@livekit/agents 1.0.19 → 1.0.21

package/dist/inference/stt.js

@@ -1,8 +1,8 @@
 import {} from "@livekit/rtc-node";
-import {} from "ws";
 import { APIError, APIStatusError } from "../_exceptions.js";
 import { AudioByteStream } from "../audio.js";
 import { log } from "../log.js";
+import { createStreamChannel } from "../stream/stream_channel.js";
 import {
   STT as BaseSTT,
   SpeechStream as BaseSpeechStream,
@@ -80,6 +80,32 @@ class STT extends BaseSTT {
     this.streams.add(stream);
     return stream;
   }
+  async connectWs(timeout) {
+    const params = {
+      settings: {
+        sample_rate: String(this.opts.sampleRate),
+        encoding: this.opts.encoding,
+        extra: this.opts.modelOptions
+      }
+    };
+    if (this.opts.model && this.opts.model !== "auto") {
+      params.model = this.opts.model;
+    }
+    if (this.opts.language) {
+      params.settings.language = this.opts.language;
+    }
+    let baseURL = this.opts.baseURL;
+    if (baseURL.startsWith("http://") || baseURL.startsWith("https://")) {
+      baseURL = baseURL.replace("http", "ws");
+    }
+    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);
+    const url = `${baseURL}/stt`;
+    const headers = { Authorization: `Bearer ${token}` };
+    const socket = await connectWs(url, headers, timeout);
+    const msg = { ...params, type: "session.create" };
+    socket.send(JSON.stringify(msg));
+    return socket;
+  }
 }
 class SpeechStream extends BaseSpeechStream {
   opts;
@@ -87,158 +113,178 @@ class SpeechStream extends BaseSpeechStream {
   speaking = false;
   speechDuration = 0;
   reconnectEvent = new Event();
+  stt;
+  connOptions;
   #logger = log();
   constructor(sttImpl, opts, connOptions) {
     super(sttImpl, opts.sampleRate, connOptions);
     this.opts = opts;
+    this.stt = sttImpl;
+    this.connOptions = connOptions;
   }
   get label() {
     return "inference.SpeechStream";
   }
   updateOptions(opts) {
     this.opts = { ...this.opts, ...opts };
+    this.reconnectEvent.set();
   }
   async run() {
-    let ws = null;
-    let closingWs = false;
-    this.reconnectEvent.set();
-    const connect = async () => {
-      const params = {
-        settings: {
-          sample_rate: String(this.opts.sampleRate),
-          encoding: this.opts.encoding,
-          extra: this.opts.modelOptions
-        }
+    while (true) {
+      let ws = null;
+      let closing = false;
+      let finalReceived = false;
+      const eventChannel = createStreamChannel();
+      const resourceCleanup = () => {
+        if (closing) return;
+        closing = true;
+        eventChannel.close();
+        ws == null ? void 0 : ws.removeAllListeners();
+        ws == null ? void 0 : ws.close();
       };
-      if (this.opts.model && this.opts.model !== "auto") {
-        params.model = this.opts.model;
-      }
-      if (this.opts.language) {
-        params.settings.language = this.opts.language;
-      }
-      let baseURL = this.opts.baseURL;
-      if (baseURL.startsWith("http://") || baseURL.startsWith("https://")) {
-        baseURL = baseURL.replace("http", "ws");
-      }
-      const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);
-      const url = `${baseURL}/stt`;
-      const headers = { Authorization: `Bearer ${token}` };
-      const socket = await connectWs(url, headers, 1e4);
-      const msg = { ...params, type: "session.create" };
-      socket.send(JSON.stringify(msg));
-      return socket;
-    };
-    const send = async (socket, signal) => {
-      const audioStream = new AudioByteStream(
-        this.opts.sampleRate,
-        1,
-        Math.floor(this.opts.sampleRate / 20)
-        // 50ms
-      );
-      for await (const ev of this.input) {
-        if (signal.aborted) break;
-        let frames;
-        if (ev === SpeechStream.FLUSH_SENTINEL) {
-          frames = audioStream.flush();
-        } else {
-          const frame = ev;
-          frames = audioStream.write(new Int16Array(frame.data).buffer);
-        }
-        for (const frame of frames) {
-          this.speechDuration += frame.samplesPerChannel / frame.sampleRate;
-          const base64 = Buffer.from(frame.data.buffer).toString("base64");
-          const msg = { type: "input_audio", audio: base64 };
-          socket.send(JSON.stringify(msg));
-        }
-      }
-      closingWs = true;
-      socket.send(JSON.stringify({ type: "session.finalize" }));
-    };
-    const recv = async (socket, signal) => {
-      while (!this.closed && !signal.aborted) {
-        const dataPromise = new Promise((resolve, reject) => {
-          const messageHandler = (d) => {
-            resolve(d.toString());
-            removeListeners();
+      const createWsListener = async (ws2, signal) => {
+        return new Promise((resolve, reject) => {
+          const onAbort = () => {
+            resourceCleanup();
+            reject(new Error("WebSocket connection aborted"));
           };
-          const errorHandler = (e) => {
+          signal.addEventListener("abort", onAbort, { once: true });
+          ws2.on("message", (data) => {
+            const json = JSON.parse(data.toString());
+            eventChannel.write(json);
+          });
+          ws2.on("error", (e) => {
+            this.#logger.error({ error: e }, "WebSocket error");
+            resourceCleanup();
             reject(e);
-            removeListeners();
-          };
-          const closeHandler = (code) => {
-            if (closingWs) {
-              resolve("");
+          });
+          ws2.on("close", (code) => {
+            resourceCleanup();
+            if (!closing) return this.#logger.error("WebSocket closed unexpectedly");
+            if (finalReceived) return resolve();
+            reject(
+              new APIStatusError({
+                message: "LiveKit STT connection closed unexpectedly",
+                options: { statusCode: code }
+              })
+            );
+          });
+        });
+      };
+      const send = async (socket, signal) => {
+        const audioStream = new AudioByteStream(
+          this.opts.sampleRate,
+          1,
+          Math.floor(this.opts.sampleRate / 20)
+          // 50ms
+        );
+        const abortPromise = new Promise((_, reject) => {
+          if (signal.aborted) {
+            return reject(new Error("Send aborted"));
+          }
+          const onAbort = () => reject(new Error("Send aborted"));
+          signal.addEventListener("abort", onAbort, { once: true });
+        });
+        const iterator = this.input[Symbol.asyncIterator]();
+        try {
+          while (true) {
+            const result = await Promise.race([iterator.next(), abortPromise]);
+            if (result.done) break;
+            const ev = result.value;
+            let frames;
+            if (ev === SpeechStream.FLUSH_SENTINEL) {
+              frames = audioStream.flush();
             } else {
-              reject(
-                new APIStatusError({
-                  message: "LiveKit STT connection closed unexpectedly",
-                  options: { statusCode: code }
-                })
-              );
+              const frame = ev;
+              frames = audioStream.write(new Int16Array(frame.data).buffer);
             }
-            removeListeners();
-          };
-          const removeListeners = () => {
-            socket.removeListener("message", messageHandler);
-            socket.removeListener("error", errorHandler);
-            socket.removeListener("close", closeHandler);
-          };
-          socket.once("message", messageHandler);
-          socket.once("error", errorHandler);
-          socket.once("close", closeHandler);
-        });
-        const data = await Promise.race([dataPromise, waitForAbort(signal)]);
-        if (!data || signal.aborted) return;
-        const json = JSON.parse(data);
-        const type = json.type;
-        switch (type) {
-          case "session.created":
-          case "session.finalized":
-          case "session.closed":
-            break;
-          case "interim_transcript":
-            this.processTranscript(json, false);
-            break;
-          case "final_transcript":
-            this.processTranscript(json, true);
-            break;
-          case "error":
-            this.#logger.error("received error from LiveKit STT: %o", json);
-            throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);
-          default:
-            this.#logger.warn("received unexpected message from LiveKit STT: %o", json);
-            break;
+            for (const frame of frames) {
+              this.speechDuration += frame.samplesPerChannel / frame.sampleRate;
+              const base64 = Buffer.from(frame.data.buffer).toString("base64");
+              const msg = { type: "input_audio", audio: base64 };
+              socket.send(JSON.stringify(msg));
+            }
+          }
+          closing = true;
+          socket.send(JSON.stringify({ type: "session.finalize" }));
+        } catch (e) {
+          if (e.message === "Send aborted") {
+            return;
+          }
+          throw e;
         }
-      }
-    };
-    while (true) {
+      };
+      const recv = async (signal) => {
+        const serverEventStream = eventChannel.stream();
+        const reader = serverEventStream.getReader();
+        try {
+          while (!this.closed && !signal.aborted) {
+            const result = await reader.read();
+            if (signal.aborted) return;
+            if (result.done) return;
+            const json = result.value;
+            const type = json.type;
+            switch (type) {
+              case "session.created":
+              case "session.finalized":
+                break;
+              case "session.closed":
+                finalReceived = true;
+                resourceCleanup();
+                break;
+              case "interim_transcript":
+                this.processTranscript(json, false);
+                break;
+              case "final_transcript":
+                this.processTranscript(json, true);
+                break;
+              case "error":
+                this.#logger.error({ error: json }, "Received error from LiveKit STT");
+                resourceCleanup();
+                throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);
+              default:
+                this.#logger.warn(
+                  { message: json },
+                  "Received unexpected message from LiveKit STT"
+                );
+                break;
+            }
+          }
+        } finally {
+          reader.releaseLock();
+          try {
+            await serverEventStream.cancel();
+          } catch (e) {
+            this.#logger.debug("Error cancelling serverEventStream (may already be cancelled):", e);
+          }
+        }
+      };
       try {
-        ws = await connect();
-        const sendTask = Task.from(async ({ signal }) => {
-          await send(ws, signal);
-        });
-        const recvTask = Task.from(async ({ signal }) => {
-          await recv(ws, signal);
-        });
-        const tasks = [sendTask, recvTask];
-        const waitReconnectTask = Task.from(async ({ signal }) => {
-          await Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]);
-        });
+        ws = await this.stt.connectWs(this.connOptions.timeoutMs);
+        const controller = new AbortController();
+        const sendTask = Task.from(({ signal }) => send(ws, signal), controller);
+        const wsListenerTask = Task.from(({ signal }) => createWsListener(ws, signal), controller);
+        const recvTask = Task.from(({ signal }) => recv(signal), controller);
+        const waitReconnectTask = Task.from(
+          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),
+          controller
+        );
         try {
           await Promise.race([
-            Promise.all(tasks.map((task) => task.result)),
+            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),
             waitReconnectTask.result
           ]);
           if (!waitReconnectTask.done) break;
           this.reconnectEvent.clear();
         } finally {
-          await cancelAndWait([sendTask, recvTask, waitReconnectTask], DEFAULT_CANCEL_TIMEOUT);
+          await cancelAndWait(
+            [sendTask, wsListenerTask, recvTask, waitReconnectTask],
+            DEFAULT_CANCEL_TIMEOUT
+          );
+          resourceCleanup();
         }
       } finally {
-        try {
-          if (ws) ws.close();
-        } catch {
-        }
+        resourceCleanup();
       }
     }
   }

package/dist/inference/stt.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/inference/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame } from '@livekit/rtc-node';\nimport type { WebSocket } from 'ws';\nimport { type RawData } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport {\n  STT as BaseSTT,\n  SpeechStream as BaseSpeechStream,\n  type SpeechData,\n  type SpeechEvent,\n  SpeechEventType,\n} from '../stt/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { type AudioBuffer, Event, Task, cancelAndWait, shortuuid, waitForAbort } from '../utils.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type DeepgramModels =\n  | 'deepgram'\n  | 'deepgram/nova-3'\n  | 'deepgram/nova-3-general'\n  | 'deepgram/nova-3-medical'\n  | 'deepgram/nova-2-conversationalai'\n  | 'deepgram/nova-2'\n  | 'deepgram/nova-2-general'\n  | 'deepgram/nova-2-medical'\n  | 'deepgram/nova-2-phonecall';\n\nexport type CartesiaModels = 'cartesia' | 'cartesia/ink-whisper';\n\nexport type AssemblyaiModels = 'assemblyai' | 'assemblyai/universal-streaming';\n\nexport interface CartesiaOptions {\n  min_volume?: number; // default: not specified\n  max_silence_duration_secs?: number; // default: not specified\n}\n\nexport interface DeepgramOptions {\n  filler_words?: boolean; // default: true\n  interim_results?: boolean; // default: true\n  endpointing?: number; // default: 25 (ms)\n  punctuate?: boolean; // default: false\n  smart_format?: boolean;\n  keywords?: Array<[string, number]>;\n  keyterms?: string[];\n  profanity_filter?: boolean;\n  numerals?: boolean;\n  mip_opt_out?: boolean;\n}\n\nexport interface AssemblyAIOptions {\n  format_turns?: boolean; // default: false\n  end_of_turn_confidence_threshold?: number; // default: 0.01\n  min_end_of_turn_silence_when_confident?: number; // default: 0\n  max_turn_silence?: number; // default: not specified\n  keyterms_prompt?: string[]; // default: not specified\n}\n\nexport type STTLanguages =\n  | 'multi'\n  | 'en'\n  | 'de'\n  | 'es'\n  | 'fr'\n  | 'ja'\n  | 'pt'\n  | 'zh'\n  | 'hi'\n  | AnyString;\n\ntype _STTModels = DeepgramModels | CartesiaModels | AssemblyaiModels;\n\nexport type STTModels = _STTModels | 'auto' | AnyString;\n\nexport type ModelWithLanguage = `${_STTModels}:${STTLanguages}` | STTModels;\n\nexport type STTOptions<TModel extends STTModels> = TModel extends DeepgramModels\n  ? DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? AssemblyAIOptions\n      : Record<string, unknown>;\n\nexport type STTEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: STTEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'wss://agent-gateway.livekit.cloud/v1';\nconst DEFAULT_CANCEL_TIMEOUT = 5000;\n\nexport interface InferenceSTTOptions<TModel extends STTModels> {\n  model?: TModel;\n  language?: STTLanguages;\n  encoding: STTEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: STTOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference STT\n */\nexport class STT<TModel extends STTModels> extends BaseSTT {\n  private opts: InferenceSTTOptions<TModel>;\n  private streams: Set<SpeechStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts?: {\n    model?: TModel;\n    language?: STTLanguages;\n    baseURL?: string;\n    encoding?: STTEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label(): string {\n    return 'inference.STT';\n  }\n\n  static fromModelString(modelString: string): STT<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, language] = modelString.split(':') as [AnyString, STTLanguages];\n      return new STT({ model, language });\n    }\n    return new STT({ model: modelString });\n  }\n\n  protected async _recognize(_: AudioBuffer): Promise<SpeechEvent> {\n    throw new Error('LiveKit STT does not support batch recognition, use stream() instead');\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  stream(options?: {\n    language?: STTLanguages | string;\n    connOptions?: APIConnectOptions;\n  }): SpeechStream<TModel> {\n    const { language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n}\n\nexport class SpeechStream<TModel extends STTModels> extends BaseSpeechStream {\n  private opts: InferenceSTTOptions<TModel>;\n  private requestId = shortuuid('stt_request_');\n  private speaking = false;\n  private speechDuration = 0;\n  private reconnectEvent = new Event();\n\n  #logger = log();\n\n  constructor(\n    sttImpl: STT<TModel>,\n    opts: InferenceSTTOptions<TModel>,\n    connOptions: APIConnectOptions,\n  ) {\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.opts = opts;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n  }\n\n  protected async run(): Promise<void> {\n    let ws: WebSocket | null = null;\n    let closingWs = false;\n\n    this.reconnectEvent.set();\n\n    const connect = async () => {\n      const params = {\n        settings: {\n          sample_rate: String(this.opts.sampleRate),\n          encoding: this.opts.encoding,\n          extra: this.opts.modelOptions,\n        },\n      } as Record<string, unknown>;\n\n      if (this.opts.model && this.opts.model !== 'auto') {\n        params.model = this.opts.model;\n      }\n\n      if (this.opts.language) {\n        (params.settings as Record<string, unknown>).language = this.opts.language;\n      }\n\n      let baseURL = this.opts.baseURL;\n      if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n        baseURL = baseURL.replace('http', 'ws');\n      }\n\n      const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n      const url = `${baseURL}/stt`;\n      const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n      const socket = await connectWs(url, headers, 10000);\n      const msg = { ...params, type: 'session.create' };\n      socket.send(JSON.stringify(msg));\n\n      return socket;\n    };\n\n    const send = async (socket: WebSocket, signal: AbortSignal) => {\n      const audioStream = new AudioByteStream(\n        this.opts.sampleRate,\n        1,\n        Math.floor(this.opts.sampleRate / 20), // 50ms\n      );\n\n      for await (const ev of this.input) {\n        if (signal.aborted) break;\n        let frames: AudioFrame[];\n\n        if (ev === SpeechStream.FLUSH_SENTINEL) {\n          frames = audioStream.flush();\n        } else {\n          const frame = ev as AudioFrame;\n          frames = audioStream.write(new Int16Array(frame.data).buffer);\n        }\n\n        for (const frame of frames) {\n          this.speechDuration += frame.samplesPerChannel / frame.sampleRate;\n          const base64 = Buffer.from(frame.data.buffer).toString('base64');\n          const msg = { type: 'input_audio', audio: base64 };\n          socket.send(JSON.stringify(msg));\n        }\n      }\n\n      closingWs = true;\n      socket.send(JSON.stringify({ type: 'session.finalize' }));\n    };\n\n    const recv = async (socket: WebSocket, signal: AbortSignal) => {\n      while (!this.closed && !signal.aborted) {\n        const dataPromise = new Promise<string>((resolve, reject) => {\n          const messageHandler = (d: RawData) => {\n            resolve(d.toString());\n            removeListeners();\n          };\n          const errorHandler = (e: Error) => {\n            reject(e);\n            removeListeners();\n          };\n          const closeHandler = (code: number) => {\n            if (closingWs) {\n              resolve('');\n            } else {\n              reject(\n                new APIStatusError({\n                  message: 'LiveKit STT connection closed unexpectedly',\n                  options: { statusCode: code },\n                }),\n              );\n            }\n            removeListeners();\n          };\n          const removeListeners = () => {\n            socket.removeListener('message', messageHandler);\n            socket.removeListener('error', errorHandler);\n            socket.removeListener('close', closeHandler);\n          };\n          socket.once('message', messageHandler);\n          socket.once('error', errorHandler);\n          socket.once('close', closeHandler);\n        });\n\n        const data = await Promise.race([dataPromise, waitForAbort(signal)]);\n\n        if (!data || signal.aborted) return;\n\n        const json = JSON.parse(data);\n        const type = json.type as string | undefined;\n\n        switch (type) {\n          case 'session.created':\n          case 'session.finalized':\n          case 'session.closed':\n            break;\n          case 'interim_transcript':\n            this.processTranscript(json, false);\n            break;\n          case 'final_transcript':\n            this.processTranscript(json, true);\n            break;\n          case 'error':\n            this.#logger.error('received error from LiveKit STT: %o', json);\n            throw new APIError(`LiveKit STT returned error: ${JSON.stringify(json)}`);\n          default:\n            this.#logger.warn('received unexpected message from LiveKit STT: %o', json);\n            break;\n        }\n      }\n    };\n\n    while (true) {\n      try {\n        ws = await connect();\n\n        const sendTask = Task.from(async ({ signal }) => {\n          await send(ws!, signal);\n        });\n\n        const recvTask = Task.from(async ({ signal }) => {\n          await recv(ws!, signal);\n        });\n\n        const tasks = [sendTask, recvTask];\n        const waitReconnectTask = Task.from(async ({ signal }) => {\n          await Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]);\n        });\n\n        try {\n          await Promise.race([\n            Promise.all(tasks.map((task) => task.result)),\n            waitReconnectTask.result,\n          ]);\n\n          if (!waitReconnectTask.done) break;\n          this.reconnectEvent.clear();\n        } finally {\n          await cancelAndWait([sendTask, recvTask, waitReconnectTask], DEFAULT_CANCEL_TIMEOUT);\n        }\n      } finally {\n        try {\n          if (ws) ws.close();\n        } catch {}\n      }\n    }\n  }\n\n  private processTranscript(data: Record<string, any>, isFinal: boolean) {\n    const requestId = data.request_id ?? this.requestId;\n    const text = data.transcript ?? '';\n    const language = data.language ?? this.opts.language ?? 'en';\n\n    if (!text && !isFinal) return;\n\n    // We'll have a more accurate way of detecting when speech started when we have VAD\n    if (!this.speaking) {\n      this.speaking = true;\n      this.queue.put({ type: SpeechEventType.START_OF_SPEECH });\n    }\n\n    const speechData: SpeechData = {\n      language,\n      startTime: data.start ?? 0,\n      endTime: data.duration ?? 0,\n      confidence: data.confidence ?? 1.0,\n      text,\n    };\n\n    if (isFinal) {\n      if (this.speechDuration > 0) {\n        this.queue.put({\n          type: SpeechEventType.RECOGNITION_USAGE,\n          requestId,\n          recognitionUsage: { audioDuration: this.speechDuration },\n        });\n        this.speechDuration = 0;\n      }\n\n      this.queue.put({\n        type: SpeechEventType.FINAL_TRANSCRIPT,\n        requestId,\n        alternatives: [speechData],\n      });\n\n      if (this.speaking) {\n        this.speaking = false;\n        this.queue.put({ type: 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+{"version":3,"sources":["../../src/inference/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame } from '@livekit/rtc-node';\nimport type { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport {\n  STT as BaseSTT,\n  SpeechStream as BaseSpeechStream,\n  type SpeechData,\n  type SpeechEvent,\n  SpeechEventType,\n} from '../stt/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { type AudioBuffer, Event, Task, cancelAndWait, shortuuid, waitForAbort } from '../utils.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type DeepgramModels =\n  | 'deepgram'\n  | 'deepgram/nova-3'\n  | 'deepgram/nova-3-general'\n  | 'deepgram/nova-3-medical'\n  | 'deepgram/nova-2-conversationalai'\n  | 'deepgram/nova-2'\n  | 'deepgram/nova-2-general'\n  | 'deepgram/nova-2-medical'\n  | 'deepgram/nova-2-phonecall';\n\nexport type CartesiaModels = 'cartesia' | 'cartesia/ink-whisper';\n\nexport type AssemblyaiModels = 'assemblyai' | 'assemblyai/universal-streaming';\n\nexport interface CartesiaOptions {\n  min_volume?: number; // default: not specified\n  max_silence_duration_secs?: number; // default: not specified\n}\n\nexport interface DeepgramOptions {\n  filler_words?: boolean; // default: true\n  interim_results?: boolean; // default: true\n  endpointing?: number; // default: 25 (ms)\n  punctuate?: boolean; // default: false\n  smart_format?: boolean;\n  keywords?: Array<[string, number]>;\n  keyterms?: string[];\n  profanity_filter?: boolean;\n  numerals?: boolean;\n  mip_opt_out?: boolean;\n}\n\nexport interface AssemblyAIOptions {\n  format_turns?: boolean; // default: false\n  end_of_turn_confidence_threshold?: number; // default: 0.01\n  min_end_of_turn_silence_when_confident?: number; // default: 0\n  max_turn_silence?: number; // default: not specified\n  keyterms_prompt?: string[]; // default: not specified\n}\n\nexport type STTLanguages =\n  | 'multi'\n  | 'en'\n  | 'de'\n  | 'es'\n  | 'fr'\n  | 'ja'\n  | 'pt'\n  | 'zh'\n  | 'hi'\n  | AnyString;\n\ntype _STTModels = DeepgramModels | CartesiaModels | AssemblyaiModels;\n\nexport type STTModels = _STTModels | 'auto' | AnyString;\n\nexport type ModelWithLanguage = `${_STTModels}:${STTLanguages}` | STTModels;\n\nexport type STTOptions<TModel extends STTModels> = TModel extends DeepgramModels\n  ? DeepgramOptions\n  : TModel extends CartesiaModels\n    ? CartesiaOptions\n    : TModel extends AssemblyaiModels\n      ? 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modelOptions?: STTOptions<TModel>;\n  }) {\n    super({ streaming: true, interimResults: true });\n\n    const {\n      model,\n      language,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      sampleRate = DEFAULT_SAMPLE_RATE,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as STTOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    this.opts = {\n      model,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: 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{ language, connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const streamOpts = {\n      ...this.opts,\n      language: language ?? this.opts.language,\n    } as InferenceSTTOptions<TModel>;\n\n    const stream = new SpeechStream(this, streamOpts, connOptions);\n    this.streams.add(stream);\n\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    const params = {\n      settings: {\n        sample_rate: String(this.opts.sampleRate),\n        encoding: this.opts.encoding,\n        extra: this.opts.modelOptions,\n      },\n    } as Record<string, unknown>;\n\n    if (this.opts.model && this.opts.model !== 'auto') {\n      params.model = this.opts.model;\n    }\n\n    if (this.opts.language) {\n      (params.settings as Record<string, unknown>).language = this.opts.language;\n    }\n\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 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sttImpl;\n    this.connOptions = connOptions;\n  }\n\n  get label(): string {\n    return 'inference.SpeechStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceSTTOptions<TModel>, 'model' | 'language'>>): void {\n    this.opts = { ...this.opts, ...opts };\n    this.reconnectEvent.set();\n  }\n\n  protected async run(): Promise<void> {\n    while (true) {\n      // Create fresh resources for each connection attempt\n      let ws: WebSocket | null = null;\n      let closing = false;\n      let finalReceived = false;\n\n      type SttServerEvent = Record<string, any>;\n      const eventChannel = createStreamChannel<SttServerEvent>();\n\n      const resourceCleanup = () => {\n        if (closing) return;\n        closing = true;\n        eventChannel.close();\n        ws?.removeAllListeners();\n        ws?.close();\n      };\n\n      const createWsListener = async (ws: WebSocket, signal: AbortSignal) => {\n        return new Promise<void>((resolve, reject) => {\n          const onAbort = () => {\n            resourceCleanup();\n            reject(new Error('WebSocket connection aborted'));\n          };\n\n          signal.addEventListener('abort', onAbort, { once: true });\n\n          ws.on('message', (data) => {\n            const json = JSON.parse(data.toString()) as SttServerEvent;\n            eventChannel.write(json);\n          });\n\n          ws.on('error', (e) => {\n            this.#logger.error({ error: e }, 'WebSocket error');\n            resourceCleanup();\n            reject(e);\n          });\n\n          ws.on('close', (code: number) => {\n            resourceCleanup();\n\n            if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n            if (finalReceived) return resolve();\n\n            reject(\n              new APIStatusError({\n                message: 'LiveKit STT connection closed unexpectedly',\n                options: { statusCode: code },\n              }),\n            );\n          });\n        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eventChannel.stream();\n        const reader = serverEventStream.getReader();\n\n        try {\n          while (!this.closed && !signal.aborted) {\n            const result = await reader.read();\n            if (signal.aborted) return;\n            if (result.done) return;\n\n            const json = result.value;\n            const type = json.type as string | undefined;\n\n            switch (type) {\n              case 'session.created':\n              case 'session.finalized':\n                break;\n              case 'session.closed':\n                finalReceived = true;\n                resourceCleanup();\n                break;\n              case 'interim_transcript':\n                this.processTranscript(json, false);\n                break;\n              case 'final_transcript':\n                this.processTranscript(json, true);\n                break;\n              case 'error':\n                this.#logger.error({ error: json }, 'Received error from LiveKit 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=> createWsListener(ws!, signal), controller);\n        const recvTask = Task.from(({ signal }) => recv(signal), controller);\n        const waitReconnectTask = Task.from(\n          ({ signal }) => Promise.race([this.reconnectEvent.wait(), waitForAbort(signal)]),\n          controller,\n        );\n\n        try {\n          await Promise.race([\n            Promise.all([sendTask.result, wsListenerTask.result, recvTask.result]),\n            waitReconnectTask.result,\n          ]);\n\n          // If reconnect didn't trigger, tasks finished - exit loop\n          if (!waitReconnectTask.done) break;\n\n          // Reconnect triggered - clear event and continue loop\n          this.reconnectEvent.clear();\n        } finally {\n          // Cancel all tasks to ensure cleanup\n          await cancelAndWait(\n            [sendTask, wsListenerTask, recvTask, waitReconnectTask],\n            DEFAULT_CANCEL_TIMEOUT,\n          );\n          resourceCleanup();\n        }\n      } finally {\n  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SpeechEventType.END_OF_SPEECH });\n      }\n    } else {\n      this.queue.put({\n        type: SpeechEventType.INTERIM_TRANSCRIPT,\n        requestId,\n        alternatives: [speechData],\n      });\n    }\n  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package/dist/stt/stt.cjs

@@ -66,6 +66,9 @@ class STT extends import_node_events.EventEmitter {
     });
     return event;
   }
+  async close() {
+    return;
+  }
 }
 class SpeechStream {
   static FLUSH_SENTINEL = Symbol("FLUSH_SENTINEL");

package/dist/stt/stt.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n  protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   */\n  abstract stream(): SpeechStream;\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().then(() => this.queue.close()));\n  }\n\n  private async mainTask() {\n    for (let i = 0; i < this._connOptions.maxRetry + 1; i++) {\n      try {\n        return await this.run();\n      } catch (error) {\n        if (error instanceof APIError) {\n          const retryInterval = this._connOptions._intervalForRetry(i);\n\n          if (this._connOptions.maxRetry === 0 || !error.retryable) {\n            this.emitError({ error, recoverable: false });\n            throw error;\n          } else if (i === this._connOptions.maxRetry) {\n            this.emitError({ error, recoverable: false });\n            throw new APIConnectionError({\n              message: `failed to recognize speech after ${this._connOptions.maxRetry + 1} attempts`,\n              options: { retryable: false },\n            });\n          } else {\n            // Don't emit error event for recoverable errors during retry loop\n            // to avoid ERR_UNHANDLED_ERROR or premature session termination\n            this.logger.warn(\n              { tts: this.#stt.label, attempt: i + 1, error },\n              `failed to recognize speech, retrying in ${retryInterval}s`,\n            );\n          }\n\n          if (retryInterval > 0) {\n            await delay(retryInterval);\n          }\n        } else {\n          this.emitError({ error: toError(error), recoverable: false });\n          throw error;\n        }\n      }\n    }\n  }\n\n  private emitError({ error, recoverable }: { error: Error; recoverable: boolean }) {\n    this.#stt.emit('error', {\n      type: 'stt_error',\n      timestamp: Date.now(),\n      label: this.#stt.label,\n      error,\n      recoverable,\n    });\n  }\n\n  protected async pumpInput() {\n    // TODO(AJS-35): Implement STT with webstreams API\n    const inputStream = this.deferredInputStream.stream;\n    const reader = inputStream.getReader();\n\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        this.pushFrame(value);\n      }\n    } catch (error) {\n      this.logger.error('Error in STTStream mainTask:', error);\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    for await (const event of this.queue) {\n      this.output.put(event);\n      if (event.type !== SpeechEventType.RECOGNITION_USAGE) continue;\n      const metrics: STTMetrics = {\n        type: 'stt_metrics',\n        timestamp: Date.now(),\n        requestId: event.requestId!,\n        durationMs: 0,\n        label: this.#stt.label,\n        audioDurationMs: Math.round(event.recognitionUsage!.audioDuration * 1000),\n        streamed: true,\n      };\n      this.#stt.emit('metrics_collected', metrics);\n    }\n    this.output.close();\n  }\n\n  protected abstract run(): Promise<void>;\n\n  updateInputStream(audioStream: ReadableStream<AudioFrame>) {\n    this.deferredInputStream.setSource(audioStream);\n  }\n\n  detachInputStream() {\n    this.deferredInputStream.detachSource();\n  }\n\n  /** Push an audio frame to the STT */\n  pushFrame(frame: AudioFrame) {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n\n    if (this.neededSampleRate && frame.sampleRate !== this.neededSampleRate) {\n      if (!this.resampler) {\n        this.resampler = new AudioResampler(frame.sampleRate, this.neededSampleRate);\n      }\n    }\n\n    if (this.resampler) {\n      const frames = this.resampler.push(frame);\n      for (const frame of frames) {\n        this.input.put(frame);\n      }\n    } else {\n      this.input.put(frame);\n    }\n  }\n\n  /** Flush the STT, causing it to process all pending text */\n  flush() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.put(SpeechStream.FLUSH_SENTINEL);\n  }\n\n  /** Mark the input as ended and forbid additional pushes */\n  endInput() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.close();\n  }\n\n  next(): Promise<IteratorResult<SpeechEvent>> {\n    return this.output.next();\n  }\n\n  /** Close both the input and output of the STT stream */\n  close() {\n    this.input.close();\n    this.queue.close();\n    this.output.close();\n    this.closed = true;\n  }\n\n  [Symbol.asyncIterator](): SpeechStream {\n    return this;\n  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+{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n  protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   */\n  abstract stream(): SpeechStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. 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error event for recoverable errors during retry loop\n            // to avoid ERR_UNHANDLED_ERROR or premature session termination\n            this.logger.warn(\n              { tts: this.#stt.label, attempt: i + 1, error },\n              `failed to recognize speech, retrying in ${retryInterval}s`,\n            );\n          }\n\n          if (retryInterval > 0) {\n            await delay(retryInterval);\n          }\n        } else {\n          this.emitError({ error: toError(error), recoverable: false });\n          throw error;\n        }\n      }\n    }\n  }\n\n  private emitError({ error, recoverable }: { error: Error; recoverable: boolean }) {\n    this.#stt.emit('error', {\n      type: 'stt_error',\n      timestamp: Date.now(),\n      label: this.#stt.label,\n      error,\n      recoverable,\n    });\n  }\n\n  protected async pumpInput() {\n    // TODO(AJS-35): Implement STT with webstreams API\n    const inputStream = this.deferredInputStream.stream;\n    const reader = 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package/dist/stt/stt.d.cts

@@ -98,6 +98,7 @@ export declare abstract class STT extends STT_base {
      * transcriptions
      */
     abstract stream(): SpeechStream;
+    close(): Promise<void>;
 }
 /**
  * An instance of a speech-to-text stream, as an asynchronous iterable iterator.

package/dist/stt/stt.d.ts

@@ -98,6 +98,7 @@ export declare abstract class STT extends STT_base {
      * transcriptions
      */
     abstract stream(): SpeechStream;
+    close(): Promise<void>;
 }
 /**
  * An instance of a speech-to-text stream, as an asynchronous iterable iterator.

package/dist/stt/stt.d.ts.map

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

@@ -41,6 +41,9 @@ class STT extends EventEmitter {
     });
     return event;
   }
+  async close() {
+    return;
+  }
 }
 class SpeechStream {
   static FLUSH_SENTINEL = Symbol("FLUSH_SENTINEL");