@livekit/agents-plugin-cartesia 0.0.0-next-20260624041820

package/dist/stt.cjs

@@ -0,0 +1,422 @@
+"use strict";
+var __defProp = Object.defineProperty;
+var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
+var __getOwnPropNames = Object.getOwnPropertyNames;
+var __hasOwnProp = Object.prototype.hasOwnProperty;
+var __export = (target, all) => {
+  for (var name in all)
+    __defProp(target, name, { get: all[name], enumerable: true });
+};
+var __copyProps = (to, from, except, desc) => {
+  if (from && typeof from === "object" || typeof from === "function") {
+    for (let key of __getOwnPropNames(from))
+      if (!__hasOwnProp.call(to, key) && key !== except)
+        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
+  }
+  return to;
+};
+var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
+var stt_exports = {};
+__export(stt_exports, {
+  STT: () => STT,
+  SpeechStream: () => SpeechStream
+});
+module.exports = __toCommonJS(stt_exports);
+var import_agents = require("@livekit/agents");
+var import_ws = require("ws");
+var import_models = require("./models.cjs");
+const AUTHORIZATION_HEADER = "X-API-Key";
+const VERSION_HEADER = "Cartesia-Version";
+const REQUEST_ID_HEADER = "cartesia-request-id";
+const API_VERSION = "2026-03-01";
+const DRAIN_TIMEOUT_MS = 5e3;
+const KEEPALIVE_INTERVAL_MS = 3e4;
+const defaultSTTOptions = {
+  model: "ink-2",
+  /** recommended default */
+  sampleRate: 16e3,
+  /** recommended default */
+  audioChunkDurationMS: 160,
+  baseUrl: "https://api.cartesia.ai",
+  language: "en"
+};
+function mergeSTTOptions(base, override) {
+  return {
+    apiKey: override.apiKey ?? base.apiKey,
+    baseUrl: override.baseUrl ?? base.baseUrl,
+    model: override.model ?? base.model,
+    sampleRate: override.sampleRate ?? base.sampleRate,
+    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,
+    language: override.language !== void 0 ? (0, import_agents.normalizeLanguage)(override.language) : base.language
+  };
+}
+function resolveSTTModel(language) {
+  return (0, import_agents.getBaseLanguage)(language) === "en" ? "ink-2" : "ink-whisper";
+}
+class STT extends import_agents.stt.STT {
+  #opts;
+  constructor(opts = {}) {
+    super({
+      streaming: true,
+      interimResults: true,
+      alignedTranscript: false,
+      diarization: false
+    });
+    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;
+    if (!apiKey) {
+      throw new Error(
+        "Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY"
+      );
+    }
+    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);
+    if (opts.model === void 0) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
+  }
+  get label() {
+    return "cartesia.STT";
+  }
+  get model() {
+    return this.#opts.model;
+  }
+  get provider() {
+    return "Cartesia";
+  }
+  async _recognize() {
+    throw new Error("Cartesia STT does not support batch recognition, use stream() instead");
+  }
+  stream(options) {
+    return new SpeechStream(this, this.#opts, options == null ? void 0 : options.connOptions);
+  }
+  updateOptions(opts) {
+    this.#opts = mergeSTTOptions(this.#opts, opts);
+    if (opts.language !== void 0 && opts.model === void 0) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
+  }
+}
+class SpeechStream extends import_agents.stt.SpeechStream {
+  #opts;
+  #logger = (0, import_agents.log)();
+  #ws = null;
+  #requestId;
+  #speaking = false;
+  #currentTranscript = "";
+  #speechDuration = 0;
+  #closingWs = false;
+  constructor(sttInstance, opts, connOptions) {
+    super(sttInstance, opts.sampleRate, connOptions);
+    this.#opts = { ...opts };
+  }
+  get label() {
+    return "cartesia.SpeechStream";
+  }
+  async run() {
+    var _a;
+    if (this.closed) {
+      this.close();
+      return;
+    }
+    this.#speaking = false;
+    this.#currentTranscript = "";
+    this.#speechDuration = 0;
+    this.#closingWs = false;
+    let keepaliveInterval;
+    const abortController = new AbortController();
+    try {
+      const url = this.#getCartesiaUrl();
+      this.#logger.debug(`Connecting to Cartesia STT: ${url}`);
+      const ws = new import_ws.WebSocket(url, {
+        headers: {
+          [AUTHORIZATION_HEADER]: this.#opts.apiKey,
+          [VERSION_HEADER]: API_VERSION
+        }
+      });
+      this.#ws = ws;
+      ws.once("upgrade", (response) => {
+        const headerValue = response.headers[REQUEST_ID_HEADER];
+        const requestId = Array.isArray(headerValue) ? headerValue[0] : headerValue;
+        if (requestId) {
+          this.#requestId = requestId;
+          this.#logger.debug({ cartesiaRequestId: requestId }, "Cartesia STT WebSocket connected");
+        }
+      });
+      await new Promise((resolve, reject) => {
+        const onOpen = () => {
+          ws.off("error", onError);
+          resolve();
+        };
+        const onError = (err) => {
+          ws.off("open", onOpen);
+          reject(new import_agents.APIConnectionError({ message: err.message, options: { retryable: true } }));
+        };
+        ws.once("open", onOpen);
+        ws.once("error", onError);
+      });
+      let firstError;
+      const stopPeer = (err) => {
+        if (firstError === void 0) firstError = err;
+        abortController.abort();
+        if (ws.readyState === import_ws.WebSocket.OPEN) ws.close();
+      };
+      const sendPromise = this.#sendTask(ws, abortController.signal).catch(stopPeer);
+      const recvPromise = this.#recvTask(ws).catch(stopPeer);
+      keepaliveInterval = setInterval(() => {
+        try {
+          if (ws.readyState === import_ws.WebSocket.OPEN) {
+            ws.ping();
+          }
+        } catch {
+        }
+      }, KEEPALIVE_INTERVAL_MS);
+      await Promise.all([sendPromise, recvPromise]);
+      if (firstError !== void 0) throw firstError;
+    } catch (error) {
+      this.#logger.error("Cartesia STT stream error", { error });
+      throw error;
+    } finally {
+      abortController.abort();
+      if (keepaliveInterval) clearInterval(keepaliveInterval);
+      if (((_a = this.#ws) == null ? void 0 : _a.readyState) === import_ws.WebSocket.OPEN) {
+        this.#ws.close();
+      }
+    }
+    this.close();
+  }
+  async #sendTask(ws, abortSignal) {
+    const samplesPerChunk = Math.floor(
+      this.#opts.sampleRate * this.#opts.audioChunkDurationMS / 1e3
+    );
+    const audioBstream = new import_agents.AudioByteStream(this.#opts.sampleRate, 1, samplesPerChunk);
+    let hasEnded = false;
+    const iterator = this.input[Symbol.asyncIterator]();
+    const abortPromise = (0, import_agents.waitForAbort)(abortSignal);
+    while (true) {
+      const result = await Promise.race([iterator.next(), abortPromise]);
+      if (result === void 0) return;
+      if (result.done) {
+        hasEnded = true;
+      } else {
+        const data = result.value;
+        const frames = [];
+        if (data === SpeechStream.FLUSH_SENTINEL) {
+          frames.push(...audioBstream.flush());
+        } else {
+          frames.push(...audioBstream.write(data.data));
+        }
+        for (const frame of frames) {
+          this.#speechDuration += (0, import_agents.calculateAudioDurationSeconds)(frame);
+          if (ws.readyState === import_ws.WebSocket.OPEN) {
+            ws.send(frame.data);
+          }
+        }
+      }
+      if (hasEnded) break;
+    }
+    if (ws.readyState === import_ws.WebSocket.OPEN) {
+      this.#closingWs = true;
+      this.#logger.debug("Sending close message to Cartesia STT");
+      ws.send(JSON.stringify({ type: "close" }));
+      const drainTimer = setTimeout(() => {
+        this.#logger.warn(
+          `Cartesia STT did not close within ${DRAIN_TIMEOUT_MS}ms after done; forcing close`
+        );
+        if (ws.readyState === import_ws.WebSocket.OPEN || ws.readyState === import_ws.WebSocket.CLOSING) {
+          ws.terminate();
+        }
+      }, DRAIN_TIMEOUT_MS);
+      ws.once("close", () => clearTimeout(drainTimer));
+    }
+  }
+  async #recvTask(ws) {
+    let pendingError = void 0;
+    return new Promise((resolve, reject) => {
+      const settle = (err) => {
+        if (err) reject(err);
+        else resolve();
+      };
+      ws.on("message", (data, isBinary) => {
+        if (isBinary) {
+          this.#logger.warn("Received unexpected binary message from Cartesia STT");
+          return;
+        }
+        let msg;
+        try {
+          msg = JSON.parse(data.toString());
+        } catch (error) {
+          this.#logger.error("Failed to parse Cartesia STT message", { error });
+          return;
+        }
+        try {
+          if (msg.type === "error") {
+            this.#logger.error("Cartesia sent an error", msg);
+            if (msg.status_code === void 0 || msg.status_code >= 500) {
+              pendingError = new import_agents.APIConnectionError({
+                message: msg.message || msg.title,
+                options: {
+                  retryable: true
+                }
+              });
+              ws.close();
+              return;
+            }
+          } else {
+            this.#processStreamEvent(msg);
+          }
+        } catch (error) {
+          this.#logger.error("Failed to process Cartesia STT message", { error });
+        }
+      });
+      ws.on("close", (code, reason) => {
+        if (this.#speaking) {
+          if (this.#currentTranscript) {
+            this.#sendTranscriptEvent(
+              import_agents.stt.SpeechEventType.FINAL_TRANSCRIPT,
+              this.#currentTranscript
+            );
+          }
+          if (!this.queue.closed)
+            this.queue.put({
+              type: import_agents.stt.SpeechEventType.END_OF_SPEECH,
+              requestId: this.#requestId
+            });
+          this.#logger.debug("Cartesia STT END_OF_SPEECH");
+          if (this.#speechDuration > 0 && !this.queue.closed) {
+            this.queue.put({
+              type: import_agents.stt.SpeechEventType.RECOGNITION_USAGE,
+              requestId: this.#requestId,
+              recognitionUsage: { audioDuration: this.#speechDuration }
+            });
+            this.#speechDuration = 0;
+          }
+          this.#speaking = false;
+          this.#currentTranscript = "";
+        }
+        if (pendingError) {
+          settle(pendingError);
+          return;
+        }
+        if (this.#closingWs || this.closed) {
+          settle();
+          return;
+        }
+        this.#logger.warn(`Cartesia STT WebSocket closed: ${code} ${reason.toString()}`);
+        settle(
+          new import_agents.APIConnectionError({
+            message: `Cartesia STT connection closed unexpectedly (code=${code})`,
+            options: { retryable: true }
+          })
+        );
+      });
+      ws.on("error", (err) => {
+        if (this.closed) {
+          settle();
+          return;
+        }
+        settle(new import_agents.APIConnectionError({ message: err.message, options: { retryable: true } }));
+      });
+    });
+  }
+  #processStreamEvent(data) {
+    if (data.request_id) {
+      this.#requestId = data.request_id;
+    }
+    switch (data.type) {
+      case "connected":
+        return;
+      case "turn.start": {
+        if (this.#speaking) return;
+        this.#speaking = true;
+        this.#currentTranscript = "";
+        if (!this.queue.closed)
+          this.queue.put({
+            type: import_agents.stt.SpeechEventType.START_OF_SPEECH,
+            requestId: this.#requestId
+          });
+        this.#logger.debug("Cartesia STT START_OF_SPEECH");
+        return;
+      }
+      case "turn.update": {
+        if (!this.#speaking) return;
+        const transcript = data.transcript;
+        if (!transcript) return;
+        if (this.#currentTranscript === transcript) return;
+        this.#currentTranscript = transcript;
+        this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.INTERIM_TRANSCRIPT, transcript);
+        return;
+      }
+      case "turn.eager_end": {
+        if (!this.#speaking) return;
+        const transcript = data.transcript || this.#currentTranscript;
+        if (!transcript) return;
+        this.#currentTranscript = transcript;
+        this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.PREFLIGHT_TRANSCRIPT, transcript);
+        return;
+      }
+      case "turn.resume":
+        if (!this.#speaking || !this.#currentTranscript) return;
+        this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.INTERIM_TRANSCRIPT, this.#currentTranscript);
+        return;
+      case "turn.end": {
+        if (!this.#speaking) return;
+        const transcript = data.transcript || this.#currentTranscript;
+        this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.FINAL_TRANSCRIPT, transcript);
+        this.#speaking = false;
+        if (!this.queue.closed)
+          this.queue.put({
+            type: import_agents.stt.SpeechEventType.END_OF_SPEECH,
+            requestId: this.#requestId
+          });
+        this.#logger.debug("Cartesia STT END_OF_SPEECH");
+        this.#currentTranscript = "";
+        if (this.#speechDuration > 0 && !this.queue.closed) {
+          this.queue.put({
+            type: import_agents.stt.SpeechEventType.RECOGNITION_USAGE,
+            requestId: this.#requestId,
+            recognitionUsage: { audioDuration: this.#speechDuration }
+          });
+          this.#speechDuration = 0;
+        }
+        return;
+      }
+      default:
+        this.#logger.warn("received unexpected message from Cartesia STT", { data });
+    }
+  }
+  #sendTranscriptEvent(eventType, transcript) {
+    if (this.queue.closed) return;
+    this.queue.put({
+      type: eventType,
+      requestId: this.#requestId,
+      alternatives: [
+        {
+          language: (0, import_agents.asLanguageCode)(this.#opts.language),
+          text: transcript,
+          startTime: 0,
+          endTime: 0,
+          confidence: 0
+        }
+      ]
+    });
+  }
+  #getCartesiaUrl() {
+    const params = new URLSearchParams({
+      model: this.#opts.model,
+      sample_rate: this.#opts.sampleRate.toString(),
+      encoding: import_models.AUDIO_ENCODING
+    });
+    const wsBase = this.#opts.baseUrl.replace(/^http/, "ws");
+    return `${wsBase}/stt/turns/websocket?${params.toString()}`;
+  }
+  close() {
+    var _a;
+    super.close();
+    (_a = this.#ws) == null ? void 0 : _a.close();
+  }
+}
+// Annotate the CommonJS export names for ESM import in node:
+0 && (module.exports = {
+  STT,
+  SpeechStream
+});
+//# sourceMappingURL=stt.cjs.map

package/dist/stt.cjs.map

@@ -0,0 +1 @@
+{"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  AudioByteStream,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { IncomingMessage } from 'node:http';\nimport { WebSocket } from 'ws';\nimport { AUDIO_ENCODING, type STTModel } from './models.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst REQUEST_ID_HEADER = 'cartesia-request-id';\nconst API_VERSION = '2026-03-01';\nconst DRAIN_TIMEOUT_MS = 5000;\nconst KEEPALIVE_INTERVAL_MS = 30000;\n\n/**\n * Fires once when the WebSocket connection is established.\n *\n * You do not need to wait for this event before sending audio.\n */\ntype STTConnectedEvent = {\n  type: 'connected';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Model predicts {@link stt.SpeechEventType.START_OF_SPEECH}.\n */\ntype STTTurnStartEvent = {\n  type: 'turn.start';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires repeatedly as the model transcribes the current user turn.\n *\n * Can be used for {@link stt.SpeechEventType.INTERIM_TRANSCRIPT}.\n */\ntype STTTurnUpdateEvent = {\n  type: 'turn.update';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires when the model predicts that the user might be done speaking.\n *\n * Can be used for {@link stt.SpeechEventType.PREFLIGHT_TRANSCRIPT}.\n */\ntype STTTurnEagerEndEvent = {\n  type: 'turn.eager_end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires after {@link STTTurnEagerEndEvent} if the user turn has not actually ended.\n */\ntype STTTurnResumeEvent = {\n  type: 'turn.resume';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Marks the end of a user turn.\n *\n * This is used for {@link stt.SpeechEventType.END_OF_SPEECH} and {@link stt.SpeechEventType.FINAL_TRANSCRIPT}.\n */\ntype STTTurnEndEvent = {\n  type: 'turn.end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\ntype STTErrorEvent = {\n  type: 'error';\n  error_code?: string;\n  status_code?: number;\n  title?: string;\n  message?: string;\n  doc_url?: string;\n  request_id?: string;\n};\n\n/**\n * Server-sent event on the `/stt/turns/websocket` endpoint.\n *\n * See https://docs.cartesia.ai/api-reference/stt/turns/websocket.\n */\ntype STTEventMessage =\n  | STTConnectedEvent\n  | STTTurnStartEvent\n  | STTTurnUpdateEvent\n  | STTTurnEagerEndEvent\n  | STTTurnResumeEvent\n  | STTTurnEndEvent\n  | STTErrorEvent;\n\nexport type STTOptions = {\n  apiKey: string;\n  // eslint-disable-next-line @typescript-eslint/ban-types\n  model: STTModel | (string & {});\n  sampleRate: number;\n  baseUrl: string;\n  audioChunkDurationMS: number;\n  language: string;\n};\n\nconst defaultSTTOptions = {\n  model: 'ink-2' satisfies STTModel,\n  /** recommended default */\n  sampleRate: 16_000,\n  /** recommended default */\n  audioChunkDurationMS: 160,\n  baseUrl: 'https://api.cartesia.ai',\n  language: 'en',\n};\n\nfunction mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {\n  return {\n    apiKey: override.apiKey ?? base.apiKey,\n    baseUrl: override.baseUrl ?? base.baseUrl,\n    model: override.model ?? base.model,\n    sampleRate: override.sampleRate ?? base.sampleRate,\n    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,\n    language:\n      override.language !== undefined ? normalizeLanguage(override.language) : base.language,\n  };\n}\n\n/**\n * Resolve the STT model from the configured language when the caller did not\n * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports\n * English, so non-English languages route to the multilingual `ink-whisper`.\n */\nfunction resolveSTTModel(language: string): STTModel {\n  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';\n}\n\n/**\n * Cartesia speech to text.\n *\n * Supports:\n *  - Streaming\n *  - Turn detection\n *  - Interim results\n *\n * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest\n *\n * @example\n * ```typescript\n * import { voice } from '@livekit/agents';\n * import { STT, TTS } from '@livekit/agents-plugin-cartesia';\n *\n * const session = new voice.AgentSession({\n *   stt: new STT(),\n *   llm: new LLM(), // choose your favorite LLM\n *   tts: new TTS(),\n *   turnHandling: { turnDetection: 'stt' },\n * });\n * ```\n */\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: false,\n      diarization: false,\n    });\n\n    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;\n    if (!apiKey) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);\n\n    // Route to the multilingual model for non-English languages unless the\n    // caller pinned a model. Matches the Python plugin's language→model logic.\n    if (opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n\n  override get label(): string {\n    return 'cartesia.STT';\n  }\n\n  override get model(): string {\n    return this.#opts.model;\n  }\n\n  override get provider(): string {\n    return 'Cartesia';\n  }\n\n  protected override async _recognize(): Promise<stt.SpeechEvent> {\n    throw new Error('Cartesia STT does not support batch recognition, use stream() instead');\n  }\n\n  override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = mergeSTTOptions(this.#opts, opts);\n\n    // Keep the model in sync with a newly set language (e.g. switching to a\n    // non-English language must move off the English-only ink-2), unless the\n    // caller pinned a model in the same call. Mirrors the constructor.\n    if (opts.language !== undefined && opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #ws: WebSocket | null = null;\n  #requestId: string | undefined;\n  #speaking = false;\n  #currentTranscript = '';\n  #speechDuration = 0;\n  #closingWs = false;\n\n  constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(sttInstance, opts.sampleRate, connOptions);\n    this.#opts = { ...opts };\n  }\n\n  override get label(): string {\n    return 'cartesia.SpeechStream';\n  }\n\n  protected override async run() {\n    if (this.closed) {\n      this.close();\n      return;\n    }\n\n    this.#speaking = false;\n    this.#currentTranscript = '';\n    this.#speechDuration = 0;\n    this.#closingWs = false;\n\n    let keepaliveInterval: NodeJS.Timeout | undefined;\n    const abortController = new AbortController();\n\n    try {\n      const url = this.#getCartesiaUrl();\n      this.#logger.debug(`Connecting to Cartesia STT: ${url}`);\n\n      const ws = new WebSocket(url, {\n        headers: {\n          [AUTHORIZATION_HEADER]: this.#opts.apiKey,\n          [VERSION_HEADER]: API_VERSION,\n        },\n      });\n      this.#ws = ws;\n\n      // Cartesia returns the request id on the WS upgrade response, before any\n      // turn events are sent. Capture it so logs and metrics emitted on the\n      // connection (and any pre-`turn.start` errors) are attributable.\n      ws.once('upgrade', (response: IncomingMessage) => {\n        const headerValue = response.headers[REQUEST_ID_HEADER];\n        const requestId = Array.isArray(headerValue) ? headerValue[0] : headerValue;\n        if (requestId) {\n          this.#requestId = requestId;\n          this.#logger.debug({ cartesiaRequestId: requestId }, 'Cartesia STT WebSocket connected');\n        }\n      });\n\n      await new Promise<void>((resolve, reject) => {\n        const onOpen = () => {\n          ws.off('error', onError);\n          resolve();\n        };\n        const onError = (err: Error) => {\n          ws.off('open', onOpen);\n          reject(new APIConnectionError({ message: err.message, options: { retryable: true } }));\n        };\n\n        ws.once('open', onOpen);\n        ws.once('error', onError);\n      });\n\n      // If one task fails, abort its peer and close the WS so the peer can\n      // exit before we re-throw. Otherwise a dangling task would survive\n      // into the next retry attempt (sendTask blocked on input.next() would\n      // steal frames; recvTask's close handler could still push events to\n      // the queue and surface an unhandled rejection).\n      let firstError: unknown;\n      const stopPeer = (err: unknown) => {\n        if (firstError === undefined) firstError = err;\n        abortController.abort();\n        if (ws.readyState === WebSocket.OPEN) ws.close();\n      };\n      const sendPromise = this.#sendTask(ws, abortController.signal).catch(stopPeer);\n      const recvPromise = this.#recvTask(ws).catch(stopPeer);\n\n      keepaliveInterval = setInterval(() => {\n        try {\n          if (ws.readyState === WebSocket.OPEN) {\n            ws.ping();\n          }\n        } catch {\n          // ignore\n        }\n      }, KEEPALIVE_INTERVAL_MS);\n      await Promise.all([sendPromise, recvPromise]);\n      if (firstError !== undefined) throw firstError;\n    } catch (error) {\n      this.#logger.error('Cartesia STT stream error', { error });\n      throw error;\n    } finally {\n      abortController.abort();\n      if (keepaliveInterval) clearInterval(keepaliveInterval);\n      if (this.#ws?.readyState === WebSocket.OPEN) {\n        this.#ws.close();\n      }\n    }\n\n    this.close();\n  }\n\n  async #sendTask(ws: WebSocket, abortSignal: AbortSignal) {\n    const samplesPerChunk = Math.floor(\n      (this.#opts.sampleRate * this.#opts.audioChunkDurationMS) / 1000,\n    );\n    const audioBstream = new AudioByteStream(this.#opts.sampleRate, 1, samplesPerChunk);\n\n    let hasEnded = false;\n    const iterator = this.input[Symbol.asyncIterator]();\n    const abortPromise = waitForAbort(abortSignal);\n\n    while (true) {\n      const result = await Promise.race([iterator.next(), abortPromise]);\n\n      if (result === undefined) return; // aborted\n\n      if (result.done) {\n        hasEnded = true;\n      } else {\n        const data = result.value;\n        const frames: AudioFrame[] = [];\n\n        if (data === SpeechStream.FLUSH_SENTINEL) {\n          frames.push(...audioBstream.flush());\n        } else {\n          // Pass the typed array directly so AudioByteStream uses byteOffset/\n          // byteLength — `.buffer` would include foreign bytes when rtc-node\n          // hands us a view over a pooled allocator.\n          frames.push(...audioBstream.write(data.data));\n        }\n\n        for (const frame of frames) {\n          this.#speechDuration += calculateAudioDurationSeconds(frame);\n          if (ws.readyState === WebSocket.OPEN) {\n            ws.send(frame.data);\n          }\n        }\n      }\n\n      if (hasEnded) break;\n    }\n\n    if (ws.readyState === WebSocket.OPEN) {\n      this.#closingWs = true;\n      this.#logger.debug('Sending close message to Cartesia STT');\n      ws.send(JSON.stringify({ type: 'close' }));\n\n      // After `close`, the server should flush any pending turn events and\n      // close the socket. If it hangs, force the issue rather than letting\n      // the stream wait forever.\n      const drainTimer = setTimeout(() => {\n        this.#logger.warn(\n          `Cartesia STT did not close within ${DRAIN_TIMEOUT_MS}ms after done; forcing close`,\n        );\n        if (ws.readyState === WebSocket.OPEN || ws.readyState === WebSocket.CLOSING) {\n          ws.terminate();\n        }\n      }, DRAIN_TIMEOUT_MS);\n      ws.once('close', () => clearTimeout(drainTimer));\n    }\n  }\n\n  async #recvTask(ws: WebSocket) {\n    let pendingError: Error | undefined = undefined;\n\n    return new Promise<void>((resolve, reject) => {\n      const settle = (err?: Error) => {\n        if (err) reject(err);\n        else resolve();\n      };\n\n      ws.on('message', (data: Buffer, isBinary: boolean) => {\n        if (isBinary) {\n          this.#logger.warn('Received unexpected binary message from Cartesia STT');\n          return;\n        }\n        let msg: STTEventMessage;\n        try {\n          msg = JSON.parse(data.toString());\n        } catch (error) {\n          this.#logger.error('Failed to parse Cartesia STT message', { error });\n          return;\n        }\n        try {\n          if (msg.type === 'error') {\n            this.#logger.error('Cartesia sent an error', msg);\n\n            // do not close the websocket on bad requests since that may be caused by invalid messages\n            if (msg.status_code === undefined || msg.status_code >= 500) {\n              // Defer until the WS is fully closed so we don't race the `close`\n              // handler with a rejected `recvTask`. Close the socket explicitly\n              // in case the server doesn't, otherwise recv would hang.\n              pendingError = new APIConnectionError({\n                message: msg.message || msg.title,\n                options: {\n                  retryable: true,\n                },\n              });\n              ws.close();\n              return;\n            }\n          } else {\n            this.#processStreamEvent(msg);\n          }\n        } catch (error) {\n          this.#logger.error('Failed to process Cartesia STT message', { error });\n        }\n      });\n\n      ws.on('close', (code, reason) => {\n        // If a turn was in progress, close it out so consumers see a balanced\n        // START_OF_SPEECH/END_OF_SPEECH pair and don't carry stale turn state\n        // into the next reconnect.\n        if (this.#speaking) {\n          if (this.#currentTranscript) {\n            this.#sendTranscriptEvent(\n              stt.SpeechEventType.FINAL_TRANSCRIPT,\n              this.#currentTranscript,\n            );\n          }\n          if (!this.queue.closed)\n            this.queue.put({\n              type: stt.SpeechEventType.END_OF_SPEECH,\n              requestId: this.#requestId,\n            });\n          this.#logger.debug('Cartesia STT END_OF_SPEECH');\n          if (this.#speechDuration > 0 && !this.queue.closed) {\n            this.queue.put({\n              type: stt.SpeechEventType.RECOGNITION_USAGE,\n              requestId: this.#requestId,\n              recognitionUsage: { audioDuration: this.#speechDuration },\n            });\n            this.#speechDuration = 0;\n          }\n          this.#speaking = false;\n          this.#currentTranscript = '';\n        }\n\n        if (pendingError) {\n          settle(pendingError);\n          return;\n        }\n        if (this.#closingWs || this.closed) {\n          settle();\n          return;\n        }\n        this.#logger.warn(`Cartesia STT WebSocket closed: ${code} ${reason.toString()}`);\n        settle(\n          new APIConnectionError({\n            message: `Cartesia STT connection closed unexpectedly (code=${code})`,\n            options: { retryable: true },\n          }),\n        );\n      });\n\n      ws.on('error', (err) => {\n        if (this.closed) {\n          settle();\n          return;\n        }\n        settle(new APIConnectionError({ message: err.message, options: { retryable: true } }));\n      });\n    });\n  }\n\n  #processStreamEvent(data: Exclude<STTEventMessage, { type: 'error' }>) {\n    if (data.request_id) {\n      this.#requestId = data.request_id;\n    }\n\n    switch (data.type) {\n      case 'connected':\n        return;\n\n      case 'turn.start': {\n        if (this.#speaking) return;\n        this.#speaking = true;\n        this.#currentTranscript = '';\n        if (!this.queue.closed)\n          this.queue.put({\n            type: stt.SpeechEventType.START_OF_SPEECH,\n            requestId: this.#requestId,\n          });\n        this.#logger.debug('Cartesia STT START_OF_SPEECH');\n        return;\n      }\n\n      case 'turn.update': {\n        if (!this.#speaking) return;\n        const transcript = data.transcript;\n        if (!transcript) return;\n        // Only emit interim updates when the cumulative transcript actually\n        // changed; this avoids canceling preflight generation needlessly.\n        if (this.#currentTranscript === transcript) return;\n        this.#currentTranscript = transcript;\n        this.#sendTranscriptEvent(stt.SpeechEventType.INTERIM_TRANSCRIPT, transcript);\n        return;\n      }\n\n      case 'turn.eager_end': {\n        if (!this.#speaking) return;\n        const transcript = data.transcript || this.#currentTranscript;\n        if (!transcript) return;\n        this.#currentTranscript = transcript;\n        this.#sendTranscriptEvent(stt.SpeechEventType.PREFLIGHT_TRANSCRIPT, transcript);\n        return;\n      }\n\n      case 'turn.resume':\n        // turn.resume has no transcript; re-emit the latest cumulative transcript\n        // as an interim event so the pipeline cancels the pending preflight.\n        if (!this.#speaking || !this.#currentTranscript) return;\n        this.#sendTranscriptEvent(stt.SpeechEventType.INTERIM_TRANSCRIPT, this.#currentTranscript);\n        return;\n\n      case 'turn.end': {\n        if (!this.#speaking) return;\n        const transcript = data.transcript || this.#currentTranscript;\n\n        this.#sendTranscriptEvent(stt.SpeechEventType.FINAL_TRANSCRIPT, transcript);\n\n        this.#speaking = false;\n        if (!this.queue.closed)\n          this.queue.put({\n            type: stt.SpeechEventType.END_OF_SPEECH,\n            requestId: this.#requestId,\n          });\n        this.#logger.debug('Cartesia STT END_OF_SPEECH');\n        this.#currentTranscript = '';\n\n        if (this.#speechDuration > 0 && !this.queue.closed) {\n          this.queue.put({\n            type: stt.SpeechEventType.RECOGNITION_USAGE,\n            requestId: this.#requestId,\n            recognitionUsage: { audioDuration: this.#speechDuration },\n          });\n          this.#speechDuration = 0;\n        }\n\n        return;\n      }\n\n      default:\n        this.#logger.warn('received unexpected message from Cartesia STT', { data });\n    }\n  }\n\n  #sendTranscriptEvent(eventType: stt.SpeechEventType, transcript: string) {\n    if (this.queue.closed) return;\n    this.queue.put({\n      type: eventType,\n      requestId: this.#requestId,\n      alternatives: [\n        {\n          language: asLanguageCode(this.#opts.language),\n          text: transcript,\n          startTime: 0,\n          endTime: 0,\n          confidence: 0,\n        },\n      ],\n    });\n  }\n\n  #getCartesiaUrl(): string {\n    // The Cartesia /stt/turns/websocket endpoint only accepts model, sample_rate\n    // and encoding — there is no `language` query param. 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package/dist/stt.d.cts

@@ -0,0 +1,53 @@
+import { type APIConnectOptions, stt } from '@livekit/agents';
+import { type STTModel } from './models.js';
+export type STTOptions = {
+    apiKey: string;
+    model: STTModel | (string & {});
+    sampleRate: number;
+    baseUrl: string;
+    audioChunkDurationMS: number;
+    language: string;
+};
+/**
+ * Cartesia speech to text.
+ *
+ * Supports:
+ *  - Streaming
+ *  - Turn detection
+ *  - Interim results
+ *
+ * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest
+ *
+ * @example
+ * ```typescript
+ * import { voice } from '@livekit/agents';
+ * import { STT, TTS } from '@livekit/agents-plugin-cartesia';
+ *
+ * const session = new voice.AgentSession({
+ *   stt: new STT(),
+ *   llm: new LLM(), // choose your favorite LLM
+ *   tts: new TTS(),
+ *   turnHandling: { turnDetection: 'stt' },
+ * });
+ * ```
+ */
+export declare class STT extends stt.STT {
+    #private;
+    constructor(opts?: Partial<STTOptions>);
+    get label(): string;
+    get model(): string;
+    get provider(): string;
+    protected _recognize(): Promise<stt.SpeechEvent>;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
+}
+export declare class SpeechStream extends stt.SpeechStream {
+    #private;
+    constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions);
+    get label(): string;
+    protected run(): Promise<void>;
+    close(): void;
+}
+//# sourceMappingURL=stt.d.ts.map

package/dist/stt.d.ts

@@ -0,0 +1,53 @@
+import { type APIConnectOptions, stt } from '@livekit/agents';
+import { type STTModel } from './models.js';
+export type STTOptions = {
+    apiKey: string;
+    model: STTModel | (string & {});
+    sampleRate: number;
+    baseUrl: string;
+    audioChunkDurationMS: number;
+    language: string;
+};
+/**
+ * Cartesia speech to text.
+ *
+ * Supports:
+ *  - Streaming
+ *  - Turn detection
+ *  - Interim results
+ *
+ * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest
+ *
+ * @example
+ * ```typescript
+ * import { voice } from '@livekit/agents';
+ * import { STT, TTS } from '@livekit/agents-plugin-cartesia';
+ *
+ * const session = new voice.AgentSession({
+ *   stt: new STT(),
+ *   llm: new LLM(), // choose your favorite LLM
+ *   tts: new TTS(),
+ *   turnHandling: { turnDetection: 'stt' },
+ * });
+ * ```
+ */
+export declare class STT extends stt.STT {
+    #private;
+    constructor(opts?: Partial<STTOptions>);
+    get label(): string;
+    get model(): string;
+    get provider(): string;
+    protected _recognize(): Promise<stt.SpeechEvent>;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
+}
+export declare class SpeechStream extends stt.SpeechStream {
+    #private;
+    constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions);
+    get label(): string;
+    protected run(): Promise<void>;
+    close(): void;
+}
+//# sourceMappingURL=stt.d.ts.map

package/dist/stt.d.ts.map

@@ -0,0 +1 @@
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