@livekit/agents-plugin-cartesia 1.4.8 → 1.4.11

package/dist/index.cjs

@@ -22,7 +22,7 @@ class CartesiaPlugin extends import_agents.Plugin {
   constructor() {
     super({
       title: "cartesia",
-      version: "1.4.8",
+      version: "1.4.11",
       package: "@livekit/agents-plugin-cartesia"
     });
   }

package/dist/index.js

@@ -5,7 +5,7 @@ class CartesiaPlugin extends Plugin {
   constructor() {
     super({
       title: "cartesia",
-      version: "1.4.8",
+      version: "1.4.11",
       package: "@livekit/agents-plugin-cartesia"
     });
   }

package/dist/models.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\n/**\n * LiveKit uses this encoding for all audio\n */\nexport const AUDIO_ENCODING = 'pcm_s16le';\n\n// ============================================================================\n//                                   TTS\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSModels =\n  | 'sonic'\n  | 'sonic-2'\n  | 'sonic-3'\n  | 'sonic-lite'\n  | 'sonic-preview'\n  | 'sonic-turbo';\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSLanguages = 'en' | 'es' | 'fr' | 'de' | 'pt' | 'zh' | 'ja';\n\nexport const TTSDefaultVoiceId = 'f786b574-daa5-4673-aa0c-cbe3e8534c02';\n\nexport const isSonic3 = (model: string): boolean => model.startsWith('sonic-3');\n\nexport type TTSVoiceSpeed = 'fastest' | 'fast' | 'normal' | 'slow' | 'slowest';\n\nexport type TTSVoiceEmotion =\n  | 'anger:lowest'\n  | 'anger:low'\n  | 'anger'\n  | 'anger:high'\n  | 'anger:highest'\n  | 'positivity:lowest'\n  | 'positivity:low'\n  | 'positivity'\n  | 'positivity:high'\n  | 'positivity:highest'\n  | 'surprise:lowest'\n  | 'surprise:low'\n  | 'surprise'\n  | 'surprise:high'\n  | 'surprise:highest'\n  | 'sadness:lowest'\n  | 'sadness:low'\n  | 'sadness'\n  | 'sadness:high'\n  | 'sadness:highest'\n  | 'curiosity:lowest'\n  | 'curiosity:low'\n  | 'curiosity'\n  | 'curiosity:high'\n  | 'curiosity:highest';\n\n/**\n * @deprecated Encoding should not be parameterized. Only `pcm_s16le`is allowed. Prefer using {@link AUDIO_ENCODING}.\n */\nexport type TTSEncoding = 'pcm_s16le';\n\n// ============================================================================\n//                                   STT\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.\n */\nexport type STTModel = 'ink-2';\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAOO,MAAM,iBAAiB;AAsBvB,MAAM,oBAAoB;AAE1B,MAAM,WAAW,CAAC,UAA2B,MAAM,WAAW,SAAS;","names":[]}
+{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\n/**\n * LiveKit uses this encoding for all audio\n */\nexport const AUDIO_ENCODING = 'pcm_s16le';\n\n// ============================================================================\n//                                   TTS\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSModels =\n  | 'sonic'\n  | 'sonic-2'\n  | 'sonic-3'\n  | 'sonic-lite'\n  | 'sonic-preview'\n  | 'sonic-turbo';\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSLanguages = 'en' | 'es' | 'fr' | 'de' | 'pt' | 'zh' | 'ja';\n\nexport const TTSDefaultVoiceId = 'f786b574-daa5-4673-aa0c-cbe3e8534c02';\n\nexport const isSonic3 = (model: string): boolean => model.startsWith('sonic-3');\n\nexport type TTSVoiceSpeed = 'fastest' | 'fast' | 'normal' | 'slow' | 'slowest';\n\nexport type TTSVoiceEmotion =\n  | 'anger:lowest'\n  | 'anger:low'\n  | 'anger'\n  | 'anger:high'\n  | 'anger:highest'\n  | 'positivity:lowest'\n  | 'positivity:low'\n  | 'positivity'\n  | 'positivity:high'\n  | 'positivity:highest'\n  | 'surprise:lowest'\n  | 'surprise:low'\n  | 'surprise'\n  | 'surprise:high'\n  | 'surprise:highest'\n  | 'sadness:lowest'\n  | 'sadness:low'\n  | 'sadness'\n  | 'sadness:high'\n  | 'sadness:highest'\n  | 'curiosity:lowest'\n  | 'curiosity:low'\n  | 'curiosity'\n  | 'curiosity:high'\n  | 'curiosity:highest';\n\n/**\n * @deprecated Encoding should not be parameterized. Only `pcm_s16le`is allowed. Prefer using {@link AUDIO_ENCODING}.\n */\nexport type TTSEncoding = 'pcm_s16le';\n\n// ============================================================================\n//                                   STT\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.\n *\n * `ink-2` currently supports English only; `ink-whisper` is multilingual.\n */\nexport type STTModel = 'ink-2' | 'ink-whisper';\n"],"mappings":";;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAAA;AAOO,MAAM,iBAAiB;AAsBvB,MAAM,oBAAoB;AAE1B,MAAM,WAAW,CAAC,UAA2B,MAAM,WAAW,SAAS;","names":[]}

package/dist/models.d.cts

@@ -20,6 +20,8 @@ export type TTSVoiceEmotion = 'anger:lowest' | 'anger:low' | 'anger' | 'anger:hi
 export type TTSEncoding = 'pcm_s16le';
 /**
  * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.
+ *
+ * `ink-2` currently supports English only; `ink-whisper` is multilingual.
  */
-export type STTModel = 'ink-2';
+export type STTModel = 'ink-2' | 'ink-whisper';
 //# sourceMappingURL=models.d.ts.map

package/dist/models.d.ts

@@ -20,6 +20,8 @@ export type TTSVoiceEmotion = 'anger:lowest' | 'anger:low' | 'anger' | 'anger:hi
 export type TTSEncoding = 'pcm_s16le';
 /**
  * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.
+ *
+ * `ink-2` currently supports English only; `ink-whisper` is multilingual.
  */
-export type STTModel = 'ink-2';
+export type STTModel = 'ink-2' | 'ink-whisper';
 //# sourceMappingURL=models.d.ts.map

package/dist/models.d.ts.map

@@ -1 +1 @@
-{"version":3,"file":"models.d.ts","sourceRoot":"","sources":["../src/models.ts"],"names":[],"mappings":"AAIA;;GAEG;AACH,eAAO,MAAM,cAAc,cAAc,CAAC;AAM1C;;GAEG;AACH,MAAM,MAAM,SAAS,GACjB,OAAO,GACP,SAAS,GACT,SAAS,GACT,YAAY,GACZ,eAAe,GACf,aAAa,CAAC;AAElB;;GAEG;AACH,MAAM,MAAM,YAAY,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,CAAC;AAE1E,eAAO,MAAM,iBAAiB,yCAAyC,CAAC;AAExE,eAAO,MAAM,QAAQ,GAAI,OAAO,MAAM,KAAG,OAAsC,CAAC;AAEhF,MAAM,MAAM,aAAa,GAAG,SAAS,GAAG,MAAM,GAAG,QAAQ,GAAG,MAAM,GAAG,SAAS,CAAC;AAE/E,MAAM,MAAM,eAAe,GACvB,cAAc,GACd,WAAW,GACX,OAAO,GACP,YAAY,GACZ,eAAe,GACf,mBAAmB,GACnB,gBAAgB,GAChB,YAAY,GACZ,iBAAiB,GACjB,oBAAoB,GACpB,iBAAiB,GACjB,cAAc,GACd,UAAU,GACV,eAAe,GACf,kBAAkB,GAClB,gBAAgB,GAChB,aAAa,GACb,SAAS,GACT,cAAc,GACd,iBAAiB,GACjB,kBAAkB,GAClB,eAAe,GACf,WAAW,GACX,gBAAgB,GAChB,mBAAmB,CAAC;AAExB;;GAEG;AACH,MAAM,MAAM,WAAW,GAAG,WAAW,CAAC;AAMtC;;GAEG;AACH,MAAM,MAAM,QAAQ,GAAG,OAAO,CAAC"}
+{"version":3,"file":"models.d.ts","sourceRoot":"","sources":["../src/models.ts"],"names":[],"mappings":"AAIA;;GAEG;AACH,eAAO,MAAM,cAAc,cAAc,CAAC;AAM1C;;GAEG;AACH,MAAM,MAAM,SAAS,GACjB,OAAO,GACP,SAAS,GACT,SAAS,GACT,YAAY,GACZ,eAAe,GACf,aAAa,CAAC;AAElB;;GAEG;AACH,MAAM,MAAM,YAAY,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,GAAG,IAAI,CAAC;AAE1E,eAAO,MAAM,iBAAiB,yCAAyC,CAAC;AAExE,eAAO,MAAM,QAAQ,GAAI,OAAO,MAAM,KAAG,OAAsC,CAAC;AAEhF,MAAM,MAAM,aAAa,GAAG,SAAS,GAAG,MAAM,GAAG,QAAQ,GAAG,MAAM,GAAG,SAAS,CAAC;AAE/E,MAAM,MAAM,eAAe,GACvB,cAAc,GACd,WAAW,GACX,OAAO,GACP,YAAY,GACZ,eAAe,GACf,mBAAmB,GACnB,gBAAgB,GAChB,YAAY,GACZ,iBAAiB,GACjB,oBAAoB,GACpB,iBAAiB,GACjB,cAAc,GACd,UAAU,GACV,eAAe,GACf,kBAAkB,GAClB,gBAAgB,GAChB,aAAa,GACb,SAAS,GACT,cAAc,GACd,iBAAiB,GACjB,kBAAkB,GAClB,eAAe,GACf,WAAW,GACX,gBAAgB,GAChB,mBAAmB,CAAC;AAExB;;GAEG;AACH,MAAM,MAAM,WAAW,GAAG,WAAW,CAAC;AAMtC;;;;GAIG;AACH,MAAM,MAAM,QAAQ,GAAG,OAAO,GAAG,aAAa,CAAC"}

package/dist/models.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\n/**\n * LiveKit uses this encoding for all audio\n */\nexport const AUDIO_ENCODING = 'pcm_s16le';\n\n// ============================================================================\n//                                   TTS\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSModels =\n  | 'sonic'\n  | 'sonic-2'\n  | 'sonic-3'\n  | 'sonic-lite'\n  | 'sonic-preview'\n  | 'sonic-turbo';\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSLanguages = 'en' | 'es' | 'fr' | 'de' | 'pt' | 'zh' | 'ja';\n\nexport const TTSDefaultVoiceId = 'f786b574-daa5-4673-aa0c-cbe3e8534c02';\n\nexport const isSonic3 = (model: string): boolean => model.startsWith('sonic-3');\n\nexport type TTSVoiceSpeed = 'fastest' | 'fast' | 'normal' | 'slow' | 'slowest';\n\nexport type TTSVoiceEmotion =\n  | 'anger:lowest'\n  | 'anger:low'\n  | 'anger'\n  | 'anger:high'\n  | 'anger:highest'\n  | 'positivity:lowest'\n  | 'positivity:low'\n  | 'positivity'\n  | 'positivity:high'\n  | 'positivity:highest'\n  | 'surprise:lowest'\n  | 'surprise:low'\n  | 'surprise'\n  | 'surprise:high'\n  | 'surprise:highest'\n  | 'sadness:lowest'\n  | 'sadness:low'\n  | 'sadness'\n  | 'sadness:high'\n  | 'sadness:highest'\n  | 'curiosity:lowest'\n  | 'curiosity:low'\n  | 'curiosity'\n  | 'curiosity:high'\n  | 'curiosity:highest';\n\n/**\n * @deprecated Encoding should not be parameterized. Only `pcm_s16le`is allowed. Prefer using {@link AUDIO_ENCODING}.\n */\nexport type TTSEncoding = 'pcm_s16le';\n\n// ============================================================================\n//                                   STT\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.\n */\nexport type STTModel = 'ink-2';\n"],"mappings":"AAOO,MAAM,iBAAiB;AAsBvB,MAAM,oBAAoB;AAE1B,MAAM,WAAW,CAAC,UAA2B,MAAM,WAAW,SAAS;","names":[]}
+{"version":3,"sources":["../src/models.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\n\n/**\n * LiveKit uses this encoding for all audio\n */\nexport const AUDIO_ENCODING = 'pcm_s16le';\n\n// ============================================================================\n//                                   TTS\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSModels =\n  | 'sonic'\n  | 'sonic-2'\n  | 'sonic-3'\n  | 'sonic-lite'\n  | 'sonic-preview'\n  | 'sonic-turbo';\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/tts-models/latest) for all options.\n */\nexport type TTSLanguages = 'en' | 'es' | 'fr' | 'de' | 'pt' | 'zh' | 'ja';\n\nexport const TTSDefaultVoiceId = 'f786b574-daa5-4673-aa0c-cbe3e8534c02';\n\nexport const isSonic3 = (model: string): boolean => model.startsWith('sonic-3');\n\nexport type TTSVoiceSpeed = 'fastest' | 'fast' | 'normal' | 'slow' | 'slowest';\n\nexport type TTSVoiceEmotion =\n  | 'anger:lowest'\n  | 'anger:low'\n  | 'anger'\n  | 'anger:high'\n  | 'anger:highest'\n  | 'positivity:lowest'\n  | 'positivity:low'\n  | 'positivity'\n  | 'positivity:high'\n  | 'positivity:highest'\n  | 'surprise:lowest'\n  | 'surprise:low'\n  | 'surprise'\n  | 'surprise:high'\n  | 'surprise:highest'\n  | 'sadness:lowest'\n  | 'sadness:low'\n  | 'sadness'\n  | 'sadness:high'\n  | 'sadness:highest'\n  | 'curiosity:lowest'\n  | 'curiosity:low'\n  | 'curiosity'\n  | 'curiosity:high'\n  | 'curiosity:highest';\n\n/**\n * @deprecated Encoding should not be parameterized. Only `pcm_s16le`is allowed. Prefer using {@link AUDIO_ENCODING}.\n */\nexport type TTSEncoding = 'pcm_s16le';\n\n// ============================================================================\n//                                   STT\n// ============================================================================\n\n/**\n * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.\n *\n * `ink-2` currently supports English only; `ink-whisper` is multilingual.\n */\nexport type STTModel = 'ink-2' | 'ink-whisper';\n"],"mappings":"AAOO,MAAM,iBAAiB;AAsBvB,MAAM,oBAAoB;AAE1B,MAAM,WAAW,CAAC,UAA2B,MAAM,WAAW,SAAS;","names":[]}

package/dist/stt.cjs

@@ -37,7 +37,8 @@ const defaultSTTOptions = {
   sampleRate: 16e3,
   /** recommended default */
   audioChunkDurationMS: 160,
-  baseUrl: "https://api.cartesia.ai"
+  baseUrl: "https://api.cartesia.ai",
+  language: "en"
 };
 function mergeSTTOptions(base, override) {
   return {
@@ -45,9 +46,13 @@ function mergeSTTOptions(base, override) {
     baseUrl: override.baseUrl ?? base.baseUrl,
     model: override.model ?? base.model,
     sampleRate: override.sampleRate ?? base.sampleRate,
-    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS
+    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 = {}) {
@@ -64,6 +69,9 @@ class STT extends import_agents.stt.STT {
       );
     }
     this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);
+    if (opts.model === void 0) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
   }
   get label() {
     return "cartesia.STT";
@@ -80,6 +88,12 @@ class STT extends import_agents.stt.STT {
   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;
@@ -333,7 +347,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
       }
       case "turn.eager_end": {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
         if (!transcript) return;
         this.#currentTranscript = transcript;
         this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.PREFLIGHT_TRANSCRIPT, transcript);
@@ -345,7 +359,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
         return;
       case "turn.end": {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
         this.#sendTranscriptEvent(import_agents.stt.SpeechEventType.FINAL_TRANSCRIPT, transcript);
         this.#speaking = false;
         if (!this.queue.closed)
@@ -376,8 +390,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
       requestId: this.#requestId,
       alternatives: [
         {
-          // Cartesia STT only supports English at this time.
-          language: (0, import_agents.asLanguageCode)("en"),
+          language: (0, import_agents.asLanguageCode)(this.#opts.language),
           text: transcript,
           startTime: 0,
           endTime: 0,

package/dist/stt.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  AudioByteStream,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  log,\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};\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};\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  };\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\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\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;\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;\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          // Cartesia STT only supports English at this time.\n          language: asLanguageCode('en'),\n          text: transcript,\n          startTime: 0,\n          endTime: 0,\n          confidence: 0,\n        },\n      ],\n    });\n  }\n\n  #getCartesiaUrl(): string {\n    const params = new URLSearchParams({\n      model: this.#opts.model,\n      sample_rate: this.#opts.sampleRate.toString(),\n      encoding: AUDIO_ENCODING,\n    });\n\n    const wsBase = this.#opts.baseUrl.replace(/^http/, 'ws');\n    return `${wsBase}/stt/turns/websocket?${params.toString()}`;\n  }\n\n  override close() {\n    super.close();\n    this.#ws?.close();\n  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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. 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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. Language selection is\n    // expressed through the model (ink-2 for English, ink-whisper otherwise),\n    // so #opts.language is used only to tag emitted transcripts.\n    const params = new URLSearchParams({\n      model: this.#opts.model,\n      sample_rate: this.#opts.sampleRate.toString(),\n      encoding: AUDIO_ENCODING,\n    });\n\n    const wsBase = this.#opts.baseUrl.replace(/^http/, 'ws');\n    return `${wsBase}/stt/turns/websocket?${params.toString()}`;\n  }\n\n  override close() {\n    super.close();\n    this.#ws?.close();\n  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package/dist/stt.d.cts

@@ -6,6 +6,7 @@ export type STTOptions = {
     sampleRate: number;
     baseUrl: string;
     audioChunkDurationMS: number;
+    language: string;
 };
 /**
  * Cartesia speech to text.
@@ -40,6 +41,7 @@ export declare class STT extends stt.STT {
     stream(options?: {
         connOptions?: APIConnectOptions;
     }): SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
 }
 export declare class SpeechStream extends stt.SpeechStream {
     #private;

package/dist/stt.d.ts

@@ -6,6 +6,7 @@ export type STTOptions = {
     sampleRate: number;
     baseUrl: string;
     audioChunkDurationMS: number;
+    language: string;
 };
 /**
  * Cartesia speech to text.
@@ -40,6 +41,7 @@ export declare class STT extends stt.STT {
     stream(options?: {
         connOptions?: APIConnectOptions;
     }): SpeechStream;
+    updateOptions(opts: Partial<STTOptions>): void;
 }
 export declare class SpeechStream extends stt.SpeechStream {
     #private;

package/dist/stt.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"stt.d.ts","sourceRoot":"","sources":["../src/stt.ts"],"names":[],"mappings":"AAGA,OAAO,EACL,KAAK,iBAAiB,EAQtB,GAAG,EAEJ,MAAM,iBAAiB,CAAC;AAIzB,OAAO,EAAkB,KAAK,QAAQ,EAAE,MAAM,aAAa,CAAC;AAuH5D,MAAM,MAAM,UAAU,GAAG;IACvB,MAAM,EAAE,MAAM,CAAC;IAEf,KAAK,EAAE,QAAQ,GAAG,CAAC,MAAM,GAAG,EAAE,CAAC,CAAC;IAChC,UAAU,EAAE,MAAM,CAAC;IACnB,OAAO,EAAE,MAAM,CAAC;IAChB,oBAAoB,EAAE,MAAM,CAAC;IAC7B,QAAQ,EAAE,MAAM,CAAC;CAClB,CAAC;AAiCF;;;;;;;;;;;;;;;;;;;;;;GAsBG;AACH,qBAAa,GAAI,SAAQ,GAAG,CAAC,GAAG;;gBAGlB,IAAI,GAAE,OAAO,CAAC,UAAU,CAAM;IAwB1C,IAAa,KAAK,IAAI,MAAM,CAE3B;IAED,IAAa,KAAK,IAAI,MAAM,CAE3B;IAED,IAAa,QAAQ,IAAI,MAAM,CAE9B;cAEwB,UAAU,IAAI,OAAO,CAAC,GAAG,CAAC,WAAW,CAAC;IAItD,MAAM,CAAC,OAAO,CAAC,EAAE;QAAE,WAAW,CAAC,EAAE,iBAAiB,CAAA;KAAE,GAAG,YAAY;IAI5E,aAAa,CAAC,IAAI,EAAE,OAAO,CAAC,UAAU,CAAC;CAUxC;AAED,qBAAa,YAAa,SAAQ,GAAG,CAAC,YAAY;;gBAUpC,WAAW,EAAE,GAAG,EAAE,IAAI,EAAE,UAAU,EAAE,WAAW,CAAC,EAAE,iBAAiB;IAK/E,IAAa,KAAK,IAAI,MAAM,CAE3B;cAEwB,GAAG;IAiXnB,KAAK;CAIf"}

package/dist/stt.js

@@ -3,7 +3,9 @@ import {
   AudioByteStream,
   asLanguageCode,
   calculateAudioDurationSeconds,
+  getBaseLanguage,
   log,
+  normalizeLanguage,
   stt,
   waitForAbort
 } from "@livekit/agents";
@@ -21,7 +23,8 @@ const defaultSTTOptions = {
   sampleRate: 16e3,
   /** recommended default */
   audioChunkDurationMS: 160,
-  baseUrl: "https://api.cartesia.ai"
+  baseUrl: "https://api.cartesia.ai",
+  language: "en"
 };
 function mergeSTTOptions(base, override) {
   return {
@@ -29,9 +32,13 @@ function mergeSTTOptions(base, override) {
     baseUrl: override.baseUrl ?? base.baseUrl,
     model: override.model ?? base.model,
     sampleRate: override.sampleRate ?? base.sampleRate,
-    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS
+    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,
+    language: override.language !== void 0 ? normalizeLanguage(override.language) : base.language
   };
 }
+function resolveSTTModel(language) {
+  return getBaseLanguage(language) === "en" ? "ink-2" : "ink-whisper";
+}
 class STT extends stt.STT {
   #opts;
   constructor(opts = {}) {
@@ -48,6 +55,9 @@ class STT extends stt.STT {
       );
     }
     this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);
+    if (opts.model === void 0) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
   }
   get label() {
     return "cartesia.STT";
@@ -64,6 +74,12 @@ class STT extends stt.STT {
   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 stt.SpeechStream {
   #opts;
@@ -317,7 +333,7 @@ class SpeechStream extends stt.SpeechStream {
       }
       case "turn.eager_end": {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
         if (!transcript) return;
         this.#currentTranscript = transcript;
         this.#sendTranscriptEvent(stt.SpeechEventType.PREFLIGHT_TRANSCRIPT, transcript);
@@ -329,7 +345,7 @@ class SpeechStream extends stt.SpeechStream {
         return;
       case "turn.end": {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
         this.#sendTranscriptEvent(stt.SpeechEventType.FINAL_TRANSCRIPT, transcript);
         this.#speaking = false;
         if (!this.queue.closed)
@@ -360,8 +376,7 @@ class SpeechStream extends stt.SpeechStream {
       requestId: this.#requestId,
       alternatives: [
         {
-          // Cartesia STT only supports English at this time.
-          language: asLanguageCode("en"),
+          language: asLanguageCode(this.#opts.language),
           text: transcript,
           startTime: 0,
           endTime: 0,

package/dist/stt.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2026 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  AudioByteStream,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  log,\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};\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};\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  };\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\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\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;\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;\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          // Cartesia STT only supports English at this time.\n          language: asLanguageCode('en'),\n          text: transcript,\n          startTime: 0,\n          endTime: 0,\n          confidence: 0,\n        },\n      ],\n    });\n  }\n\n  #getCartesiaUrl(): string {\n    const params = new URLSearchParams({\n      model: this.#opts.model,\n      sample_rate: this.#opts.sampleRate.toString(),\n      encoding: AUDIO_ENCODING,\n    });\n\n    const wsBase = 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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. Language selection is\n    // expressed through the model (ink-2 for English, ink-whisper otherwise),\n    // so #opts.language is used only to tag emitted transcripts.\n    const params = new URLSearchParams({\n      model: this.#opts.model,\n      sample_rate: this.#opts.sampleRate.toString(),\n      encoding: AUDIO_ENCODING,\n    });\n\n    const wsBase = this.#opts.baseUrl.replace(/^http/, 'ws');\n    return `${wsBase}/stt/turns/websocket?${params.toString()}`;\n  }\n\n  override close() {\n    super.close();\n    this.#ws?.close();\n  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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@livekit/agents-plugin-cartesia",
-  "version": "1.4.8",
+  "version": "1.4.11",
   "description": "Cartesia plugin for LiveKit Node Agents",
   "main": "dist/index.js",
   "require": "dist/index.cjs",
@@ -30,10 +30,10 @@
     "@types/ws": "^8.5.10",
     "tsup": "^8.3.5",
     "typescript": "^5.0.0",
-    "@livekit/agents": "1.4.8",
-    "@livekit/agents-plugin-openai": "1.4.8",
-    "@livekit/agents-plugin-silero": "1.4.8",
-    "@livekit/agents-plugins-test": "1.4.8"
+    "@livekit/agents": "1.4.11",
+    "@livekit/agents-plugin-openai": "1.4.11",
+    "@livekit/agents-plugin-silero": "1.4.11",
+    "@livekit/agents-plugins-test": "1.4.11"
   },
   "dependencies": {
     "ws": "^8.18.0"
@@ -41,7 +41,7 @@
   "peerDependencies": {
     "@livekit/rtc-node": "^0.13.29",
     "zod": "^3.25.76 || ^4.1.8",
-    "@livekit/agents": "1.4.8"
+    "@livekit/agents": "1.4.11"
   },
   "scripts": {
     "build": "tsup --onSuccess \"pnpm build:types\"",

package/src/models.ts

@@ -71,5 +71,7 @@ export type TTSEncoding = 'pcm_s16le';
 
 /**
  * See [the docs](https://docs.cartesia.ai/build-with-cartesia/stt-models/latest) for all options.
+ *
+ * `ink-2` currently supports English only; `ink-whisper` is multilingual.
  */
-export type STTModel = 'ink-2';
+export type STTModel = 'ink-2' | 'ink-whisper';

package/src/stt.ts

@@ -7,7 +7,9 @@ import {
   AudioByteStream,
   asLanguageCode,
   calculateAudioDurationSeconds,
+  getBaseLanguage,
   log,
+  normalizeLanguage,
   stt,
   waitForAbort,
 } from '@livekit/agents';
@@ -140,6 +142,7 @@ export type STTOptions = {
   sampleRate: number;
   baseUrl: string;
   audioChunkDurationMS: number;
+  language: string;
 };
 
 const defaultSTTOptions = {
@@ -149,6 +152,7 @@ const defaultSTTOptions = {
   /** recommended default */
   audioChunkDurationMS: 160,
   baseUrl: 'https://api.cartesia.ai',
+  language: 'en',
 };
 
 function mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {
@@ -158,9 +162,20 @@ function mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOp
     model: override.model ?? base.model,
     sampleRate: override.sampleRate ?? base.sampleRate,
     audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,
+    language:
+      override.language !== undefined ? normalizeLanguage(override.language) : base.language,
   };
 }
 
+/**
+ * Resolve the STT model from the configured language when the caller did not
+ * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports
+ * English, so non-English languages route to the multilingual `ink-whisper`.
+ */
+function resolveSTTModel(language: string): STTModel {
+  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';
+}
+
 /**
  * Cartesia speech to text.
  *
@@ -203,6 +218,12 @@ export class STT extends stt.STT {
     }
 
     this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);
+
+    // Route to the multilingual model for non-English languages unless the
+    // caller pinned a model. Matches the Python plugin's language→model logic.
+    if (opts.model === undefined) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
   }
 
   override get label(): string {
@@ -224,6 +245,17 @@ export class STT extends stt.STT {
   override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {
     return new SpeechStream(this, this.#opts, options?.connOptions);
   }
+
+  updateOptions(opts: Partial<STTOptions>) {
+    this.#opts = mergeSTTOptions(this.#opts, opts);
+
+    // Keep the model in sync with a newly set language (e.g. switching to a
+    // non-English language must move off the English-only ink-2), unless the
+    // caller pinned a model in the same call. Mirrors the constructor.
+    if (opts.language !== undefined && opts.model === undefined) {
+      this.#opts.model = resolveSTTModel(this.#opts.language);
+    }
+  }
 }
 
 export class SpeechStream extends stt.SpeechStream {
@@ -536,7 +568,7 @@ export class SpeechStream extends stt.SpeechStream {
 
       case 'turn.eager_end': {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
         if (!transcript) return;
         this.#currentTranscript = transcript;
         this.#sendTranscriptEvent(stt.SpeechEventType.PREFLIGHT_TRANSCRIPT, transcript);
@@ -552,7 +584,7 @@ export class SpeechStream extends stt.SpeechStream {
 
       case 'turn.end': {
         if (!this.#speaking) return;
-        const transcript = data.transcript;
+        const transcript = data.transcript || this.#currentTranscript;
 
         this.#sendTranscriptEvent(stt.SpeechEventType.FINAL_TRANSCRIPT, transcript);
 
@@ -589,8 +621,7 @@ export class SpeechStream extends stt.SpeechStream {
       requestId: this.#requestId,
       alternatives: [
         {
-          // Cartesia STT only supports English at this time.
-          language: asLanguageCode('en'),
+          language: asLanguageCode(this.#opts.language),
           text: transcript,
           startTime: 0,
           endTime: 0,
@@ -601,6 +632,10 @@ export class SpeechStream extends stt.SpeechStream {
   }
 
   #getCartesiaUrl(): string {
+    // The Cartesia /stt/turns/websocket endpoint only accepts model, sample_rate
+    // and encoding — there is no `language` query param. Language selection is
+    // expressed through the model (ink-2 for English, ink-whisper otherwise),
+    // so #opts.language is used only to tag emitted transcripts.
     const params = new URLSearchParams({
       model: this.#opts.model,
       sample_rate: this.#opts.sampleRate.toString(),