\n\n        if (transcript.message_type === 'PartialTranscript') {\n          this.queue.put({\n            type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n            alternatives: [assemblyTranscriptToSpeechData(transcript)],\n          });\n        } else if (transcript.message_type === 'FinalTranscript') {\n          this.queue.put({\n            type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n            alternatives: [assemblyTranscriptToSpeechData(transcript)],\n          });\n        }\n      });\n\n      await this.#transcriber.connect();\n\n      const sendTask = async () => {\n        const samples100Ms = Math.floor(this.#opts.sampleRate / 10);\n        const stream = new AudioByteStream(this.#opts.sampleRate, 1, samples100Ms);\n\n        for await (const data of this.input) {\n          if (this.closed) break;\n\n          let frames: AudioFrame[];\n          if (data === SpeechStream.FLUSH_SENTINEL) {\n            frames = stream.flush();\n          } else if (data.sampleRate === this.#opts.sampleRate) {\n            frames = stream.write(data.data.buffer);\n          } else {\n            throw new Error(`Sample rate or channel count of frame does not match`);\n          }\n\n          for await (const frame of frames) {\n            this.#transcriber?.sendAudio(new Uint8Array(frame.data.buffer));\n          }\n        }\n\n        if (this.#transcriber) {\n          await this.#transcriber.close();\n        }\n      };\n\n      await sendTask();\n    } catch (error: unknown) {\n      this.#logger.child({ error }).error(`Error in AssemblyAI STT`);\n\n      if (!this.closed) {\n        setTimeout(() => this.#run(), 5000);\n      }\n    }\n  }\n}\n\nconst assemblyTranscriptToSpeechData = (transcript: RealtimeTranscript): stt.SpeechData => {\n  return {\n    language: 'en-US',\n    startTime: transcript.audio_start || 0,\n    endTime: transcript.audio_end || 0,\n    confidence: transcript.confidence || 1.0,\n    text: transcript.text || '',\n  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+{"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n//\nimport {\n  type APIConnectOptions,\n  type AudioBuffer,\n  AudioByteStream,\n  Future,\n  Task,\n  createTimedString,\n  delay,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { RawData } from 'ws';\nimport { WebSocket } from 'ws';\nimport type { STTEncoding, STTModels } from './models.js';\n\n// AssemblyAI Universal-Streaming (v3) message envelope. All fields are optional\n// since we narrow on `type` before reading anything else.\ninterface StreamEventMessage {\n  type?: 'Begin' | 'SpeechStarted' | 'Turn' | 'Termination' | string;\n  // Begin\n  id?: string;\n  expires_at?: number;\n  // Turn\n  transcript?: string;\n  utterance?: string;\n  end_of_turn?: boolean;\n  end_of_turn_confidence?: number;\n  turn_is_formatted?: boolean;\n  language_code?: string;\n  speaker_label?: string;\n  words?: Array<{\n    text?: string;\n    start?: number;\n    end?: number;\n    confidence?: number;\n    speaker?: string;\n  }>;\n  // Termination\n  audio_duration_seconds?: number;\n  session_duration_seconds?: number;\n}\n\nexport interface STTOptions {\n  apiKey?: string;\n  sampleRate: number;\n  /**\n   * How large each chunk of audio is before being sent to AssemblyAI, in\n   * milliseconds. Corresponds to Python's `buffer_size_seconds` (seconds there,\n   * ms here per this repo's time-unit convention).\n   */\n  bufferSizeMs: number;\n  encoding: STTEncoding;\n  speechModel: STTModels;\n  languageDetection?: boolean;\n  endOfTurnConfidenceThreshold?: number;\n  /** Minimum silence (ms) before a confident end-of-turn is finalized. */\n  minTurnSilence?: number;\n  /** Maximum silence (ms) before end-of-turn is forced regardless of confidence. */\n  maxTurnSilence?: number;\n  formatTurns?: boolean;\n  keytermsPrompt?: string[];\n  /** Only supported with the `u3-rt-pro` model. */\n  prompt?: string;\n  vadThreshold?: number;\n  /**\n   * Enable speaker diarization. Note: AssemblyAI will return per-word speaker\n   * labels, but the JS framework's `stt.SpeechData` type does not yet expose\n   * a `speakerId` field (unlike the Python framework), so the labels are not\n   * currently surfaced on emitted events. Setting this to `true` still has\n   * effect server-side. Once the base `SpeechData` interface gains speaker\n   * support, `#processStreamEvent` should forward `data.words[].speaker` too.\n   */\n  speakerLabels?: boolean;\n  maxSpeakers?: number;\n  domain?: string;\n  baseUrl: string;\n}\n\nconst defaultSTTOptions: STTOptions = {\n  apiKey: process.env.ASSEMBLYAI_API_KEY,\n  sampleRate: 16000,\n  bufferSizeMs: 50,\n  encoding: 'pcm_s16le',\n  speechModel: 'universal-streaming-english',\n  baseUrl: 'wss://streaming.assemblyai.com',\n};\n\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n  #streams = new Set<WeakRef<SpeechStream>>();\n  label = 'assemblyai.STT';\n\n  get model(): string {\n    return this.#opts.speechModel;\n  }\n\n  get provider(): string {\n    return 'AssemblyAI';\n  }\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: 'word',\n    });\n\n    if (opts.speechModel === 'u3-pro') {\n      log().warn(\"'u3-pro' is deprecated, use 'u3-rt-pro' instead.\");\n      opts.speechModel = 'u3-rt-pro';\n    }\n\n    if (opts.prompt !== undefined && opts.speechModel !== 'u3-rt-pro') {\n      throw new Error(\"The 'prompt' parameter is only supported with the 'u3-rt-pro' model.\");\n    }\n\n    const apiKey = opts.apiKey ?? defaultSTTOptions.apiKey;\n    if (!apiKey) {\n      throw new Error(\n        'AssemblyAI API key is required. Pass one in via the `apiKey` parameter, or set it as the `ASSEMBLYAI_API_KEY` environment variable',\n      );\n    }\n\n    // Minimize latency; matches LK's end-of-turn detector well.\n    const minTurnSilence = opts.minTurnSilence ?? 100;\n\n    this.#opts = {\n      ...defaultSTTOptions,\n      ...opts,\n      apiKey,\n      minTurnSilence,\n    };\n  }\n\n  // eslint-disable-next-line @typescript-eslint/no-unused-vars\n  async _recognize(_: AudioBuffer): Promise<stt.SpeechEvent> {\n    throw new Error('Non-streaming recognize is not supported on AssemblyAI STT');\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n    for (const ref of this.#streams) {\n      const stream = ref.deref();\n      if (stream) {\n        stream.updateOptions(opts);\n      } else {\n        this.#streams.delete(ref);\n      }\n    }\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    const stream = new SpeechStream(this, this.#opts, options?.connOptions);\n    this.#streams.add(new WeakRef(stream));\n    return stream;\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  static readonly CLOSE_MSG = JSON.stringify({ type: 'Terminate' });\n\n  #opts: STTOptions;\n  #logger = log();\n  #speechDurationInS = 0;\n  #lastPreflightStartTime = 0;\n  #pendingConfigMessages: Record<string, unknown>[] = [];\n  #configMessagePending = new Future();\n  #sessionId: string | null = null;\n  #expiresAt: number | null = null;\n  label = 'assemblyai.SpeechStream';\n\n  constructor(stt: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(stt, opts.sampleRate, connOptions);\n    this.#opts = opts;\n    this.closed = false;\n  }\n\n  /**\n   * The AssemblyAI session ID. Set when the WebSocket connection is established\n   * (before any speech events). Null until the connection completes.\n   * Share this with the AssemblyAI team when reporting issues.\n   */\n  get sessionId(): string | null {\n    return this.#sessionId;\n  }\n\n  /**\n   * Unix timestamp when the AssemblyAI session expires. Set alongside\n   * {@link sessionId} when the WebSocket connection is established.\n   */\n  get expiresAt(): number | null {\n    return this.#expiresAt;\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n\n    const configMsg: Record<string, unknown> = { type: 'UpdateConfiguration' };\n    if (opts.prompt !== undefined) configMsg.prompt = opts.prompt;\n    if (opts.keytermsPrompt !== undefined) configMsg.keyterms_prompt = opts.keytermsPrompt;\n    if (opts.maxTurnSilence !== undefined) configMsg.max_turn_silence = opts.maxTurnSilence;\n    if (opts.minTurnSilence !== undefined) configMsg.min_turn_silence = opts.minTurnSilence;\n    if (opts.endOfTurnConfidenceThreshold !== undefined) {\n      configMsg.end_of_turn_confidence_threshold = opts.endOfTurnConfidenceThreshold;\n    }\n    if (opts.vadThreshold !== undefined) configMsg.vad_threshold = opts.vadThreshold;\n\n    // Only send if any actual fields (besides `type`) were specified.\n    if (Object.keys(configMsg).length > 1) {\n      this.#pendingConfigMessages.push(configMsg);\n      if (!this.#configMessagePending.done) this.#configMessagePending.resolve();\n    }\n  }\n\n  /**\n   * Force-finalize the current turn immediately.\n   */\n  forceEndpoint() {\n    this.#pendingConfigMessages.push({ type: 'ForceEndpoint' });\n    if (!this.#configMessagePending.done) this.#configMessagePending.resolve();\n  }\n\n  // Deepgram-style reconnect loop around a single websocket lifetime.\n  protected async run() {\n    const maxRetry = 32;\n    let retries = 0;\n\n    while (!this.input.closed && !this.closed) {\n      try {\n        const ws = await this.#connectWS();\n        await this.#runWS(ws);\n        retries = 0;\n      } catch (e) {\n        if (!this.closed && !this.input.closed) {\n          if (retries >= maxRetry) {\n            throw new Error(`failed to connect to AssemblyAI after ${retries} attempts: ${e}`);\n          }\n\n          const retryDelaySeconds = Math.min(retries * 5, 10);\n          retries++;\n\n          this.#logger.warn(\n            `failed to connect to AssemblyAI, retrying in ${retryDelaySeconds} seconds: ${e} (${retries}/${maxRetry})`,\n          );\n          await delay(retryDelaySeconds * 1000);\n        } else {\n          this.#logger.warn(\n            `AssemblyAI disconnected, connection is closed: ${e} (inputClosed: ${this.input.closed}, isClosed: ${this.closed})`,\n          );\n        }\n      }\n    }\n\n    this.closed = true;\n  }\n\n  async #connectWS(): Promise<WebSocket> {\n    // u3-rt-pro has different silence defaults — if unset, both min and max default to 100ms.\n    let minSilence = this.#opts.minTurnSilence;\n    let maxSilence = this.#opts.maxTurnSilence;\n    if (this.#opts.speechModel === 'u3-rt-pro') {\n      if (minSilence === undefined) minSilence = 100;\n      if (maxSilence === undefined) maxSilence = minSilence;\n    }\n\n    // Default language_detection to true for multilingual / u3-rt-pro models, false otherwise.\n    const defaultLanguageDetection =\n      this.#opts.speechModel.includes('multilingual') || this.#opts.speechModel === 'u3-rt-pro';\n    const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;\n\n    const liveConfig: Record<string, unknown> = {\n      sample_rate: this.#opts.sampleRate,\n      encoding: this.#opts.encoding,\n      speech_model: this.#opts.speechModel,\n      format_turns: this.#opts.formatTurns,\n      end_of_turn_confidence_threshold: this.#opts.endOfTurnConfidenceThreshold,\n      min_turn_silence: minSilence,\n      max_turn_silence: maxSilence,\n      keyterms_prompt:\n        this.#opts.keytermsPrompt !== undefined\n          ? JSON.stringify(this.#opts.keytermsPrompt)\n          : undefined,\n      language_detection: languageDetection,\n      prompt: this.#opts.prompt,\n      vad_threshold: this.#opts.vadThreshold,\n      speaker_labels: this.#opts.speakerLabels,\n      max_speakers: this.#opts.maxSpeakers,\n      domain: this.#opts.domain,\n    };\n\n    const url = new URL(`${this.#opts.baseUrl}/v3/ws`);\n    // Python serializes booleans as the strings \"true\"/\"false\", so we mirror that.\n    for (const [key, value] of Object.entries(liveConfig)) {\n      if (value === undefined || value === null) continue;\n      if (typeof value === 'boolean') {\n        url.searchParams.append(key, value ? 'true' : 'false');\n      } else {\n        url.searchParams.append(key, String(value));\n      }\n    }\n\n    const ws = new WebSocket(url, {\n      headers: {\n        Authorization: this.#opts.apiKey!,\n        'Content-Type': 'application/json',\n        'User-Agent': 'AssemblyAI/1.0 (integration=Livekit)',\n      },\n    });\n\n    await new Promise<void>((resolve, reject) => {\n      ws.on('open', () => resolve());\n      ws.on('error', (error) => reject(error));\n      ws.on('close', (code) => reject(new Error(`WebSocket returned ${code}`)));\n    });\n\n    return ws;\n  }\n\n  async #runWS(ws: WebSocket) {\n    let closing = false;\n    const sessionController = new AbortController();\n\n    // gets cancelled also when sendTask is complete\n    const wsMonitor = Task.from(async (controller) => {\n      const closed = new Promise<void>((_, reject) => {\n        ws.once('close', (code, reason) => {\n          if (!closing) {\n            this.#logger.error(`WebSocket closed with code ${code}: ${reason}`);\n            reject(new Error('WebSocket closed'));\n          }\n        });\n      });\n\n      await Promise.race([closed, waitForAbort(controller.signal)]);\n    });\n\n    const sendTask = async () => {\n      const samplesPerBuffer = Math.floor((this.#opts.sampleRate * this.#opts.bufferSizeMs) / 1000);\n      const audioStream = new AudioByteStream(this.#opts.sampleRate, 1, samplesPerBuffer);\n\n      const abortPromise = waitForAbort(this.abortSignal);\n      const sessionAbort = waitForAbort(sessionController.signal);\n\n      try {\n        while (!this.closed) {\n          const result = await Promise.race([this.input.next(), abortPromise, sessionAbort]);\n\n          if (result === undefined) return; // aborted\n          if (result.done) break;\n\n          const data = result.value;\n\n          let frames: AudioFrame[];\n          if (data === SpeechStream.FLUSH_SENTINEL) {\n            frames = audioStream.flush();\n          } else if (data.sampleRate === this.#opts.sampleRate && data.channels === 1) {\n            // AssemblyAI expects mono PCM. 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package/dist/stt.d.cts

@@ -1,21 +1,73 @@
-import { type AudioBuffer, stt } from '@livekit/agents';
+import { type APIConnectOptions, type AudioBuffer, stt } from '@livekit/agents';
+import type { STTEncoding, STTModels } from './models.js';
 export interface STTOptions {
     apiKey?: string;
-    interimResults: boolean;
     sampleRate: number;
-    keywords: [string, number][];
-    endUtteranceSilenceThreshold?: number;
+    /**
+     * How large each chunk of audio is before being sent to AssemblyAI, in
+     * milliseconds. Corresponds to Python's `buffer_size_seconds` (seconds there,
+     * ms here per this repo's time-unit convention).
+     */
+    bufferSizeMs: number;
+    encoding: STTEncoding;
+    speechModel: STTModels;
+    languageDetection?: boolean;
+    endOfTurnConfidenceThreshold?: number;
+    /** Minimum silence (ms) before a confident end-of-turn is finalized. */
+    minTurnSilence?: number;
+    /** Maximum silence (ms) before end-of-turn is forced regardless of confidence. */
+    maxTurnSilence?: number;
+    formatTurns?: boolean;
+    keytermsPrompt?: string[];
+    /** Only supported with the `u3-rt-pro` model. */
+    prompt?: string;
+    vadThreshold?: number;
+    /**
+     * Enable speaker diarization. Note: AssemblyAI will return per-word speaker
+     * labels, but the JS framework's `stt.SpeechData` type does not yet expose
+     * a `speakerId` field (unlike the Python framework), so the labels are not
+     * currently surfaced on emitted events. Setting this to `true` still has
+     * effect server-side. Once the base `SpeechData` interface gains speaker
+     * support, `#processStreamEvent` should forward `data.words[].speaker` too.
+     */
+    speakerLabels?: boolean;
+    maxSpeakers?: number;
+    domain?: string;
+    baseUrl: string;
 }
 export declare class STT extends stt.STT {
     #private;
     label: string;
+    get model(): string;
+    get provider(): string;
     constructor(opts?: Partial<STTOptions>);
     _recognize(_: AudioBuffer): Promise<stt.SpeechEvent>;
-    stream(): stt.SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
 }
 export declare class SpeechStream extends stt.SpeechStream {
     #private;
+    static readonly CLOSE_MSG: string;
     label: string;
-    constructor(stt: STT, opts: STTOptions);
+    constructor(stt: STT, opts: STTOptions, connOptions?: APIConnectOptions);
+    /**
+     * The AssemblyAI session ID. Set when the WebSocket connection is established
+     * (before any speech events). Null until the connection completes.
+     * Share this with the AssemblyAI team when reporting issues.
+     */
+    get sessionId(): string | null;
+    /**
+     * Unix timestamp when the AssemblyAI session expires. Set alongside
+     * {@link sessionId} when the WebSocket connection is established.
+     */
+    get expiresAt(): number | null;
+    updateOptions(opts: Partial<STTOptions>): void;
+    /**
+     * Force-finalize the current turn immediately.
+     */
+    forceEndpoint(): void;
+    protected run(): Promise<void>;
 }
 //# sourceMappingURL=stt.d.ts.map

package/dist/stt.d.ts

@@ -1,21 +1,73 @@
-import { type AudioBuffer, stt } from '@livekit/agents';
+import { type APIConnectOptions, type AudioBuffer, stt } from '@livekit/agents';
+import type { STTEncoding, STTModels } from './models.js';
 export interface STTOptions {
     apiKey?: string;
-    interimResults: boolean;
     sampleRate: number;
-    keywords: [string, number][];
-    endUtteranceSilenceThreshold?: number;
+    /**
+     * How large each chunk of audio is before being sent to AssemblyAI, in
+     * milliseconds. Corresponds to Python's `buffer_size_seconds` (seconds there,
+     * ms here per this repo's time-unit convention).
+     */
+    bufferSizeMs: number;
+    encoding: STTEncoding;
+    speechModel: STTModels;
+    languageDetection?: boolean;
+    endOfTurnConfidenceThreshold?: number;
+    /** Minimum silence (ms) before a confident end-of-turn is finalized. */
+    minTurnSilence?: number;
+    /** Maximum silence (ms) before end-of-turn is forced regardless of confidence. */
+    maxTurnSilence?: number;
+    formatTurns?: boolean;
+    keytermsPrompt?: string[];
+    /** Only supported with the `u3-rt-pro` model. */
+    prompt?: string;
+    vadThreshold?: number;
+    /**
+     * Enable speaker diarization. Note: AssemblyAI will return per-word speaker
+     * labels, but the JS framework's `stt.SpeechData` type does not yet expose
+     * a `speakerId` field (unlike the Python framework), so the labels are not
+     * currently surfaced on emitted events. Setting this to `true` still has
+     * effect server-side. Once the base `SpeechData` interface gains speaker
+     * support, `#processStreamEvent` should forward `data.words[].speaker` too.
+     */
+    speakerLabels?: boolean;
+    maxSpeakers?: number;
+    domain?: string;
+    baseUrl: string;
 }
 export declare class STT extends stt.STT {
     #private;
     label: string;
+    get model(): string;
+    get provider(): string;
     constructor(opts?: Partial<STTOptions>);
     _recognize(_: AudioBuffer): Promise<stt.SpeechEvent>;
-    stream(): stt.SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
 }
 export declare class SpeechStream extends stt.SpeechStream {
     #private;
+    static readonly CLOSE_MSG: string;
     label: string;
-    constructor(stt: STT, opts: STTOptions);
+    constructor(stt: STT, opts: STTOptions, connOptions?: APIConnectOptions);
+    /**
+     * The AssemblyAI session ID. Set when the WebSocket connection is established
+     * (before any speech events). Null until the connection completes.
+     * Share this with the AssemblyAI team when reporting issues.
+     */
+    get sessionId(): string | null;
+    /**
+     * Unix timestamp when the AssemblyAI session expires. Set alongside
+     * {@link sessionId} when the WebSocket connection is established.
+     */
+    get expiresAt(): number | null;
+    updateOptions(opts: Partial<STTOptions>): void;
+    /**
+     * Force-finalize the current turn immediately.
+     */
+    forceEndpoint(): void;
+    protected run(): Promise<void>;
 }
 //# sourceMappingURL=stt.d.ts.map

package/dist/stt.d.ts.map

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