@livekit/agents-plugin-assemblyai 1.4.6 → 1.4.8

package/dist/index.cjs

@@ -22,7 +22,7 @@ class AssemblyAIPlugin extends import_agents.Plugin {
   constructor() {
     super({
       title: "assemblyai",
-      version: "1.4.6",
+      version: "1.4.8",
       package: "@livekit/agents-plugin-assemblyai"
     });
   }

package/dist/index.js

@@ -5,7 +5,7 @@ class AssemblyAIPlugin extends Plugin {
   constructor() {
     super({
       title: "assemblyai",
-      version: "1.4.6",
+      version: "1.4.8",
       package: "@livekit/agents-plugin-assemblyai"
     });
   }

package/dist/models.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\nexport type STTModels =\n  | 'universal-streaming-english'\n  | 'universal-streaming-multilingual'\n  | 'u3-rt-pro'\n  // Deprecated alias — AssemblyAI maps this to `u3-rt-pro` server-side, but the\n  // Python plugin emits a warning and rewrites it. Kept here so TS users don't\n  // break if they already pass it.\n  | 'u3-pro';\n\nexport type STTEncoding = 'pcm_s16le' | 'pcm_mulaw';\n"],"mappings":";;;;;;;;;;;;;;AAAA;AAAA;","names":[]}
+{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\nexport type STTModels =\n  | 'universal-streaming-english'\n  | 'universal-streaming-multilingual'\n  | 'u3-rt-pro'\n  | 'u3-rt-pro-beta-1'\n  | 'universal-3-5-pro'\n  // Deprecated alias — AssemblyAI maps this to `u3-rt-pro` server-side, but the\n  // Python plugin emits a warning and rewrites it. Kept here so TS users don't\n  // break if they already pass it.\n  | 'u3-pro';\n\nexport type STTEncoding = 'pcm_s16le' | 'pcm_mulaw';\n\nexport type VoiceFocus = 'near-field' | 'far-field';\n"],"mappings":";;;;;;;;;;;;;;AAAA;AAAA;","names":[]}

package/dist/models.d.cts

@@ -1,3 +1,4 @@
-export type STTModels = 'universal-streaming-english' | 'universal-streaming-multilingual' | 'u3-rt-pro' | 'u3-pro';
+export type STTModels = 'universal-streaming-english' | 'universal-streaming-multilingual' | 'u3-rt-pro' | 'u3-rt-pro-beta-1' | 'universal-3-5-pro' | 'u3-pro';
 export type STTEncoding = 'pcm_s16le' | 'pcm_mulaw';
+export type VoiceFocus = 'near-field' | 'far-field';
 //# sourceMappingURL=models.d.ts.map

package/dist/models.d.ts

@@ -1,3 +1,4 @@
-export type STTModels = 'universal-streaming-english' | 'universal-streaming-multilingual' | 'u3-rt-pro' | 'u3-pro';
+export type STTModels = 'universal-streaming-english' | 'universal-streaming-multilingual' | 'u3-rt-pro' | 'u3-rt-pro-beta-1' | 'universal-3-5-pro' | 'u3-pro';
 export type STTEncoding = 'pcm_s16le' | 'pcm_mulaw';
+export type VoiceFocus = 'near-field' | 'far-field';
 //# sourceMappingURL=models.d.ts.map

package/dist/models.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"models.d.ts","sourceRoot":"","sources":["../src/models.ts"],"names":[],"mappings":"AAIA,MAAM,MAAM,SAAS,GACjB,6BAA6B,GAC7B,kCAAkC,GAClC,WAAW,GAIX,QAAQ,CAAC;AAEb,MAAM,MAAM,WAAW,GAAG,WAAW,GAAG,WAAW,CAAC"}
+{"version":3,"file":"models.d.ts","sourceRoot":"","sources":["../src/models.ts"],"names":[],"mappings":"AAIA,MAAM,MAAM,SAAS,GACjB,6BAA6B,GAC7B,kCAAkC,GAClC,WAAW,GACX,kBAAkB,GAClB,mBAAmB,GAInB,QAAQ,CAAC;AAEb,MAAM,MAAM,WAAW,GAAG,WAAW,GAAG,WAAW,CAAC;AAEpD,MAAM,MAAM,UAAU,GAAG,YAAY,GAAG,WAAW,CAAC"}

package/dist/stt.cjs

@@ -24,12 +24,16 @@ __export(stt_exports, {
 module.exports = __toCommonJS(stt_exports);
 var import_agents = require("@livekit/agents");
 var import_ws = require("ws");
+const U3_PRO_MODELS = ["u3-rt-pro", "u3-rt-pro-beta-1", "universal-3-5-pro"];
+function isU3ProModel(model) {
+  return U3_PRO_MODELS.includes(model);
+}
 const defaultSTTOptions = {
   apiKey: process.env.ASSEMBLYAI_API_KEY,
   sampleRate: 16e3,
   bufferSizeMs: 50,
   encoding: "pcm_s16le",
-  speechModel: "universal-streaming-english",
+  speechModel: "universal-3-5-pro",
   baseUrl: "wss://streaming.assemblyai.com"
 };
 class STT extends import_agents.stt.STT {
@@ -52,8 +56,15 @@ class STT extends import_agents.stt.STT {
       (0, import_agents.log)().warn("'u3-pro' is deprecated, use 'u3-rt-pro' instead.");
       opts.speechModel = "u3-rt-pro";
     }
-    if (opts.prompt !== void 0 && opts.speechModel !== "u3-rt-pro") {
-      throw new Error("The 'prompt' parameter is only supported with the 'u3-rt-pro' model.");
+    const speechModel = opts.speechModel ?? defaultSTTOptions.speechModel;
+    if (!isU3ProModel(speechModel)) {
+      for (const param of ["prompt", "voiceFocus", "voiceFocusThreshold", "mode"]) {
+        if (opts[param] !== void 0) {
+          throw new Error(
+            `The '${param}' parameter is only supported with the ${U3_PRO_MODELS.join(", ")} models.`
+          );
+        }
+      }
     }
     const apiKey = opts.apiKey ?? defaultSTTOptions.apiKey;
     if (!apiKey) {
@@ -176,11 +187,11 @@ class SpeechStream extends import_agents.stt.SpeechStream {
   async #connectWS() {
     let minSilence = this.#opts.minTurnSilence;
     let maxSilence = this.#opts.maxTurnSilence;
-    if (this.#opts.speechModel === "u3-rt-pro") {
+    if (isU3ProModel(this.#opts.speechModel)) {
       if (minSilence === void 0) minSilence = 100;
       if (maxSilence === void 0) maxSilence = minSilence;
     }
-    const defaultLanguageDetection = this.#opts.speechModel.includes("multilingual") || this.#opts.speechModel === "u3-rt-pro";
+    const defaultLanguageDetection = this.#opts.speechModel.includes("multilingual") || isU3ProModel(this.#opts.speechModel);
     const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;
     const liveConfig = {
       sample_rate: this.#opts.sampleRate,
@@ -196,7 +207,10 @@ class SpeechStream extends import_agents.stt.SpeechStream {
       vad_threshold: this.#opts.vadThreshold,
       speaker_labels: this.#opts.speakerLabels,
       max_speakers: this.#opts.maxSpeakers,
-      domain: this.#opts.domain
+      domain: this.#opts.domain,
+      voice_focus: this.#opts.voiceFocus,
+      voice_focus_threshold: this.#opts.voiceFocusThreshold,
+      mode: this.#opts.mode
     };
     const url = new URL(`${this.#opts.baseUrl}/v3/ws`);
     for (const [key, value] of Object.entries(liveConfig)) {

package/dist/stt.cjs.map

@@ -1 +1 @@
-{"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. The base SpeechStream only resamples\n            // sample rate, so reject any frame that is not already downmixed.\n            frames = audioStream.write(data.data.buffer as ArrayBuffer);\n          } else {\n            throw new Error('sample rate or channel count of frame does not match');\n          }\n\n          for (const frame of frames) {\n            this.#speechDurationInS += frame.samplesPerChannel / frame.sampleRate;\n            ws.send(frame.data.buffer);\n          }\n        }\n      } finally {\n        closing = true;\n        try {\n          ws.send(SpeechStream.CLOSE_MSG);\n        } catch {\n          // ignore — socket may already be closing\n        }\n        wsMonitor.cancel();\n      }\n    };\n\n    let messageHandler: ((msg: RawData, isBinary: boolean) => void) | null = null;\n    const listenTask = Task.from(async (controller) => {\n      const listenMessage = new Promise<void>((resolve, reject) => {\n        messageHandler = (msg, isBinary) => {\n          if (isBinary) {\n            this.#logger.error('unexpected binary message from AssemblyAI');\n            return;\n          }\n          try {\n            const json = JSON.parse(msg.toString()) as StreamEventMessage;\n            this.#processStreamEvent(json);\n            if (this.closed || closing) {\n              resolve();\n            }\n          } catch (err) {\n            this.#logger.error(`AssemblyAI: error processing message: ${msg}`);\n            reject(err);\n          }\n        };\n        ws.on('message', messageHandler);\n      });\n\n      await Promise.race([listenMessage, waitForAbort(controller.signal)]);\n    });\n\n    const configTask = Task.from(async (controller) => {\n      // Drain any messages queued while the socket was reconnecting.\n      while (this.#pendingConfigMessages.length > 0) {\n        const msg = this.#pendingConfigMessages.shift()!;\n        ws.send(JSON.stringify(msg));\n      }\n\n      while (!controller.signal.aborted) {\n        await Promise.race([this.#configMessagePending.await, waitForAbort(controller.signal)]);\n        if (controller.signal.aborted) return;\n\n        this.#configMessagePending = new Future();\n        while (this.#pendingConfigMessages.length > 0) {\n          const msg = this.#pendingConfigMessages.shift()!;\n          ws.send(JSON.stringify(msg));\n        }\n      }\n    });\n\n    try {\n      await Promise.all([sendTask(), listenTask.result, wsMonitor.result]);\n    } finally {\n      closing = true;\n      sessionController.abort();\n      listenTask.cancel();\n      configTask.cancel();\n      if (messageHandler) ws.off('message', messageHandler);\n      try {\n        ws.close();\n      } catch {\n        // ignore\n      }\n    }\n  }\n\n  #averageConfidence(words: Array<{ confidence?: number }>): number {\n    if (words.length === 0) return 0;\n    return words.reduce((sum, w) => sum + (w.confidence ?? 0), 0) / words.length;\n  }\n\n  #processStreamEvent(data: StreamEventMessage) {\n    const messageType = data.type;\n\n    if (messageType === 'Begin') {\n      this.#sessionId = data.id ?? null;\n      this.#expiresAt = data.expires_at ?? null;\n      this.#logger.info(\n        `AssemblyAI session started id=${this.#sessionId} expires_at=${this.#expiresAt}`,\n      );\n      return;\n    }\n\n    if (messageType === 'SpeechStarted') {\n      this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n      return;\n    }\n\n    if (messageType === 'Termination') {\n      this.#logger.debug(\n        `AssemblyAI session terminated audio_duration=${data.audio_duration_seconds}s session_duration=${data.session_duration_seconds}s`,\n      );\n      return;\n    }\n\n    if (messageType !== 'Turn') {\n      return;\n    }\n\n    const words = data.words ?? [];\n    const endOfTurn = Boolean(data.end_of_turn);\n    const turnIsFormatted = Boolean(data.turn_is_formatted);\n    const utterance = data.utterance ?? '';\n    const transcript = data.transcript ?? '';\n    const language = normalizeLanguage(data.language_code ?? 'en');\n\n    // Word timestamps are in milliseconds:\n    // https://www.assemblyai.com/docs/api-reference/streaming-api/streaming-api#receive.receiveTurn.words\n    const timedWords = words.map((word) =>\n      createTimedString({\n        text: word.text ?? '',\n        startTime: (word.start ?? 0) / 1000 + this.startTimeOffset,\n        endTime: (word.end ?? 0) / 1000 + this.startTimeOffset,\n        confidence: word.confidence ?? 0,\n        startTimeOffset: this.startTimeOffset,\n      }),\n    );\n\n    let startTime = 0;\n    let endTime = 0;\n    let confidence = 0;\n\n    // `words` are cumulative for the turn — emit as an interim transcript.\n    if (timedWords.length > 0) {\n      const interimText = timedWords.map((w) => w.text).join(' ');\n      startTime = timedWords[0]!.startTime ?? 0;\n      endTime = timedWords[timedWords.length - 1]!.endTime ?? 0;\n      confidence = this.#averageConfidence(timedWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: interimText,\n            startTime,\n            endTime,\n            confidence,\n            words: timedWords,\n          },\n        ],\n      });\n    }\n\n    // `utterance` is chunk-based (not cumulative) — emit as a preflight transcript\n    // covering only the words since the last preflight.\n    if (utterance) {\n      if (this.#lastPreflightStartTime === 0) {\n        this.#lastPreflightStartTime = startTime;\n      }\n\n      const utteranceWords = timedWords.filter(\n        (w) => w.startTime !== undefined && w.startTime >= this.#lastPreflightStartTime,\n      );\n      const utteranceConfidence = this.#averageConfidence(utteranceWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.PREFLIGHT_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: utterance,\n            startTime: this.#lastPreflightStartTime,\n            endTime,\n            confidence: utteranceConfidence,\n            words: utteranceWords,\n          },\n        ],\n      });\n      this.#lastPreflightStartTime = endTime;\n    }\n\n    // End-of-turn: emit FINAL_TRANSCRIPT + END_OF_SPEECH.\n    // If the 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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, VoiceFocus } from './models.js';\n\nconst U3_PRO_MODELS = ['u3-rt-pro', 'u3-rt-pro-beta-1', 'universal-3-5-pro'] as const;\n\nfunction isU3ProModel(model: STTModels): boolean {\n  return U3_PRO_MODELS.includes(model as (typeof U3_PRO_MODELS)[number]);\n}\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  /** Isolate the primary voice and suppress background noise. Connect-time only. */\n  voiceFocus?: VoiceFocus;\n  /** Background audio suppression aggressiveness, from 0.0 to 1.0. Connect-time only. */\n  voiceFocusThreshold?: number;\n  /**\n   * Accuracy/latency preset for u3-rt-pro: `min_latency`, `balanced`, or `max_accuracy`.\n   * Explicit silence, partials, or VAD options still take precedence over mode defaults.\n   */\n  mode?: 'min_latency' | 'balanced' | 'max_accuracy';\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-3-5-pro',\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    const speechModel = opts.speechModel ?? defaultSTTOptions.speechModel;\n    if (!isU3ProModel(speechModel)) {\n      for (const param of ['prompt', 'voiceFocus', 'voiceFocusThreshold', 'mode'] as const) {\n        if (opts[param] !== undefined) {\n          throw new Error(\n            `The '${param}' parameter is only supported with the ${U3_PRO_MODELS.join(', ')} models.`,\n          );\n        }\n      }\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: ${this.closed})`,\n          );\n        }\n      }\n    }\n\n    this.closed = true;\n  }\n\n  async #connectWS(): Promise<WebSocket> {\n    // u3-rt-pro family models default both min and max silence to 100ms when unset.\n    let minSilence = this.#opts.minTurnSilence;\n    let maxSilence = this.#opts.maxTurnSilence;\n    if (isU3ProModel(this.#opts.speechModel)) {\n      if (minSilence === undefined) minSilence = 100;\n      if (maxSilence === undefined) maxSilence = minSilence;\n    }\n\n    // Default language_detection to true for multilingual / u3-rt-pro-family models, false otherwise.\n    const defaultLanguageDetection =\n      this.#opts.speechModel.includes('multilingual') || isU3ProModel(this.#opts.speechModel);\n    const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;\n\n    const liveConfig: Record<string, unknown> = {\n      sample_rate: this.#opts.sampleRate,\n      encoding: this.#opts.encoding,\n      speech_model: this.#opts.speechModel,\n      format_turns: this.#opts.formatTurns,\n      end_of_turn_confidence_threshold: this.#opts.endOfTurnConfidenceThreshold,\n      min_turn_silence: minSilence,\n      max_turn_silence: maxSilence,\n      keyterms_prompt:\n        this.#opts.keytermsPrompt !== undefined\n          ? JSON.stringify(this.#opts.keytermsPrompt)\n          : undefined,\n      language_detection: languageDetection,\n      prompt: this.#opts.prompt,\n      vad_threshold: this.#opts.vadThreshold,\n      speaker_labels: this.#opts.speakerLabels,\n      max_speakers: this.#opts.maxSpeakers,\n      domain: this.#opts.domain,\n      voice_focus: this.#opts.voiceFocus,\n      voice_focus_threshold: this.#opts.voiceFocusThreshold,\n      mode: this.#opts.mode,\n    };\n\n    const url = new URL(`${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. The base SpeechStream only resamples\n            // sample rate, so reject any frame that is not already downmixed.\n            frames = audioStream.write(data.data.buffer as ArrayBuffer);\n          } else {\n            throw new Error('sample rate or channel count of frame does not match');\n          }\n\n          for (const frame of frames) {\n            this.#speechDurationInS += frame.samplesPerChannel / frame.sampleRate;\n            ws.send(frame.data.buffer);\n          }\n        }\n      } finally {\n        closing = true;\n        try {\n          ws.send(SpeechStream.CLOSE_MSG);\n        } catch {\n          // ignore — socket may already be closing\n        }\n        wsMonitor.cancel();\n      }\n    };\n\n    let messageHandler: ((msg: RawData, isBinary: boolean) => void) | null = null;\n    const listenTask = Task.from(async (controller) => {\n      const listenMessage = new Promise<void>((resolve, reject) => {\n        messageHandler = (msg, isBinary) => {\n          if (isBinary) {\n            this.#logger.error('unexpected binary message from AssemblyAI');\n            return;\n          }\n          try {\n            const json = JSON.parse(msg.toString()) as StreamEventMessage;\n            this.#processStreamEvent(json);\n            if (this.closed || closing) {\n              resolve();\n            }\n          } catch (err) {\n            this.#logger.error(`AssemblyAI: error processing message: ${msg}`);\n            reject(err);\n          }\n        };\n        ws.on('message', messageHandler);\n      });\n\n      await Promise.race([listenMessage, waitForAbort(controller.signal)]);\n    });\n\n    const configTask = Task.from(async (controller) => {\n      // Drain any messages queued while the socket was reconnecting.\n      while (this.#pendingConfigMessages.length > 0) {\n        const msg = this.#pendingConfigMessages.shift()!;\n        ws.send(JSON.stringify(msg));\n      }\n\n      while (!controller.signal.aborted) {\n        await Promise.race([this.#configMessagePending.await, waitForAbort(controller.signal)]);\n        if (controller.signal.aborted) return;\n\n        this.#configMessagePending = new Future();\n        while (this.#pendingConfigMessages.length > 0) {\n          const msg = this.#pendingConfigMessages.shift()!;\n          ws.send(JSON.stringify(msg));\n        }\n      }\n    });\n\n    try {\n      await Promise.all([sendTask(), listenTask.result, wsMonitor.result]);\n    } finally {\n      closing = true;\n      sessionController.abort();\n      listenTask.cancel();\n      configTask.cancel();\n      if (messageHandler) ws.off('message', messageHandler);\n      try {\n        ws.close();\n      } catch {\n        // ignore\n      }\n    }\n  }\n\n  #averageConfidence(words: Array<{ confidence?: number }>): number {\n    if (words.length === 0) return 0;\n    return words.reduce((sum, w) => sum + (w.confidence ?? 0), 0) / words.length;\n  }\n\n  #processStreamEvent(data: StreamEventMessage) {\n    const messageType = data.type;\n\n    if (messageType === 'Begin') {\n      this.#sessionId = data.id ?? null;\n      this.#expiresAt = data.expires_at ?? null;\n      this.#logger.info(\n        `AssemblyAI session started id=${this.#sessionId} expires_at=${this.#expiresAt}`,\n      );\n      return;\n    }\n\n    if (messageType === 'SpeechStarted') {\n      this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n      return;\n    }\n\n    if (messageType === 'Termination') {\n      this.#logger.debug(\n        `AssemblyAI session terminated audio_duration=${data.audio_duration_seconds}s session_duration=${data.session_duration_seconds}s`,\n      );\n      return;\n    }\n\n    if (messageType !== 'Turn') {\n      return;\n    }\n\n    const words = data.words ?? [];\n    const endOfTurn = Boolean(data.end_of_turn);\n    const turnIsFormatted = Boolean(data.turn_is_formatted);\n    const utterance = data.utterance ?? '';\n    const transcript = data.transcript ?? '';\n    const language = normalizeLanguage(data.language_code ?? 'en');\n\n    // Word timestamps are in milliseconds:\n    // https://www.assemblyai.com/docs/api-reference/streaming-api/streaming-api#receive.receiveTurn.words\n    const timedWords = words.map((word) =>\n      createTimedString({\n        text: word.text ?? '',\n        startTime: (word.start ?? 0) / 1000 + this.startTimeOffset,\n        endTime: (word.end ?? 0) / 1000 + this.startTimeOffset,\n        confidence: word.confidence ?? 0,\n        startTimeOffset: this.startTimeOffset,\n      }),\n    );\n\n    let startTime = 0;\n    let endTime = 0;\n    let confidence = 0;\n\n    // `words` are cumulative for the turn — emit as an interim transcript.\n    if (timedWords.length > 0) {\n      const interimText = timedWords.map((w) => w.text).join(' ');\n      startTime = timedWords[0]!.startTime ?? 0;\n      endTime = timedWords[timedWords.length - 1]!.endTime ?? 0;\n      confidence = this.#averageConfidence(timedWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: interimText,\n            startTime,\n            endTime,\n            confidence,\n            words: timedWords,\n          },\n        ],\n      });\n    }\n\n    // `utterance` is chunk-based (not cumulative) — emit as a preflight transcript\n    // covering only the words since the last preflight.\n    if (utterance) {\n      if (this.#lastPreflightStartTime === 0) {\n        this.#lastPreflightStartTime = startTime;\n      }\n\n      const utteranceWords = timedWords.filter(\n        (w) => w.startTime !== undefined && w.startTime >= this.#lastPreflightStartTime,\n      );\n      const utteranceConfidence = this.#averageConfidence(utteranceWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.PREFLIGHT_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: utterance,\n            startTime: this.#lastPreflightStartTime,\n            endTime,\n            confidence: utteranceConfidence,\n            words: utteranceWords,\n          },\n        ],\n      });\n      this.#lastPreflightStartTime = endTime;\n    }\n\n    // End-of-turn: emit FINAL_TRANSCRIPT + END_OF_SPEECH.\n    // If the user asked for formatted turns, wait for a formatted final.\n    const waitingForFormatted = this.#opts.formatTurns === true && !turnIsFormatted;\n    if (endOfTurn && !waitingForFormatted) {\n      this.queue.put({\n        type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: transcript,\n            startTime,\n            endTime,\n            confidence,\n            words: timedWords,\n          },\n        ],\n      });\n\n      this.queue.put({ type: stt.SpeechEventType.END_OF_SPEECH });\n      if (this.#speechDurationInS > 0) {\n        this.queue.put({\n          type: stt.SpeechEventType.RECOGNITION_USAGE,\n          // Propagate the AssemblyAI session id as the request id so metrics\n          // can be correlated back to a specific connection, mirroring how\n          // Deepgram surfaces its `request_id`.\n          requestId: this.#sessionId ?? undefined,\n          recognitionUsage: {\n            audioDuration: this.#speechDurationInS,\n          },\n        });\n        this.#speechDurationInS = 0;\n        this.#lastPreflightStartTime = 0;\n      }\n    }\n  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package/dist/stt.d.cts

@@ -1,5 +1,5 @@
 import { type APIConnectOptions, type AudioBuffer, stt } from '@livekit/agents';
-import type { STTEncoding, STTModels } from './models.js';
+import type { STTEncoding, STTModels, VoiceFocus } from './models.js';
 export interface STTOptions {
     apiKey?: string;
     sampleRate: number;
@@ -33,6 +33,15 @@ export interface STTOptions {
     speakerLabels?: boolean;
     maxSpeakers?: number;
     domain?: string;
+    /** Isolate the primary voice and suppress background noise. Connect-time only. */
+    voiceFocus?: VoiceFocus;
+    /** Background audio suppression aggressiveness, from 0.0 to 1.0. Connect-time only. */
+    voiceFocusThreshold?: number;
+    /**
+     * Accuracy/latency preset for u3-rt-pro: `min_latency`, `balanced`, or `max_accuracy`.
+     * Explicit silence, partials, or VAD options still take precedence over mode defaults.
+     */
+    mode?: 'min_latency' | 'balanced' | 'max_accuracy';
     baseUrl: string;
 }
 export declare class STT extends stt.STT {

package/dist/stt.d.ts

@@ -1,5 +1,5 @@
 import { type APIConnectOptions, type AudioBuffer, stt } from '@livekit/agents';
-import type { STTEncoding, STTModels } from './models.js';
+import type { STTEncoding, STTModels, VoiceFocus } from './models.js';
 export interface STTOptions {
     apiKey?: string;
     sampleRate: number;
@@ -33,6 +33,15 @@ export interface STTOptions {
     speakerLabels?: boolean;
     maxSpeakers?: number;
     domain?: string;
+    /** Isolate the primary voice and suppress background noise. Connect-time only. */
+    voiceFocus?: VoiceFocus;
+    /** Background audio suppression aggressiveness, from 0.0 to 1.0. Connect-time only. */
+    voiceFocusThreshold?: number;
+    /**
+     * Accuracy/latency preset for u3-rt-pro: `min_latency`, `balanced`, or `max_accuracy`.
+     * Explicit silence, partials, or VAD options still take precedence over mode defaults.
+     */
+    mode?: 'min_latency' | 'balanced' | 'max_accuracy';
     baseUrl: string;
 }
 export declare class STT extends stt.STT {

package/dist/stt.d.ts.map

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

@@ -10,12 +10,16 @@ import {
   waitForAbort
 } from "@livekit/agents";
 import { WebSocket } from "ws";
+const U3_PRO_MODELS = ["u3-rt-pro", "u3-rt-pro-beta-1", "universal-3-5-pro"];
+function isU3ProModel(model) {
+  return U3_PRO_MODELS.includes(model);
+}
 const defaultSTTOptions = {
   apiKey: process.env.ASSEMBLYAI_API_KEY,
   sampleRate: 16e3,
   bufferSizeMs: 50,
   encoding: "pcm_s16le",
-  speechModel: "universal-streaming-english",
+  speechModel: "universal-3-5-pro",
   baseUrl: "wss://streaming.assemblyai.com"
 };
 class STT extends stt.STT {
@@ -38,8 +42,15 @@ class STT extends stt.STT {
       log().warn("'u3-pro' is deprecated, use 'u3-rt-pro' instead.");
       opts.speechModel = "u3-rt-pro";
     }
-    if (opts.prompt !== void 0 && opts.speechModel !== "u3-rt-pro") {
-      throw new Error("The 'prompt' parameter is only supported with the 'u3-rt-pro' model.");
+    const speechModel = opts.speechModel ?? defaultSTTOptions.speechModel;
+    if (!isU3ProModel(speechModel)) {
+      for (const param of ["prompt", "voiceFocus", "voiceFocusThreshold", "mode"]) {
+        if (opts[param] !== void 0) {
+          throw new Error(
+            `The '${param}' parameter is only supported with the ${U3_PRO_MODELS.join(", ")} models.`
+          );
+        }
+      }
     }
     const apiKey = opts.apiKey ?? defaultSTTOptions.apiKey;
     if (!apiKey) {
@@ -162,11 +173,11 @@ class SpeechStream extends stt.SpeechStream {
   async #connectWS() {
     let minSilence = this.#opts.minTurnSilence;
     let maxSilence = this.#opts.maxTurnSilence;
-    if (this.#opts.speechModel === "u3-rt-pro") {
+    if (isU3ProModel(this.#opts.speechModel)) {
       if (minSilence === void 0) minSilence = 100;
       if (maxSilence === void 0) maxSilence = minSilence;
     }
-    const defaultLanguageDetection = this.#opts.speechModel.includes("multilingual") || this.#opts.speechModel === "u3-rt-pro";
+    const defaultLanguageDetection = this.#opts.speechModel.includes("multilingual") || isU3ProModel(this.#opts.speechModel);
     const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;
     const liveConfig = {
       sample_rate: this.#opts.sampleRate,
@@ -182,7 +193,10 @@ class SpeechStream extends stt.SpeechStream {
       vad_threshold: this.#opts.vadThreshold,
       speaker_labels: this.#opts.speakerLabels,
       max_speakers: this.#opts.maxSpeakers,
-      domain: this.#opts.domain
+      domain: this.#opts.domain,
+      voice_focus: this.#opts.voiceFocus,
+      voice_focus_threshold: this.#opts.voiceFocusThreshold,
+      mode: this.#opts.mode
     };
     const url = new URL(`${this.#opts.baseUrl}/v3/ws`);
     for (const [key, value] of Object.entries(liveConfig)) {

package/dist/stt.js.map

@@ -1 +1 @@
-{"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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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  /** Isolate the primary voice and suppress background noise. Connect-time only. */\n  voiceFocus?: VoiceFocus;\n  /** Background audio suppression aggressiveness, from 0.0 to 1.0. Connect-time only. */\n  voiceFocusThreshold?: number;\n  /**\n   * Accuracy/latency preset for u3-rt-pro: `min_latency`, `balanced`, or `max_accuracy`.\n   * Explicit silence, partials, or VAD options still take precedence over mode defaults.\n   */\n  mode?: 'min_latency' | 'balanced' | 'max_accuracy';\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-3-5-pro',\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    const speechModel = opts.speechModel ?? defaultSTTOptions.speechModel;\n    if (!isU3ProModel(speechModel)) {\n      for (const param of ['prompt', 'voiceFocus', 'voiceFocusThreshold', 'mode'] as const) {\n        if (opts[param] !== undefined) {\n          throw new Error(\n            `The '${param}' parameter is only supported with the ${U3_PRO_MODELS.join(', ')} models.`,\n          );\n        }\n      }\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: ${this.closed})`,\n          );\n        }\n      }\n    }\n\n    this.closed = true;\n  }\n\n  async #connectWS(): Promise<WebSocket> {\n    // u3-rt-pro family models default both min and max silence to 100ms when unset.\n    let minSilence = this.#opts.minTurnSilence;\n    let maxSilence = this.#opts.maxTurnSilence;\n    if (isU3ProModel(this.#opts.speechModel)) {\n      if (minSilence === undefined) minSilence = 100;\n      if (maxSilence === undefined) maxSilence = minSilence;\n    }\n\n    // Default language_detection to true for multilingual / u3-rt-pro-family models, false otherwise.\n    const defaultLanguageDetection =\n      this.#opts.speechModel.includes('multilingual') || isU3ProModel(this.#opts.speechModel);\n    const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;\n\n    const liveConfig: Record<string, unknown> = {\n      sample_rate: this.#opts.sampleRate,\n      encoding: this.#opts.encoding,\n      speech_model: this.#opts.speechModel,\n      format_turns: this.#opts.formatTurns,\n      end_of_turn_confidence_threshold: this.#opts.endOfTurnConfidenceThreshold,\n      min_turn_silence: minSilence,\n      max_turn_silence: maxSilence,\n      keyterms_prompt:\n        this.#opts.keytermsPrompt !== undefined\n          ? JSON.stringify(this.#opts.keytermsPrompt)\n          : undefined,\n      language_detection: languageDetection,\n      prompt: this.#opts.prompt,\n      vad_threshold: this.#opts.vadThreshold,\n      speaker_labels: this.#opts.speakerLabels,\n      max_speakers: this.#opts.maxSpeakers,\n      domain: this.#opts.domain,\n      voice_focus: this.#opts.voiceFocus,\n      voice_focus_threshold: this.#opts.voiceFocusThreshold,\n      mode: this.#opts.mode,\n    };\n\n    const url = new URL(`${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. The base SpeechStream only resamples\n            // sample rate, so reject any frame that is not already downmixed.\n            frames = audioStream.write(data.data.buffer as ArrayBuffer);\n          } else {\n            throw new Error('sample rate or channel count of frame does not match');\n          }\n\n          for (const frame of frames) {\n            this.#speechDurationInS += frame.samplesPerChannel / frame.sampleRate;\n            ws.send(frame.data.buffer);\n          }\n        }\n      } finally {\n        closing = true;\n        try {\n          ws.send(SpeechStream.CLOSE_MSG);\n        } catch {\n          // ignore — socket may already be closing\n        }\n        wsMonitor.cancel();\n      }\n    };\n\n    let messageHandler: ((msg: RawData, isBinary: boolean) => void) | null = null;\n    const listenTask = Task.from(async (controller) => {\n      const listenMessage = new Promise<void>((resolve, reject) => {\n        messageHandler = (msg, isBinary) => {\n          if (isBinary) {\n            this.#logger.error('unexpected binary message from AssemblyAI');\n            return;\n          }\n          try {\n            const json = JSON.parse(msg.toString()) as StreamEventMessage;\n            this.#processStreamEvent(json);\n            if (this.closed || closing) {\n              resolve();\n            }\n          } catch (err) {\n            this.#logger.error(`AssemblyAI: error processing message: ${msg}`);\n            reject(err);\n          }\n        };\n        ws.on('message', messageHandler);\n      });\n\n      await Promise.race([listenMessage, waitForAbort(controller.signal)]);\n    });\n\n    const configTask = Task.from(async (controller) => {\n      // Drain any messages queued while the socket was reconnecting.\n      while (this.#pendingConfigMessages.length > 0) {\n        const msg = this.#pendingConfigMessages.shift()!;\n        ws.send(JSON.stringify(msg));\n      }\n\n      while (!controller.signal.aborted) {\n        await Promise.race([this.#configMessagePending.await, waitForAbort(controller.signal)]);\n        if (controller.signal.aborted) return;\n\n        this.#configMessagePending = new Future();\n        while (this.#pendingConfigMessages.length > 0) {\n          const msg = this.#pendingConfigMessages.shift()!;\n          ws.send(JSON.stringify(msg));\n        }\n      }\n    });\n\n    try {\n      await Promise.all([sendTask(), listenTask.result, wsMonitor.result]);\n    } finally {\n      closing = true;\n      sessionController.abort();\n      listenTask.cancel();\n      configTask.cancel();\n      if (messageHandler) ws.off('message', messageHandler);\n      try {\n        ws.close();\n      } catch {\n        // ignore\n      }\n    }\n  }\n\n  #averageConfidence(words: Array<{ confidence?: number }>): number {\n    if (words.length === 0) return 0;\n    return words.reduce((sum, w) => sum + (w.confidence ?? 0), 0) / words.length;\n  }\n\n  #processStreamEvent(data: StreamEventMessage) {\n    const messageType = data.type;\n\n    if (messageType === 'Begin') {\n      this.#sessionId = data.id ?? null;\n      this.#expiresAt = data.expires_at ?? null;\n      this.#logger.info(\n        `AssemblyAI session started id=${this.#sessionId} expires_at=${this.#expiresAt}`,\n      );\n      return;\n    }\n\n    if (messageType === 'SpeechStarted') {\n      this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n      return;\n    }\n\n    if (messageType === 'Termination') {\n      this.#logger.debug(\n        `AssemblyAI session terminated audio_duration=${data.audio_duration_seconds}s session_duration=${data.session_duration_seconds}s`,\n      );\n      return;\n    }\n\n    if (messageType !== 'Turn') {\n      return;\n    }\n\n    const words = data.words ?? [];\n    const endOfTurn = Boolean(data.end_of_turn);\n    const turnIsFormatted = Boolean(data.turn_is_formatted);\n    const utterance = data.utterance ?? '';\n    const transcript = data.transcript ?? '';\n    const language = normalizeLanguage(data.language_code ?? 'en');\n\n    // Word timestamps are in milliseconds:\n    // https://www.assemblyai.com/docs/api-reference/streaming-api/streaming-api#receive.receiveTurn.words\n    const timedWords = words.map((word) =>\n      createTimedString({\n        text: word.text ?? '',\n        startTime: (word.start ?? 0) / 1000 + this.startTimeOffset,\n        endTime: (word.end ?? 0) / 1000 + this.startTimeOffset,\n        confidence: word.confidence ?? 0,\n        startTimeOffset: this.startTimeOffset,\n      }),\n    );\n\n    let startTime = 0;\n    let endTime = 0;\n    let confidence = 0;\n\n    // `words` are cumulative for the turn — emit as an interim transcript.\n    if (timedWords.length > 0) {\n      const interimText = timedWords.map((w) => w.text).join(' ');\n      startTime = timedWords[0]!.startTime ?? 0;\n      endTime = timedWords[timedWords.length - 1]!.endTime ?? 0;\n      confidence = this.#averageConfidence(timedWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: interimText,\n            startTime,\n            endTime,\n            confidence,\n            words: timedWords,\n          },\n        ],\n      });\n    }\n\n    // `utterance` is chunk-based (not cumulative) — emit as a preflight transcript\n    // covering only the words since the last preflight.\n    if (utterance) {\n      if (this.#lastPreflightStartTime === 0) {\n        this.#lastPreflightStartTime = startTime;\n      }\n\n      const utteranceWords = timedWords.filter(\n        (w) => w.startTime !== undefined && w.startTime >= this.#lastPreflightStartTime,\n      );\n      const utteranceConfidence = this.#averageConfidence(utteranceWords);\n\n      this.queue.put({\n        type: stt.SpeechEventType.PREFLIGHT_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: utterance,\n            startTime: this.#lastPreflightStartTime,\n            endTime,\n            confidence: utteranceConfidence,\n            words: utteranceWords,\n          },\n        ],\n      });\n      this.#lastPreflightStartTime = endTime;\n    }\n\n    // End-of-turn: emit FINAL_TRANSCRIPT + END_OF_SPEECH.\n    // If the user asked for formatted turns, wait for a formatted final.\n    const waitingForFormatted = this.#opts.formatTurns === true && !turnIsFormatted;\n    if (endOfTurn && !waitingForFormatted) {\n      this.queue.put({\n        type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n        alternatives: [\n          {\n            language,\n            text: transcript,\n            startTime,\n            endTime,\n            confidence,\n            words: timedWords,\n          },\n        ],\n      });\n\n      this.queue.put({ type: stt.SpeechEventType.END_OF_SPEECH });\n      if (this.#speechDurationInS > 0) {\n        this.queue.put({\n          type: stt.SpeechEventType.RECOGNITION_USAGE,\n          // Propagate the AssemblyAI session id as the request id so metrics\n          // can be correlated back to a specific connection, mirroring how\n          // Deepgram surfaces its `request_id`.\n          requestId: this.#sessionId ?? undefined,\n          recognitionUsage: {\n            audioDuration: this.#speechDurationInS,\n          },\n        });\n        this.#speechDurationInS = 0;\n        this.#lastPreflightStartTime = 0;\n      }\n    }\n  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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@livekit/agents-plugin-assemblyai",
-  "version": "1.4.6",
+  "version": "1.4.8",
   "description": "AssemblyAI plugin for LiveKit Agents for Node.js",
   "main": "dist/index.js",
   "require": "dist/index.cjs",
@@ -30,16 +30,16 @@
     "@types/ws": "^8.5.10",
     "tsup": "^8.3.5",
     "typescript": "^5.0.0",
-    "@livekit/agents": "1.4.6",
-    "@livekit/agents-plugin-silero": "1.4.6",
-    "@livekit/agents-plugins-test": "1.4.6"
+    "@livekit/agents": "1.4.8",
+    "@livekit/agents-plugin-silero": "1.4.8",
+    "@livekit/agents-plugins-test": "1.4.8"
   },
   "dependencies": {
     "ws": "^8.16.0"
   },
   "peerDependencies": {
     "@livekit/rtc-node": "^0.13.29",
-    "@livekit/agents": "1.4.6"
+    "@livekit/agents": "1.4.8"
   },
   "scripts": {
     "build": "tsup --onSuccess \"pnpm build:types\"",

package/src/models.ts

@@ -6,9 +6,13 @@ export type STTModels =
   | 'universal-streaming-english'
   | 'universal-streaming-multilingual'
   | 'u3-rt-pro'
+  | 'u3-rt-pro-beta-1'
+  | 'universal-3-5-pro'
   // Deprecated alias — AssemblyAI maps this to `u3-rt-pro` server-side, but the
   // Python plugin emits a warning and rewrites it. Kept here so TS users don't
   // break if they already pass it.
   | 'u3-pro';
 
 export type STTEncoding = 'pcm_s16le' | 'pcm_mulaw';
+
+export type VoiceFocus = 'near-field' | 'far-field';

package/src/stt.ts

@@ -18,7 +18,13 @@ import {
 import type { AudioFrame } from '@livekit/rtc-node';
 import type { RawData } from 'ws';
 import { WebSocket } from 'ws';
-import type { STTEncoding, STTModels } from './models.js';
+import type { STTEncoding, STTModels, VoiceFocus } from './models.js';
+
+const U3_PRO_MODELS = ['u3-rt-pro', 'u3-rt-pro-beta-1', 'universal-3-5-pro'] as const;
+
+function isU3ProModel(model: STTModels): boolean {
+  return U3_PRO_MODELS.includes(model as (typeof U3_PRO_MODELS)[number]);
+}
 
 // AssemblyAI Universal-Streaming (v3) message envelope. All fields are optional
 // since we narrow on `type` before reading anything else.
@@ -80,6 +86,15 @@ export interface STTOptions {
   speakerLabels?: boolean;
   maxSpeakers?: number;
   domain?: string;
+  /** Isolate the primary voice and suppress background noise. Connect-time only. */
+  voiceFocus?: VoiceFocus;
+  /** Background audio suppression aggressiveness, from 0.0 to 1.0. Connect-time only. */
+  voiceFocusThreshold?: number;
+  /**
+   * Accuracy/latency preset for u3-rt-pro: `min_latency`, `balanced`, or `max_accuracy`.
+   * Explicit silence, partials, or VAD options still take precedence over mode defaults.
+   */
+  mode?: 'min_latency' | 'balanced' | 'max_accuracy';
   baseUrl: string;
 }
 
@@ -88,7 +103,7 @@ const defaultSTTOptions: STTOptions = {
   sampleRate: 16000,
   bufferSizeMs: 50,
   encoding: 'pcm_s16le',
-  speechModel: 'universal-streaming-english',
+  speechModel: 'universal-3-5-pro',
   baseUrl: 'wss://streaming.assemblyai.com',
 };
 
@@ -117,8 +132,15 @@ export class STT extends stt.STT {
       opts.speechModel = 'u3-rt-pro';
     }
 
-    if (opts.prompt !== undefined && opts.speechModel !== 'u3-rt-pro') {
-      throw new Error("The 'prompt' parameter is only supported with the 'u3-rt-pro' model.");
+    const speechModel = opts.speechModel ?? defaultSTTOptions.speechModel;
+    if (!isU3ProModel(speechModel)) {
+      for (const param of ['prompt', 'voiceFocus', 'voiceFocusThreshold', 'mode'] as const) {
+        if (opts[param] !== undefined) {
+          throw new Error(
+            `The '${param}' parameter is only supported with the ${U3_PRO_MODELS.join(', ')} models.`,
+          );
+        }
+      }
     }
 
     const apiKey = opts.apiKey ?? defaultSTTOptions.apiKey;
@@ -262,17 +284,17 @@ export class SpeechStream extends stt.SpeechStream {
   }
 
   async #connectWS(): Promise<WebSocket> {
-    // u3-rt-pro has different silence defaults — if unset, both min and max default to 100ms.
+    // u3-rt-pro family models default both min and max silence to 100ms when unset.
     let minSilence = this.#opts.minTurnSilence;
     let maxSilence = this.#opts.maxTurnSilence;
-    if (this.#opts.speechModel === 'u3-rt-pro') {
+    if (isU3ProModel(this.#opts.speechModel)) {
       if (minSilence === undefined) minSilence = 100;
       if (maxSilence === undefined) maxSilence = minSilence;
     }
 
-    // Default language_detection to true for multilingual / u3-rt-pro models, false otherwise.
+    // Default language_detection to true for multilingual / u3-rt-pro-family models, false otherwise.
     const defaultLanguageDetection =
-      this.#opts.speechModel.includes('multilingual') || this.#opts.speechModel === 'u3-rt-pro';
+      this.#opts.speechModel.includes('multilingual') || isU3ProModel(this.#opts.speechModel);
     const languageDetection = this.#opts.languageDetection ?? defaultLanguageDetection;
 
     const liveConfig: Record<string, unknown> = {
@@ -293,6 +315,9 @@ export class SpeechStream extends stt.SpeechStream {
       speaker_labels: this.#opts.speakerLabels,
       max_speakers: this.#opts.maxSpeakers,
       domain: this.#opts.domain,
+      voice_focus: this.#opts.voiceFocus,
+      voice_focus_threshold: this.#opts.voiceFocusThreshold,
+      mode: this.#opts.mode,
     };
 
     const url = new URL(`${this.#opts.baseUrl}/v3/ws`);