@livekit/agents-plugin-cartesia 1.6.4 → 1.7.0

package/dist/index.cjs

@@ -22,7 +22,7 @@ class CartesiaPlugin extends import_agents.Plugin {
   constructor() {
     super({
       title: "cartesia",
-      version: "1.6.4",
+      version: "1.7.0",
       package: "@livekit/agents-plugin-cartesia"
     });
   }

package/dist/index.js

@@ -5,7 +5,7 @@ class CartesiaPlugin extends Plugin {
   constructor() {
     super({
       title: "cartesia",
-      version: "1.6.4",
+      version: "1.7.0",
       package: "@livekit/agents-plugin-cartesia"
     });
   }

package/dist/stt.cjs

@@ -291,7 +291,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
         }
         try {
           if (msg.type === "error") {
-            this.#logger.error("Cartesia sent an error", msg);
+            this.#logger.error({ "lk.pii.message": msg }, "Cartesia sent an error");
             if (msg.status_code === void 0 || msg.status_code >= 500) {
               pendingError = new import_agents.APIConnectionError({
                 message: msg.message || msg.title,
@@ -422,7 +422,7 @@ class SpeechStream extends import_agents.stt.SpeechStream {
         return;
       }
       default:
-        this.#logger.warn("received unexpected message from Cartesia STT", { data });
+        this.#logger.warn({ "lk.pii.data": data }, "received unexpected message from Cartesia STT");
     }
   }
   #sendTranscriptEvent(eventType, transcript) {

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  APIStatusError,\n  AudioByteStream,\n  DEFAULT_API_CONNECT_OPTIONS,\n  asError,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { IncomingMessage } from 'node:http';\nimport { WebSocket } from 'ws';\nimport { AUDIO_ENCODING, type STTModel } from './models.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst REQUEST_ID_HEADER = 'cartesia-request-id';\nconst API_VERSION = '2026-03-01';\nconst DRAIN_TIMEOUT_MS = 5000;\nconst KEEPALIVE_INTERVAL_MS = 30000;\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\n/**\n * Fires once when the WebSocket connection is established.\n *\n * You do not need to wait for this event before sending audio.\n */\ntype STTConnectedEvent = {\n  type: 'connected';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Model predicts {@link stt.SpeechEventType.START_OF_SPEECH}.\n */\ntype STTTurnStartEvent = {\n  type: 'turn.start';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires repeatedly as the model transcribes the current user turn.\n *\n * Can be used for {@link stt.SpeechEventType.INTERIM_TRANSCRIPT}.\n */\ntype STTTurnUpdateEvent = {\n  type: 'turn.update';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires when the model predicts that the user might be done speaking.\n *\n * Can be used for {@link stt.SpeechEventType.PREFLIGHT_TRANSCRIPT}.\n */\ntype STTTurnEagerEndEvent = {\n  type: 'turn.eager_end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires after {@link STTTurnEagerEndEvent} if the user turn has not actually ended.\n */\ntype STTTurnResumeEvent = {\n  type: 'turn.resume';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Marks the end of a user turn.\n *\n * This is used for {@link stt.SpeechEventType.END_OF_SPEECH} and {@link stt.SpeechEventType.FINAL_TRANSCRIPT}.\n */\ntype STTTurnEndEvent = {\n  type: 'turn.end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\ntype STTErrorEvent = {\n  type: 'error';\n  error_code?: string;\n  status_code?: number;\n  title?: string;\n  message?: string;\n  doc_url?: string;\n  request_id?: string;\n};\n\n/**\n * Server-sent event on the `/stt/turns/websocket` endpoint.\n *\n * See https://docs.cartesia.ai/api-reference/stt/turns/websocket.\n */\ntype STTEventMessage =\n  | STTConnectedEvent\n  | STTTurnStartEvent\n  | STTTurnUpdateEvent\n  | STTTurnEagerEndEvent\n  | STTTurnResumeEvent\n  | STTTurnEndEvent\n  | STTErrorEvent;\n\nexport type STTOptions = {\n  apiKey: string;\n  // eslint-disable-next-line @typescript-eslint/ban-types\n  model: STTModel | (string & {});\n  sampleRate: number;\n  baseUrl: string;\n  audioChunkDurationMS: number;\n  language: string;\n};\n\nconst defaultSTTOptions = {\n  model: 'ink-2' satisfies STTModel,\n  /** recommended default */\n  sampleRate: 16_000,\n  /** recommended default */\n  audioChunkDurationMS: 160,\n  baseUrl: 'https://api.cartesia.ai',\n  language: 'en',\n};\n\nfunction mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {\n  return {\n    apiKey: override.apiKey ?? base.apiKey,\n    baseUrl: override.baseUrl ?? base.baseUrl,\n    model: override.model ?? base.model,\n    sampleRate: override.sampleRate ?? base.sampleRate,\n    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,\n    language:\n      override.language !== undefined ? normalizeLanguage(override.language) : base.language,\n  };\n}\n\n/**\n * Resolve the STT model from the configured language when the caller did not\n * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports\n * English, so non-English languages route to the multilingual `ink-whisper`.\n */\nfunction resolveSTTModel(language: string): STTModel {\n  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';\n}\n\n/**\n * Cartesia speech to text.\n *\n * Supports:\n *  - Streaming\n *  - Turn detection\n *  - Interim results\n *\n * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest\n *\n * @example\n * ```typescript\n * import { voice } from '@livekit/agents';\n * import { STT, TTS } from '@livekit/agents-plugin-cartesia';\n *\n * const session = new voice.AgentSession({\n *   stt: new STT(),\n *   llm: new LLM(), // choose your favorite LLM\n *   tts: new TTS(),\n *   turnHandling: { turnDetection: 'stt' },\n * });\n * ```\n */\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: false,\n      diarization: false,\n    });\n\n    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;\n    if (!apiKey) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);\n\n    // Route to the multilingual model for non-English languages unless the\n    // caller pinned a model. Matches the Python plugin's language→model logic.\n    if (opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n\n  override get label(): string {\n    return 'cartesia.STT';\n  }\n\n  override get model(): string {\n    return this.#opts.model;\n  }\n\n  override get provider(): string {\n    return 'Cartesia';\n  }\n\n  protected override async _recognize(): Promise<stt.SpeechEvent> {\n    throw new Error('Cartesia STT does not support batch recognition, use stream() instead');\n  }\n\n  override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = mergeSTTOptions(this.#opts, opts);\n\n    // Keep the model in sync with a newly set language (e.g. switching to a\n    // non-English language must move off the English-only ink-2), unless the\n    // caller pinned a model in the same call. Mirrors the constructor.\n    if (opts.language !== undefined && opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #ws: WebSocket | null = null;\n  #requestId: string | undefined;\n  #speaking = false;\n  #currentTranscript = '';\n  #speechDuration = 0;\n  #closingWs = false;\n  #connectTimeout: number;\n\n  constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(sttInstance, opts.sampleRate, connOptions);\n    this.#opts = { ...opts };\n    this.#connectTimeout = connOptions?.timeoutMs ?? DEFAULT_API_CONNECT_OPTIONS.timeoutMs;\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      let ws: WebSocket;\n      try {\n        ws = new WebSocket(url, {\n          handshakeTimeout: this.#connectTimeout,\n          headers: {\n            [AUTHORIZATION_HEADER]: this.#opts.apiKey,\n            [VERSION_HEADER]: API_VERSION,\n          },\n        });\n      } catch (error) {\n        throw new APIConnectionError({\n          message: sanitizedErrorName(asError(error)),\n          options: { retryable: true },\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 cleanup = () => {\n          ws.off('open', onOpen);\n          ws.off('unexpected-response', onUnexpectedResponse);\n          ws.off('error', onError);\n        };\n        const onOpen = () => {\n          cleanup();\n          resolve();\n        };\n        const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n          cleanup();\n          ws.on('error', () => {});\n          ws.close();\n          // Authentication headers can appear in WebSocket handshake errors.\n          const statusCode = response.statusCode ?? -1;\n          reject(\n            new APIStatusError({\n              message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n              options: { statusCode },\n            }),\n          );\n        };\n        const onError = (err: Error) => {\n          cleanup();\n          // Transport errors can contain credentials in URLs.\n          reject(\n            new APIConnectionError({\n              message: sanitizedErrorName(err),\n              options: { retryable: true },\n            }),\n          );\n        };\n\n        ws.once('open', onOpen);\n        ws.once('unexpected-response', onUnexpectedResponse);\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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+{"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  APIStatusError,\n  AudioByteStream,\n  DEFAULT_API_CONNECT_OPTIONS,\n  asError,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { IncomingMessage } from 'node:http';\nimport { WebSocket } from 'ws';\nimport { AUDIO_ENCODING, type STTModel } from './models.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst REQUEST_ID_HEADER = 'cartesia-request-id';\nconst API_VERSION = '2026-03-01';\nconst DRAIN_TIMEOUT_MS = 5000;\nconst KEEPALIVE_INTERVAL_MS = 30000;\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\n/**\n * Fires once when the WebSocket connection is established.\n *\n * You do not need to wait for this event before sending audio.\n */\ntype STTConnectedEvent = {\n  type: 'connected';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Model predicts {@link stt.SpeechEventType.START_OF_SPEECH}.\n */\ntype STTTurnStartEvent = {\n  type: 'turn.start';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires repeatedly as the model transcribes the current user turn.\n *\n * Can be used for {@link stt.SpeechEventType.INTERIM_TRANSCRIPT}.\n */\ntype STTTurnUpdateEvent = {\n  type: 'turn.update';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires when the model predicts that the user might be done speaking.\n *\n * Can be used for {@link stt.SpeechEventType.PREFLIGHT_TRANSCRIPT}.\n */\ntype STTTurnEagerEndEvent = {\n  type: 'turn.eager_end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires after {@link STTTurnEagerEndEvent} if the user turn has not actually ended.\n */\ntype STTTurnResumeEvent = {\n  type: 'turn.resume';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Marks the end of a user turn.\n *\n * This is used for {@link stt.SpeechEventType.END_OF_SPEECH} and {@link stt.SpeechEventType.FINAL_TRANSCRIPT}.\n */\ntype STTTurnEndEvent = {\n  type: 'turn.end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\ntype STTErrorEvent = {\n  type: 'error';\n  error_code?: string;\n  status_code?: number;\n  title?: string;\n  message?: string;\n  doc_url?: string;\n  request_id?: string;\n};\n\n/**\n * Server-sent event on the `/stt/turns/websocket` endpoint.\n *\n * See https://docs.cartesia.ai/api-reference/stt/turns/websocket.\n */\ntype STTEventMessage =\n  | STTConnectedEvent\n  | STTTurnStartEvent\n  | STTTurnUpdateEvent\n  | STTTurnEagerEndEvent\n  | STTTurnResumeEvent\n  | STTTurnEndEvent\n  | STTErrorEvent;\n\nexport type STTOptions = {\n  apiKey: string;\n  // eslint-disable-next-line @typescript-eslint/ban-types\n  model: STTModel | (string & {});\n  sampleRate: number;\n  baseUrl: string;\n  audioChunkDurationMS: number;\n  language: string;\n};\n\nconst defaultSTTOptions = {\n  model: 'ink-2' satisfies STTModel,\n  /** recommended default */\n  sampleRate: 16_000,\n  /** recommended default */\n  audioChunkDurationMS: 160,\n  baseUrl: 'https://api.cartesia.ai',\n  language: 'en',\n};\n\nfunction mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {\n  return {\n    apiKey: override.apiKey ?? base.apiKey,\n    baseUrl: override.baseUrl ?? base.baseUrl,\n    model: override.model ?? base.model,\n    sampleRate: override.sampleRate ?? base.sampleRate,\n    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,\n    language:\n      override.language !== undefined ? normalizeLanguage(override.language) : base.language,\n  };\n}\n\n/**\n * Resolve the STT model from the configured language when the caller did not\n * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports\n * English, so non-English languages route to the multilingual `ink-whisper`.\n */\nfunction resolveSTTModel(language: string): STTModel {\n  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';\n}\n\n/**\n * Cartesia speech to text.\n *\n * Supports:\n *  - Streaming\n *  - Turn detection\n *  - Interim results\n *\n * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest\n *\n * @example\n * ```typescript\n * import { voice } from '@livekit/agents';\n * import { STT, TTS } from '@livekit/agents-plugin-cartesia';\n *\n * const session = new voice.AgentSession({\n *   stt: new STT(),\n *   llm: new LLM(), // choose your favorite LLM\n *   tts: new TTS(),\n *   turnHandling: { turnDetection: 'stt' },\n * });\n * ```\n */\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: false,\n      diarization: false,\n    });\n\n    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;\n    if (!apiKey) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);\n\n    // Route to the multilingual model for non-English languages unless the\n    // caller pinned a model. Matches the Python plugin's language→model logic.\n    if (opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n\n  override get label(): string {\n    return 'cartesia.STT';\n  }\n\n  override get model(): string {\n    return this.#opts.model;\n  }\n\n  override get provider(): string {\n    return 'Cartesia';\n  }\n\n  protected override async _recognize(): Promise<stt.SpeechEvent> {\n    throw new Error('Cartesia STT does not support batch recognition, use stream() instead');\n  }\n\n  override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = mergeSTTOptions(this.#opts, opts);\n\n    // Keep the model in sync with a newly set language (e.g. switching to a\n    // non-English language must move off the English-only ink-2), unless the\n    // caller pinned a model in the same call. Mirrors the constructor.\n    if (opts.language !== undefined && opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #ws: WebSocket | null = null;\n  #requestId: string | undefined;\n  #speaking = false;\n  #currentTranscript = '';\n  #speechDuration = 0;\n  #closingWs = false;\n  #connectTimeout: number;\n\n  constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(sttInstance, opts.sampleRate, connOptions);\n    this.#opts = { ...opts };\n    this.#connectTimeout = connOptions?.timeoutMs ?? DEFAULT_API_CONNECT_OPTIONS.timeoutMs;\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      let ws: WebSocket;\n      try {\n        ws = new WebSocket(url, {\n          handshakeTimeout: this.#connectTimeout,\n          headers: {\n            [AUTHORIZATION_HEADER]: this.#opts.apiKey,\n            [VERSION_HEADER]: API_VERSION,\n          },\n        });\n      } catch (error) {\n        throw new APIConnectionError({\n          message: sanitizedErrorName(asError(error)),\n          options: { retryable: true },\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 cleanup = () => {\n          ws.off('open', onOpen);\n          ws.off('unexpected-response', onUnexpectedResponse);\n          ws.off('error', onError);\n        };\n        const onOpen = () => {\n          cleanup();\n          resolve();\n        };\n        const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n          cleanup();\n          ws.on('error', () => {});\n          ws.close();\n          // Authentication headers can appear in WebSocket handshake errors.\n          const statusCode = response.statusCode ?? -1;\n          reject(\n            new APIStatusError({\n              message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n              options: { statusCode },\n            }),\n          );\n        };\n        const onError = (err: Error) => {\n          cleanup();\n          // Transport errors can contain credentials in URLs.\n          reject(\n            new APIConnectionError({\n              message: sanitizedErrorName(err),\n              options: { retryable: true },\n            }),\n          );\n        };\n\n        ws.once('open', onOpen);\n        ws.once('unexpected-response', onUnexpectedResponse);\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({ 'lk.pii.message': msg }, 'Cartesia sent an error');\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({ 'lk.pii.data': data }, 'received unexpected message from Cartesia STT');\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.js

@@ -280,7 +280,7 @@ class SpeechStream extends stt.SpeechStream {
         }
         try {
           if (msg.type === "error") {
-            this.#logger.error("Cartesia sent an error", msg);
+            this.#logger.error({ "lk.pii.message": msg }, "Cartesia sent an error");
             if (msg.status_code === void 0 || msg.status_code >= 500) {
               pendingError = new APIConnectionError({
                 message: msg.message || msg.title,
@@ -411,7 +411,7 @@ class SpeechStream extends stt.SpeechStream {
         return;
       }
       default:
-        this.#logger.warn("received unexpected message from Cartesia STT", { data });
+        this.#logger.warn({ "lk.pii.data": data }, "received unexpected message from Cartesia STT");
     }
   }
   #sendTranscriptEvent(eventType, transcript) {

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  APIStatusError,\n  AudioByteStream,\n  DEFAULT_API_CONNECT_OPTIONS,\n  asError,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { IncomingMessage } from 'node:http';\nimport { WebSocket } from 'ws';\nimport { AUDIO_ENCODING, type STTModel } from './models.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst REQUEST_ID_HEADER = 'cartesia-request-id';\nconst API_VERSION = '2026-03-01';\nconst DRAIN_TIMEOUT_MS = 5000;\nconst KEEPALIVE_INTERVAL_MS = 30000;\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\n/**\n * Fires once when the WebSocket connection is established.\n *\n * You do not need to wait for this event before sending audio.\n */\ntype STTConnectedEvent = {\n  type: 'connected';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Model predicts {@link stt.SpeechEventType.START_OF_SPEECH}.\n */\ntype STTTurnStartEvent = {\n  type: 'turn.start';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires repeatedly as the model transcribes the current user turn.\n *\n * Can be used for {@link stt.SpeechEventType.INTERIM_TRANSCRIPT}.\n */\ntype STTTurnUpdateEvent = {\n  type: 'turn.update';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires when the model predicts that the user might be done speaking.\n *\n * Can be used for {@link stt.SpeechEventType.PREFLIGHT_TRANSCRIPT}.\n */\ntype STTTurnEagerEndEvent = {\n  type: 'turn.eager_end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires after {@link STTTurnEagerEndEvent} if the user turn has not actually ended.\n */\ntype STTTurnResumeEvent = {\n  type: 'turn.resume';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Marks the end of a user turn.\n *\n * This is used for {@link stt.SpeechEventType.END_OF_SPEECH} and {@link stt.SpeechEventType.FINAL_TRANSCRIPT}.\n */\ntype STTTurnEndEvent = {\n  type: 'turn.end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\ntype STTErrorEvent = {\n  type: 'error';\n  error_code?: string;\n  status_code?: number;\n  title?: string;\n  message?: string;\n  doc_url?: string;\n  request_id?: string;\n};\n\n/**\n * Server-sent event on the `/stt/turns/websocket` endpoint.\n *\n * See https://docs.cartesia.ai/api-reference/stt/turns/websocket.\n */\ntype STTEventMessage =\n  | STTConnectedEvent\n  | STTTurnStartEvent\n  | STTTurnUpdateEvent\n  | STTTurnEagerEndEvent\n  | STTTurnResumeEvent\n  | STTTurnEndEvent\n  | STTErrorEvent;\n\nexport type STTOptions = {\n  apiKey: string;\n  // eslint-disable-next-line @typescript-eslint/ban-types\n  model: STTModel | (string & {});\n  sampleRate: number;\n  baseUrl: string;\n  audioChunkDurationMS: number;\n  language: string;\n};\n\nconst defaultSTTOptions = {\n  model: 'ink-2' satisfies STTModel,\n  /** recommended default */\n  sampleRate: 16_000,\n  /** recommended default */\n  audioChunkDurationMS: 160,\n  baseUrl: 'https://api.cartesia.ai',\n  language: 'en',\n};\n\nfunction mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {\n  return {\n    apiKey: override.apiKey ?? base.apiKey,\n    baseUrl: override.baseUrl ?? base.baseUrl,\n    model: override.model ?? base.model,\n    sampleRate: override.sampleRate ?? base.sampleRate,\n    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,\n    language:\n      override.language !== undefined ? normalizeLanguage(override.language) : base.language,\n  };\n}\n\n/**\n * Resolve the STT model from the configured language when the caller did not\n * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports\n * English, so non-English languages route to the multilingual `ink-whisper`.\n */\nfunction resolveSTTModel(language: string): STTModel {\n  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';\n}\n\n/**\n * Cartesia speech to text.\n *\n * Supports:\n *  - Streaming\n *  - Turn detection\n *  - Interim results\n *\n * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest\n *\n * @example\n * ```typescript\n * import { voice } from '@livekit/agents';\n * import { STT, TTS } from '@livekit/agents-plugin-cartesia';\n *\n * const session = new voice.AgentSession({\n *   stt: new STT(),\n *   llm: new LLM(), // choose your favorite LLM\n *   tts: new TTS(),\n *   turnHandling: { turnDetection: 'stt' },\n * });\n * ```\n */\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: false,\n      diarization: false,\n    });\n\n    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;\n    if (!apiKey) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);\n\n    // Route to the multilingual model for non-English languages unless the\n    // caller pinned a model. Matches the Python plugin's language→model logic.\n    if (opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n\n  override get label(): string {\n    return 'cartesia.STT';\n  }\n\n  override get model(): string {\n    return this.#opts.model;\n  }\n\n  override get provider(): string {\n    return 'Cartesia';\n  }\n\n  protected override async _recognize(): Promise<stt.SpeechEvent> {\n    throw new Error('Cartesia STT does not support batch recognition, use stream() instead');\n  }\n\n  override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = mergeSTTOptions(this.#opts, opts);\n\n    // Keep the model in sync with a newly set language (e.g. switching to a\n    // non-English language must move off the English-only ink-2), unless the\n    // caller pinned a model in the same call. Mirrors the constructor.\n    if (opts.language !== undefined && opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #ws: WebSocket | null = null;\n  #requestId: string | undefined;\n  #speaking = false;\n  #currentTranscript = '';\n  #speechDuration = 0;\n  #closingWs = false;\n  #connectTimeout: number;\n\n  constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(sttInstance, opts.sampleRate, connOptions);\n    this.#opts = { ...opts };\n    this.#connectTimeout = connOptions?.timeoutMs ?? DEFAULT_API_CONNECT_OPTIONS.timeoutMs;\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      let ws: WebSocket;\n      try {\n        ws = new WebSocket(url, {\n          handshakeTimeout: this.#connectTimeout,\n          headers: {\n            [AUTHORIZATION_HEADER]: this.#opts.apiKey,\n            [VERSION_HEADER]: API_VERSION,\n          },\n        });\n      } catch (error) {\n        throw new APIConnectionError({\n          message: sanitizedErrorName(asError(error)),\n          options: { retryable: true },\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 cleanup = () => {\n          ws.off('open', onOpen);\n          ws.off('unexpected-response', onUnexpectedResponse);\n          ws.off('error', onError);\n        };\n        const onOpen = () => {\n          cleanup();\n          resolve();\n        };\n        const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n          cleanup();\n          ws.on('error', () => {});\n          ws.close();\n          // Authentication headers can appear in WebSocket handshake errors.\n          const statusCode = response.statusCode ?? -1;\n          reject(\n            new APIStatusError({\n              message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n              options: { statusCode },\n            }),\n          );\n        };\n        const onError = (err: Error) => {\n          cleanup();\n          // Transport errors can contain credentials in URLs.\n          reject(\n            new APIConnectionError({\n              message: sanitizedErrorName(err),\n              options: { retryable: true },\n            }),\n          );\n        };\n\n        ws.once('open', onOpen);\n        ws.once('unexpected-response', onUnexpectedResponse);\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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+{"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  APIStatusError,\n  AudioByteStream,\n  DEFAULT_API_CONNECT_OPTIONS,\n  asError,\n  asLanguageCode,\n  calculateAudioDurationSeconds,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  stt,\n  waitForAbort,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { IncomingMessage } from 'node:http';\nimport { WebSocket } from 'ws';\nimport { AUDIO_ENCODING, type STTModel } from './models.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst REQUEST_ID_HEADER = 'cartesia-request-id';\nconst API_VERSION = '2026-03-01';\nconst DRAIN_TIMEOUT_MS = 5000;\nconst KEEPALIVE_INTERVAL_MS = 30000;\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\n/**\n * Fires once when the WebSocket connection is established.\n *\n * You do not need to wait for this event before sending audio.\n */\ntype STTConnectedEvent = {\n  type: 'connected';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Model predicts {@link stt.SpeechEventType.START_OF_SPEECH}.\n */\ntype STTTurnStartEvent = {\n  type: 'turn.start';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires repeatedly as the model transcribes the current user turn.\n *\n * Can be used for {@link stt.SpeechEventType.INTERIM_TRANSCRIPT}.\n */\ntype STTTurnUpdateEvent = {\n  type: 'turn.update';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires when the model predicts that the user might be done speaking.\n *\n * Can be used for {@link stt.SpeechEventType.PREFLIGHT_TRANSCRIPT}.\n */\ntype STTTurnEagerEndEvent = {\n  type: 'turn.eager_end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Fires after {@link STTTurnEagerEndEvent} if the user turn has not actually ended.\n */\ntype STTTurnResumeEvent = {\n  type: 'turn.resume';\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\n/**\n * Marks the end of a user turn.\n *\n * This is used for {@link stt.SpeechEventType.END_OF_SPEECH} and {@link stt.SpeechEventType.FINAL_TRANSCRIPT}.\n */\ntype STTTurnEndEvent = {\n  type: 'turn.end';\n  /**\n   * Cumulative text for the current turn, i.e. the full text transcribed so far in this turn, not a delta.\n   */\n  transcript: string;\n  /**\n   * Unique identifier for this connection. Does not change between turns.\n   */\n  request_id: string;\n};\n\ntype STTErrorEvent = {\n  type: 'error';\n  error_code?: string;\n  status_code?: number;\n  title?: string;\n  message?: string;\n  doc_url?: string;\n  request_id?: string;\n};\n\n/**\n * Server-sent event on the `/stt/turns/websocket` endpoint.\n *\n * See https://docs.cartesia.ai/api-reference/stt/turns/websocket.\n */\ntype STTEventMessage =\n  | STTConnectedEvent\n  | STTTurnStartEvent\n  | STTTurnUpdateEvent\n  | STTTurnEagerEndEvent\n  | STTTurnResumeEvent\n  | STTTurnEndEvent\n  | STTErrorEvent;\n\nexport type STTOptions = {\n  apiKey: string;\n  // eslint-disable-next-line @typescript-eslint/ban-types\n  model: STTModel | (string & {});\n  sampleRate: number;\n  baseUrl: string;\n  audioChunkDurationMS: number;\n  language: string;\n};\n\nconst defaultSTTOptions = {\n  model: 'ink-2' satisfies STTModel,\n  /** recommended default */\n  sampleRate: 16_000,\n  /** recommended default */\n  audioChunkDurationMS: 160,\n  baseUrl: 'https://api.cartesia.ai',\n  language: 'en',\n};\n\nfunction mergeSTTOptions(base: STTOptions, override: Partial<STTOptions>): STTOptions {\n  return {\n    apiKey: override.apiKey ?? base.apiKey,\n    baseUrl: override.baseUrl ?? base.baseUrl,\n    model: override.model ?? base.model,\n    sampleRate: override.sampleRate ?? base.sampleRate,\n    audioChunkDurationMS: override.audioChunkDurationMS ?? base.audioChunkDurationMS,\n    language:\n      override.language !== undefined ? normalizeLanguage(override.language) : base.language,\n  };\n}\n\n/**\n * Resolve the STT model from the configured language when the caller did not\n * pass one explicitly. Mirrors the Python plugin: `ink-2` only supports\n * English, so non-English languages route to the multilingual `ink-whisper`.\n */\nfunction resolveSTTModel(language: string): STTModel {\n  return getBaseLanguage(language) === 'en' ? 'ink-2' : 'ink-whisper';\n}\n\n/**\n * Cartesia speech to text.\n *\n * Supports:\n *  - Streaming\n *  - Turn detection\n *  - Interim results\n *\n * See https://docs.cartesia.ai/build-with-cartesia/stt-models/latest\n *\n * @example\n * ```typescript\n * import { voice } from '@livekit/agents';\n * import { STT, TTS } from '@livekit/agents-plugin-cartesia';\n *\n * const session = new voice.AgentSession({\n *   stt: new STT(),\n *   llm: new LLM(), // choose your favorite LLM\n *   tts: new TTS(),\n *   turnHandling: { turnDetection: 'stt' },\n * });\n * ```\n */\nexport class STT extends stt.STT {\n  #opts: STTOptions;\n\n  constructor(opts: Partial<STTOptions> = {}) {\n    super({\n      streaming: true,\n      interimResults: true,\n      alignedTranscript: false,\n      diarization: false,\n    });\n\n    const apiKey = opts.apiKey ?? process.env.CARTESIA_API_KEY;\n    if (!apiKey) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    this.#opts = mergeSTTOptions({ ...defaultSTTOptions, apiKey }, opts);\n\n    // Route to the multilingual model for non-English languages unless the\n    // caller pinned a model. Matches the Python plugin's language→model logic.\n    if (opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n\n  override get label(): string {\n    return 'cartesia.STT';\n  }\n\n  override get model(): string {\n    return this.#opts.model;\n  }\n\n  override get provider(): string {\n    return 'Cartesia';\n  }\n\n  protected override async _recognize(): Promise<stt.SpeechEvent> {\n    throw new Error('Cartesia STT does not support batch recognition, use stream() instead');\n  }\n\n  override stream(options?: { connOptions?: APIConnectOptions }): SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n\n  updateOptions(opts: Partial<STTOptions>) {\n    this.#opts = mergeSTTOptions(this.#opts, opts);\n\n    // Keep the model in sync with a newly set language (e.g. switching to a\n    // non-English language must move off the English-only ink-2), unless the\n    // caller pinned a model in the same call. Mirrors the constructor.\n    if (opts.language !== undefined && opts.model === undefined) {\n      this.#opts.model = resolveSTTModel(this.#opts.language);\n    }\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  #opts: STTOptions;\n  #logger = log();\n  #ws: WebSocket | null = null;\n  #requestId: string | undefined;\n  #speaking = false;\n  #currentTranscript = '';\n  #speechDuration = 0;\n  #closingWs = false;\n  #connectTimeout: number;\n\n  constructor(sttInstance: STT, opts: STTOptions, connOptions?: APIConnectOptions) {\n    super(sttInstance, opts.sampleRate, connOptions);\n    this.#opts = { ...opts };\n    this.#connectTimeout = connOptions?.timeoutMs ?? DEFAULT_API_CONNECT_OPTIONS.timeoutMs;\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      let ws: WebSocket;\n      try {\n        ws = new WebSocket(url, {\n          handshakeTimeout: this.#connectTimeout,\n          headers: {\n            [AUTHORIZATION_HEADER]: this.#opts.apiKey,\n            [VERSION_HEADER]: API_VERSION,\n          },\n        });\n      } catch (error) {\n        throw new APIConnectionError({\n          message: sanitizedErrorName(asError(error)),\n          options: { retryable: true },\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 cleanup = () => {\n          ws.off('open', onOpen);\n          ws.off('unexpected-response', onUnexpectedResponse);\n          ws.off('error', onError);\n        };\n        const onOpen = () => {\n          cleanup();\n          resolve();\n        };\n        const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n          cleanup();\n          ws.on('error', () => {});\n          ws.close();\n          // Authentication headers can appear in WebSocket handshake errors.\n          const statusCode = response.statusCode ?? -1;\n          reject(\n            new APIStatusError({\n              message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n              options: { statusCode },\n            }),\n          );\n        };\n        const onError = (err: Error) => {\n          cleanup();\n          // Transport errors can contain credentials in URLs.\n          reject(\n            new APIConnectionError({\n              message: sanitizedErrorName(err),\n              options: { retryable: true },\n            }),\n          );\n        };\n\n        ws.once('open', onOpen);\n        ws.once('unexpected-response', onUnexpectedResponse);\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({ 'lk.pii.message': msg }, 'Cartesia sent an error');\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({ 'lk.pii.data': data }, 'received unexpected message from Cartesia STT');\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/tts.cjs

@@ -438,7 +438,7 @@ class SynthesizeStream extends import_agents.tts.SynthesizeStream {
               }
             }
           } else if (!(0, import_types.isFlushDoneMessage)(serverMsg) && !(0, import_types.isErrorMessage)(serverMsg)) {
-            this.#logger.warn({ message: serverMsg }, "Unknown Cartesia message");
+            this.#logger.warn({ "lk.pii.message": serverMsg }, "Unknown Cartesia message");
           }
         }
         if (timedOut) {

package/dist/tts.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  APIError,\n  APIStatusError,\n  APITimeoutError,\n  AudioByteStream,\n  ConnectionPool,\n  Future,\n  type TimedString,\n  asError,\n  createTimedString,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  shortuuid,\n  stream,\n  tokenize,\n  tts,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { request } from 'node:https';\nimport { type RawData, WebSocket } from 'ws';\nimport {\n  TTSDefaultVoiceId,\n  type TTSEncoding,\n  type TTSModels,\n  type TTSVoiceEmotion,\n  type TTSVoiceSpeed,\n  isSonic3,\n} from './models.js';\nimport {\n  type CartesiaServerMessage,\n  cartesiaMessageSchema,\n  hasWordTimestamps,\n  isChunkMessage,\n  isDoneMessage,\n  isErrorMessage,\n  isFlushDoneMessage,\n} from './types.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst API_VERSION = '2025-04-16';\nconst API_VERSION_WITH_EXPERIMENTAL_CONTROLS = '2024-11-13';\nconst MODEL_WITH_EXPERIMENTAL_CONTROLS = 'sonic-2-2025-03-07';\nconst NUM_CHANNELS = 1;\nconst BUFFERED_WORDS_COUNT = 8;\n// Cartesia refreshes a pooled socket after this long so a very long call cannot\n// keep one connection open indefinitely. Matches the Python plugin's 300s.\nconst MAX_SESSION_DURATION_MS = 300_000;\n\n// Lets each SynthesizeStream reach the pool owned by the TTS that created it,\n// without widening the constructor signature the base class fixes.\nconst connectionPools = new WeakMap<TTS, ConnectionPool<WebSocket>>();\n\nexport interface TTSOptions {\n  model: TTSModels | string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  voice: string | number[];\n  speed?: TTSVoiceSpeed | number;\n  emotion?: (TTSVoiceEmotion | string)[];\n  /**\n   * Volume of the speech. For sonic-3, the value is valid between 0.5 and 2.0.\n   * @see https://docs.cartesia.ai/api-reference/tts/bytes#body-generation-config-volume\n   */\n  volume?: number;\n  apiKey?: string;\n  language: string;\n  baseUrl: string;\n  apiVersion: string;\n\n  /**\n   * The timeout for the next chunk to be received from the Cartesia API.\n   */\n  chunkTimeout: number;\n\n  /**\n   * Whether to add word timestamps to the output. When enabled, the TTS will return\n   * timing information for each word in the transcript.\n   * @defaultValue true\n   */\n  wordTimestamps?: boolean;\n\n  pronunciationDictId?: string;\n}\n\nconst defaultTTSOptions: TTSOptions = {\n  model: 'sonic-3',\n  encoding: 'pcm_s16le',\n  sampleRate: 24000,\n  voice: TTSDefaultVoiceId,\n  apiKey: process.env.CARTESIA_API_KEY,\n  language: 'en',\n  baseUrl: 'https://api.cartesia.ai',\n  apiVersion: API_VERSION,\n  chunkTimeout: 5000,\n  wordTimestamps: true,\n};\n\nconst checkGenerationConfig = (opts: TTSOptions) => {\n  const logger = log();\n  if (isSonic3(opts.model)) {\n    if (opts.speed !== undefined && typeof opts.speed === 'number') {\n      if (opts.speed < 0.6 || opts.speed > 2.0) {\n        logger.warn('speed must be between 0.6 and 2.0 for sonic-3');\n      }\n    }\n    if (opts.volume !== undefined && (opts.volume < 0.5 || opts.volume > 2.0)) {\n      logger.warn('volume must be between 0.5 and 2.0 for sonic-3');\n    }\n  } else if (\n    opts.apiVersion !== API_VERSION_WITH_EXPERIMENTAL_CONTROLS ||\n    opts.model !== MODEL_WITH_EXPERIMENTAL_CONTROLS\n  ) {\n    if (opts.speed || opts.emotion) {\n      logger.warn(\n        { model: opts.model, speed: opts.speed, emotion: opts.emotion },\n        `speed and emotion controls are only supported for model '${MODEL_WITH_EXPERIMENTAL_CONTROLS}' ` +\n          `or sonic-3 models, see https://docs.cartesia.ai/developer-tools/changelog for details`,\n      );\n    }\n  }\n\n  if (opts.pronunciationDictId && !isSonic3(opts.model)) {\n    logger.warn(\n      { model: opts.model, pronunciationDictId: opts.pronunciationDictId },\n      'pronunciationDictId is only supported for sonic-3 models',\n    );\n  }\n};\n\nexport class TTS extends tts.TTS {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #closed = false;\n  label = 'cartesia.TTS';\n\n  get model(): string {\n    return this.#opts.model;\n  }\n\n  get provider(): string {\n    return 'Cartesia';\n  }\n\n  constructor(opts: Partial<TTSOptions> = {}) {\n    const resolvedOpts = {\n      ...defaultTTSOptions,\n      ...opts,\n    };\n\n    super(resolvedOpts.sampleRate || defaultTTSOptions.sampleRate, NUM_CHANNELS, {\n      streaming: true,\n      alignedTranscript: resolvedOpts.wordTimestamps ?? true,\n    });\n\n    this.#opts = resolvedOpts;\n    this.#opts.language = normalizeLanguage(this.#opts.language);\n\n    if (this.#opts.apiKey === undefined) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    // One socket, reused across generations. Cartesia recommends a single\n    // preconnected WebSocket for many generations because a fresh connection\n    // repays TCP/TLS setup on every turn:\n    // https://docs.cartesia.ai/use-the-api/compare-tts-endpoints\n    this.#pool = new ConnectionPool<WebSocket>({\n      connectCb: (timeoutMs) => this.#connectWebSocket(timeoutMs),\n      closeCb: async (ws) => safeCloseWebSocket(ws),\n      maxSessionDuration: MAX_SESSION_DURATION_MS,\n      markRefreshedOnGet: true,\n    });\n    connectionPools.set(this, this.#pool);\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    // Only these three fields reach Cartesia at WebSocket-handshake time (auth\n    // header, version header, host). Everything else (model, voice, encoding,\n    // sample rate, speed, emotion, volume, language) is sent in-band on each\n    // generation, so a pooled socket serves the new value without reconnecting.\n    // Reconnect only when one of the handshake inputs actually changes.\n    const handshakeChanged =\n      (opts.apiKey !== undefined && opts.apiKey !== this.#opts.apiKey) ||\n      (opts.apiVersion !== undefined && opts.apiVersion !== this.#opts.apiVersion) ||\n      (opts.baseUrl !== undefined && opts.baseUrl !== this.#opts.baseUrl);\n\n    this.#opts = { ...this.#opts, ...opts };\n    if (opts.language !== undefined) {\n      this.#opts.language = normalizeLanguage(opts.language);\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    if (handshakeChanged) {\n      this.#pool.invalidate();\n    }\n  }\n\n  synthesize(\n    text: string,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ): tts.ChunkedStream {\n    return new ChunkedStream(this, text, { ...this.#opts }, connOptions, abortSignal);\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream {\n    return new SynthesizeStream(this, { ...this.#opts }, options?.connOptions);\n  }\n\n  /**\n   * Open the pooled WebSocket ahead of the first generation so the first turn\n   * does not pay the connect. Safe to call more than once; it is a no-op when a\n   * connection already exists.\n   */\n  prewarm(): void {\n    this.#pool.prewarm();\n  }\n\n  override async close(): Promise<void> {\n    this.#closed = true;\n    await this.#pool.close();\n    await super.close();\n  }\n\n  async #connectWebSocket(timeoutMs: number): Promise<WebSocket> {\n    // Snapshot the handshake inputs. If a concurrent updateOptions() changes one\n    // of them while this connect is in flight, reconnect on the new value rather\n    // than pooling a socket built on stale credentials (mirrors the fishaudio\n    // plugin's model re-check).\n    const apiKey = this.#opts.apiKey!;\n    const apiVersion = this.#opts.apiVersion;\n    const baseUrl = this.#opts.baseUrl;\n    const url = `${baseUrl.replace(/^http/, 'ws')}/tts/websocket`;\n    const ws = await connectCartesiaWebSocket({\n      url,\n      headers: {\n        [AUTHORIZATION_HEADER]: apiKey,\n        [VERSION_HEADER]: apiVersion,\n      },\n      timeoutMs,\n    });\n    if (this.#closed) {\n      safeCloseWebSocket(ws);\n      throw new APIConnectionError({ message: 'Cartesia TTS is closed' });\n    }\n    if (\n      apiKey !== this.#opts.apiKey ||\n      apiVersion !== this.#opts.apiVersion ||\n      baseUrl !== this.#opts.baseUrl\n    ) {\n      safeCloseWebSocket(ws);\n      return await this.#connectWebSocket(timeoutMs);\n    }\n    // Drop a socket that closes (or errors) while idle in the pool. Between turns\n    // no generation listeners are attached, so without this the pool keeps a dead\n    // socket in `available` and the next turn spends a retry to discard it, or\n    // fails outright at maxRetry:0. A generation attaches its own listeners on top\n    // of these; the no-op error listener also stops an idle 'error' from crashing\n    // the process. Remove is a no-op once the socket is no longer pooled, so this\n    // is safe during an active generation and during close().\n    ws.on('error', () => {});\n    ws.on('close', () => this.#pool.remove(ws));\n    return ws;\n  }\n}\n\nexport class ChunkedStream extends tts.ChunkedStream {\n  label = 'cartesia.ChunkedStream';\n  #logger = log();\n  #opts: TTSOptions;\n  #text: string;\n\n  constructor(\n    tts: TTS,\n    text: string,\n    opts: TTSOptions,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ) {\n    super(text, tts, connOptions, abortSignal);\n    this.#text = text;\n    this.#opts = opts;\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n    const json = toCartesiaOptions(this.#opts);\n    json.transcript = this.#text;\n\n    const baseUrl = new URL(this.#opts.baseUrl);\n    const doneFut = new Future<void>();\n\n    const req = request(\n      {\n        hostname: baseUrl.hostname,\n        port: parseInt(baseUrl.port) || (baseUrl.protocol === 'https:' ? 443 : 80),\n        path: '/tts/bytes',\n        method: 'POST',\n        headers: {\n          [AUTHORIZATION_HEADER]: this.#opts.apiKey!,\n          [VERSION_HEADER]: this.#opts.apiVersion,\n        },\n        signal: this.abortSignal,\n      },\n      (res) => {\n        res.on('data', (chunk) => {\n          for (const frame of bstream.write(chunk)) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n        });\n        res.on('close', () => {\n          for (const frame of bstream.flush()) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n          this.queue.close();\n          if (!doneFut.done) doneFut.resolve();\n        });\n        res.on('error', (err) => {\n          if (err.message === 'aborted') return;\n          this.#logger.error({ err }, 'Cartesia TTS response error');\n          if (!doneFut.done) doneFut.reject(err);\n        });\n      },\n    );\n\n    req.on('error', (err) => {\n      if (err.name === 'AbortError') return;\n      this.#logger.error({ err }, 'Cartesia TTS request error');\n      if (!doneFut.done) doneFut.reject(err);\n    });\n    req.on('close', () => {\n      if (!doneFut.done) doneFut.resolve();\n    });\n    req.write(JSON.stringify(json));\n    req.end();\n\n    try {\n      await doneFut.await;\n    } catch (e) {\n      if (this.abortSignal.aborted) return;\n      if (!this.queue.closed) this.queue.close();\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nexport class SynthesizeStream extends tts.SynthesizeStream {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #logger = log();\n  #tokenizer = new tokenize.basic.SentenceTokenizer({\n    minSentenceLength: BUFFERED_WORDS_COUNT,\n  }).stream();\n  label = 'cartesia.SynthesizeStream';\n\n  constructor(tts: TTS, opts: TTSOptions, connOptions?: APIConnectOptions) {\n    super(tts, connOptions);\n    const pool = connectionPools.get(tts);\n    if (!pool) throw new Error('Cartesia connection pool is not initialized');\n    this.#pool = pool;\n    this.#opts = opts;\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    // Only finish the generation once both: 1) Cartesia returns done, AND 2) all sentences have been sent\n    let sentenceStreamClosed = false;\n\n    const sentenceStreamTask = async (ws: WebSocket) => {\n      const packet = toCartesiaOptions(this.#opts, true);\n      for await (const event of this.#tokenizer) {\n        const msg = {\n          ...packet,\n          context_id: requestId,\n          transcript: event.token + ' ',\n          continue: true,\n        };\n        this.markStarted();\n        ws.send(JSON.stringify(msg));\n      }\n\n      const endMsg = {\n        ...packet,\n        context_id: requestId,\n        transcript: ' ',\n        continue: false,\n      };\n      ws.send(JSON.stringify(endMsg));\n      // Mark sentence stream as closed\n      sentenceStreamClosed = true;\n    };\n\n    const inputTask = async () => {\n      for await (const data of this.input) {\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          this.#tokenizer.flush();\n          continue;\n        }\n        this.#tokenizer.pushText(data);\n      }\n      this.#tokenizer.endInput();\n      this.#tokenizer.close();\n    };\n\n    // Use event channel and set up listeners ONCE to avoid missing messages during listener re-registration\n    const recvTask = async (ws: WebSocket) => {\n      const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n\n      // Create event channel to buffer incoming messages\n      // This prevents message loss between listener re-registrations\n      const eventChannel = stream.createStreamChannel<RawData>();\n\n      let lastFrame: AudioFrame | undefined;\n      let pendingTimedTranscripts: TimedString[] = [];\n\n      const sendLastFrame = (segmentId: string, final: boolean) => {\n        if (lastFrame && !this.queue.closed) {\n          // Include timedTranscripts with the audio frame\n          this.queue.put({\n            requestId,\n            segmentId,\n            frame: lastFrame,\n            final,\n            timedTranscripts:\n              pendingTimedTranscripts.length > 0 ? pendingTimedTranscripts : undefined,\n          });\n          lastFrame = undefined;\n          pendingTimedTranscripts = [];\n        }\n      };\n\n      let timeout: NodeJS.Timeout | null = null;\n      // Set when the chunk watchdog fires: the socket is discarded, not pooled.\n      let timedOut = false;\n      // Set once this generation's `done` has been handled. Until then, a socket\n      // close or error is a mid-generation drop, not a normal end.\n      let completed = false;\n      // A socket close/error before completion. Thrown after the loop so the turn\n      // fails over (and the dead socket is discarded) instead of ending silently.\n      let streamError: Error | undefined;\n\n      const clearTTSChunkTimeout = () => {\n        if (timeout) {\n          clearTimeout(timeout);\n          timeout = null;\n        }\n      };\n\n      // Set up WebSocket listeners ONCE (not in a loop)\n      const onMessage = (data: RawData) => {\n        void eventChannel.write(data).catch((error: unknown) => {\n          this.#logger.debug({ error }, 'Failed writing Cartesia event to channel (likely closed)');\n        });\n      };\n\n      const onClose = (code: number, reason: Buffer) => {\n        // A close during an active generation is unexpected: the pool owns the\n        // socket lifecycle and does not close it between turns. If it happens\n        // before `done`, surface it so the turn retries rather than ending mid\n        // speech, and so withConnection discards the dead socket.\n        this.#logger.debug(`WebSocket closed with code ${code}: ${reason.toString()}`);\n        clearTTSChunkTimeout();\n        if (!completed && !timedOut && !streamError) {\n          streamError = new APIConnectionError({\n            message: `Cartesia WebSocket closed mid-generation (code=${code})`,\n          });\n        }\n        void eventChannel.close();\n      };\n\n      const onError = (err: Error) => {\n        this.#logger.error({ err }, 'Cartesia WebSocket error');\n        if (!completed && !timedOut && !streamError) {\n          streamError = err instanceof APIError ? err : toRetryableConnectionError(err);\n        }\n        void eventChannel.close();\n      };\n\n      // Attach listeners ONCE\n      ws.on('message', onMessage);\n      ws.on('close', onClose);\n      ws.on('error', onError);\n\n      try {\n        // Process messages from the channel\n        const reader = eventChannel.stream().getReader();\n\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (result.done) break;\n\n          const rawMsg = result.value;\n\n          // Parse message with Zod schema for type safety\n          let serverMsg: CartesiaServerMessage;\n          try {\n            const json = JSON.parse(rawMsg.toString());\n            serverMsg = cartesiaMessageSchema.parse(json);\n          } catch (parseErr) {\n            this.#logger.warn({ parseErr }, 'Failed to parse Cartesia message');\n            continue;\n          }\n\n          const segmentId = serverMsg.context_id;\n\n          // Handle error frames first. 4xx (e.g. empty-transcript on\n          // function-call turns) is non-fatal — log and fall through so an\n          // accompanying done:true still triggers the unified close path\n          // below. 5xx bubbles up so the base SynthesizeStream can retry.\n          if (isErrorMessage(serverMsg)) {\n            if (serverMsg.status_code >= 400 && serverMsg.status_code < 500) {\n              this.#logger.debug({ error: serverMsg.error }, 'Cartesia sent a non-fatal error');\n            } else {\n              this.#logger.error({ error: serverMsg.error }, 'Cartesia returned error');\n              throw new APIStatusError({\n                message: `Cartesia returned error: ${serverMsg.error}`,\n                options: { statusCode: serverMsg.status_code, retryable: true },\n              });\n            }\n          }\n\n          if (isChunkMessage(serverMsg)) {\n            const audioBuffer = Buffer.from(serverMsg.data, 'base64');\n            // Extract ArrayBuffer from Buffer for AudioByteStream compatibility\n            const audioData = audioBuffer.buffer.slice(\n              audioBuffer.byteOffset,\n              audioBuffer.byteOffset + audioBuffer.byteLength,\n            );\n            for (const frame of bstream.write(audioData)) {\n              sendLastFrame(segmentId, false);\n              lastFrame = frame;\n            }\n\n            // IMPORTANT: close WS if TTS chunk stream been stuck too long\n            // this allows unblock the current \"broken\" TTS node so that any future TTS nodes\n            // can continue to process the stream without been blocked by the stuck node\n            clearTTSChunkTimeout();\n            timeout = setTimeout(() => {\n              // cartesia chunk timeout quite often, so we make it a debug log\n              this.#logger.debug(\n                `Cartesia WebSocket TTS chunk stream timeout after ${this.#opts.chunkTimeout}ms`,\n              );\n              // The socket is stuck mid-generation, so it must not return to the\n              // pool. Poison it and unblock the reader; the post-loop check turns\n              // this into a retryable error so withConnection discards the socket.\n              timedOut = true;\n              safeCloseWebSocket(ws);\n              void eventChannel.close();\n            }, this.#opts.chunkTimeout);\n          } else if (this.#opts.wordTimestamps !== false && hasWordTimestamps(serverMsg)) {\n            const wordTimestamps = serverMsg.word_timestamps;\n            for (let i = 0; i < wordTimestamps.words.length; i++) {\n              const word = wordTimestamps.words[i];\n              const startTime = wordTimestamps.start[i];\n              const endTime = wordTimestamps.end[i];\n              if (word !== undefined && startTime !== undefined && endTime !== undefined) {\n                pendingTimedTranscripts.push(\n                  createTimedString({\n                    text: word + ' ', // Add space after word for consistency\n                    startTime,\n                    endTime,\n                  }),\n                );\n              }\n            }\n          } else if (isDoneMessage(serverMsg) || (isErrorMessage(serverMsg) && serverMsg.done)) {\n            // This ensures all sentences have been sent before closing\n            if (sentenceStreamClosed) {\n              for (const frame of bstream.flush()) {\n                sendLastFrame(segmentId, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(segmentId, true);\n              if (!this.queue.closed) {\n                this.queue.put(SynthesizeStream.END_OF_STREAM);\n              }\n\n              if (segmentId === requestId) {\n                clearTTSChunkTimeout();\n                completed = true;\n                // Leave the socket open so the pool reuses it on the next turn.\n                break; // Exit the loop\n              }\n            }\n            // If sentenceStreamClosed is false, continue receiving - more done messages will come\n          } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {\n            // flush_done is an ack with nothing to do; error frames without\n            // done:true were already logged above.\n            this.#logger.warn({ message: serverMsg }, 'Unknown Cartesia message');\n          }\n        }\n\n        if (timedOut) {\n          throw new APITimeoutError({\n            message: `Cartesia TTS chunk stream timed out after ${this.#opts.chunkTimeout}ms`,\n          });\n        }\n        if (streamError) {\n          throw streamError;\n        }\n      } catch (err) {\n        // Always propagate API errors so the base SynthesizeStream can retry\n        // and emit tts_error once retries are exhausted.\n        if (err instanceof APIError) throw err;\n        // skip log error for normal websocket close\n        if (err instanceof Error && !err.message.includes('WebSocket closed')) {\n          if (\n            err.message.includes('Queue is closed') ||\n            err.message.includes('Channel is closed')\n          ) {\n            this.#logger.warn(\n              { err },\n              'Channel closed during transcript processing (expected during disconnect)',\n            );\n          } else {\n            this.#logger.error({ err }, 'Error in recvTask from Cartesia WebSocket');\n          }\n        }\n      } finally {\n        // IMPORTANT: Remove listeners so connection can be reused\n        ws.off('message', onMessage);\n        ws.off('close', onClose);\n        ws.off('error', onError);\n        clearTTSChunkTimeout();\n      }\n    };\n\n    try {\n      // The pool hands back one live socket per call and reclaims it on success\n      // (put) or discards it on any thrown error (remove). A generation never\n      // closes the socket itself, so the next turn skips the handshake.\n      await this.#pool.withConnection(\n        async (ws) => {\n          if (ws.readyState !== WebSocket.OPEN) {\n            throw new APIConnectionError({ message: 'Cartesia pooled websocket is not open' });\n          }\n          await Promise.all([inputTask(), sentenceStreamTask(ws), recvTask(ws)]);\n        },\n        { timeout: this.connOptions.timeoutMs, signal: this.abortSignal },\n      );\n    } catch (e) {\n      if (this.abortSignal.aborted) {\n        return;\n      }\n      if (e instanceof APIError) throw e;\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nconst transientNetworkCodes = new Set([\n  'ETIMEDOUT',\n  'ECONNRESET',\n  'EAI_AGAIN',\n  'ENETUNREACH',\n  'ECONNREFUSED',\n  'EHOSTUNREACH',\n]);\n\nconst isRecord = (v: unknown): v is Record<string, unknown> => {\n  return v !== null && typeof v === 'object';\n};\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\nconst isAggregateErrorLike = (e: unknown): e is { errors: unknown[]; name?: string } => {\n  if (!isRecord(e)) return false;\n  return e.name === 'AggregateError' && Array.isArray(e.errors);\n};\n\nconst hasErrorCode = (e: unknown, code: string): boolean => {\n  if (isRecord(e) && e.code === code) return true;\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasErrorCode(inner, code));\n  }\n  return false;\n};\n\nconst hasAnyTransientCode = (e: unknown): boolean => {\n  if (isRecord(e) && typeof e.code === 'string') {\n    return transientNetworkCodes.has(e.code);\n  }\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasAnyTransientCode(inner));\n  }\n  return false;\n};\n\nconst toRetryableConnectionError = (e: unknown): APIConnectionError => {\n  const err = asError(e);\n  const isTimeout =\n    hasErrorCode(e, 'ETIMEDOUT') ||\n    (typeof err.message === 'string' && err.message.includes('ETIMEDOUT'));\n  const message = isTimeout\n    ? `Cartesia connection timed out`\n    : `Cartesia connection failed: ${err.message || 'unknown error'}`;\n  return isTimeout ? new APITimeoutError({ message }) : new APIConnectionError({ message });\n};\n\nconst waitForWsOpen = async ({\n  ws,\n  timeoutMs,\n  abortSignal,\n}: {\n  ws: WebSocket;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}) => {\n  if (abortSignal?.aborted) {\n    throw new Error('aborted');\n  }\n\n  const fut = new Future<void>();\n  let timeout: NodeJS.Timeout | undefined;\n\n  const cleanup = () => {\n    if (timeout) clearTimeout(timeout);\n    ws.off('open', onOpen);\n    ws.off('unexpected-response', onUnexpectedResponse);\n    ws.off('error', onError);\n    ws.off('close', onClose);\n    abortSignal?.removeEventListener('abort', onAbort);\n  };\n\n  const onOpen = () => fut.resolve();\n  const onError = (err: Error) => fut.reject(asError(err));\n  const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n    // Authentication headers can appear in WebSocket handshake errors.\n    const statusCode = response.statusCode ?? -1;\n    fut.reject(\n      new APIStatusError({\n        message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n        options: { statusCode },\n      }),\n    );\n  };\n  const onClose = (code: number, reason: Buffer) =>\n    fut.reject(\n      new Error(`WebSocket closed before open (code=${code}, reason=${reason.toString()})`),\n    );\n  const onAbort = () => fut.reject(new Error('aborted'));\n\n  ws.on('open', onOpen);\n  ws.on('unexpected-response', onUnexpectedResponse);\n  ws.on('error', onError);\n  ws.on('close', onClose);\n  abortSignal?.addEventListener('abort', onAbort, { once: true });\n\n  if (timeoutMs > 0) {\n    timeout = setTimeout(\n      () => fut.reject(new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' })),\n      timeoutMs,\n    );\n  }\n\n  try {\n    await fut.await;\n  } finally {\n    cleanup();\n  }\n};\n\nconst safeTerminateWebSocket = (ws: WebSocket) => {\n  // `ws` can emit an 'error' event during teardown (especially if CONNECTING).\n  // If there is no error listener at that moment, Node will treat it as unhandled and crash the process.\n  try {\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    // `terminate()` can throw if the socket was never established; `close()` is safer in CONNECTING.\n    if (ws.readyState === WebSocket.CONNECTING) {\n      ws.close();\n    } else {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\n// Graceful close used by the connection pool. A pooled socket is healthy when it\n// is retired (session age, option change, or TTS close), so a clean close frame\n// is preferable to an abrupt terminate; terminate remains the fallback for a\n// socket caught mid-handshake.\nconst safeCloseWebSocket = (ws: WebSocket) => {\n  try {\n    // `ws` can emit 'error' during teardown; without a listener Node treats it as\n    // unhandled and crashes the process.\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    if (ws.readyState === WebSocket.CONNECTING || ws.readyState === WebSocket.OPEN) {\n      ws.close();\n    } else if (ws.readyState !== WebSocket.CLOSED && ws.readyState !== WebSocket.CLOSING) {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\nconst connectCartesiaWebSocket = async ({\n  url,\n  headers,\n  timeoutMs,\n  abortSignal,\n}: {\n  url: string;\n  headers: Record<string, string>;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}): Promise<WebSocket> => {\n  const connectOnce = async (family?: number): Promise<WebSocket> => {\n    const ws = new WebSocket(url, { handshakeTimeout: timeoutMs, family, headers });\n    try {\n      await waitForWsOpen({ ws, timeoutMs, abortSignal });\n      return ws;\n    } catch (e) {\n      safeTerminateWebSocket(ws);\n      throw e;\n    }\n  };\n\n  let connectError: unknown;\n  try {\n    return await connectOnce();\n  } catch (e) {\n    connectError = e;\n    // Mitigation for Node.js dual-stack (IPv6/IPv4) connect flakiness (\"happy eyeballs\"):\n    // some environments surface `AggregateError` with nested `ETIMEDOUT` during the initial\n    // WebSocket open. In that case we do a one-off retry forcing IPv4 (`family: 4`) before\n    // letting the outer framework retry loop handle further attempts.\n    //\n    // If you still see `AggregateError`/`ETIMEDOUT`:\n    // - Increase the session TTS connect timeout (`connOptions.ttsConnOptions.timeoutMs`)\n    // - Or adjust Node's family autoselection behavior via `NODE_OPTIONS`, e.g.\n    //   `--network-family-autoselection-attempt-timeout=5000` (or disable it entirely).\n    if (!(e instanceof APIError) && (hasAnyTransientCode(e) || isAggregateErrorLike(e))) {\n      try {\n        return await connectOnce(4);\n      } catch (retryError) {\n        connectError = retryError;\n      }\n    }\n  }\n\n  if (connectError instanceof APIError) throw connectError;\n  const error = asError(connectError);\n  const isTimeout =\n    hasErrorCode(connectError, 'ETIMEDOUT') || /timed?\\s*out|timeout/i.test(error.message);\n  if (isTimeout) {\n    throw new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' });\n  }\n  // Transport errors can contain credentials in URLs.\n  throw new APIConnectionError({ message: sanitizedErrorName(error) });\n};\n\nconst toCartesiaOptions = (\n  opts: TTSOptions,\n  streaming: boolean = false,\n): { [id: string]: unknown } => {\n  const voice: { [id: string]: unknown } = {};\n  if (typeof opts.voice === 'string') {\n    voice.mode = 'id';\n    voice.id = opts.voice;\n  } else {\n    voice.mode = 'embedding';\n    voice.embedding = opts.voice;\n  }\n\n  if (opts.apiVersion === API_VERSION_WITH_EXPERIMENTAL_CONTROLS) {\n    const voiceControls: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      voiceControls.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      voiceControls.emotion = opts.emotion;\n    }\n    if (Object.keys(voiceControls).length) {\n      voice.__experimental_controls = voiceControls;\n    }\n  }\n\n  const result: { [id: string]: unknown } = {\n    model_id: opts.model,\n    voice,\n    output_format: {\n      container: 'raw',\n      encoding: opts.encoding,\n      sample_rate: opts.sampleRate,\n    },\n    language: getBaseLanguage(opts.language),\n    max_buffer_delay_ms: 0,\n  };\n\n  if (opts.pronunciationDictId) {\n    result.pronunciation_dict_id = opts.pronunciationDictId;\n  }\n\n  if (opts.apiVersion > API_VERSION_WITH_EXPERIMENTAL_CONTROLS && isSonic3(opts.model)) {\n    const generationConfig: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      generationConfig.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      generationConfig.emotion = opts.emotion[0];\n    }\n    if (opts.volume) {\n      generationConfig.volume = opts.volume;\n    }\n    if (Object.keys(generationConfig).length) {\n      result.generation_config = generationConfig;\n    }\n  }\n\n  if (streaming && opts.wordTimestamps !== false) {\n    result.add_timestamps = true;\n  }\n\n  return 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+{"version":3,"sources":["../src/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  APIError,\n  APIStatusError,\n  APITimeoutError,\n  AudioByteStream,\n  ConnectionPool,\n  Future,\n  type TimedString,\n  asError,\n  createTimedString,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  shortuuid,\n  stream,\n  tokenize,\n  tts,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { request } from 'node:https';\nimport { type RawData, WebSocket } from 'ws';\nimport {\n  TTSDefaultVoiceId,\n  type TTSEncoding,\n  type TTSModels,\n  type TTSVoiceEmotion,\n  type TTSVoiceSpeed,\n  isSonic3,\n} from './models.js';\nimport {\n  type CartesiaServerMessage,\n  cartesiaMessageSchema,\n  hasWordTimestamps,\n  isChunkMessage,\n  isDoneMessage,\n  isErrorMessage,\n  isFlushDoneMessage,\n} from './types.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst API_VERSION = '2025-04-16';\nconst API_VERSION_WITH_EXPERIMENTAL_CONTROLS = '2024-11-13';\nconst MODEL_WITH_EXPERIMENTAL_CONTROLS = 'sonic-2-2025-03-07';\nconst NUM_CHANNELS = 1;\nconst BUFFERED_WORDS_COUNT = 8;\n// Cartesia refreshes a pooled socket after this long so a very long call cannot\n// keep one connection open indefinitely. Matches the Python plugin's 300s.\nconst MAX_SESSION_DURATION_MS = 300_000;\n\n// Lets each SynthesizeStream reach the pool owned by the TTS that created it,\n// without widening the constructor signature the base class fixes.\nconst connectionPools = new WeakMap<TTS, ConnectionPool<WebSocket>>();\n\nexport interface TTSOptions {\n  model: TTSModels | string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  voice: string | number[];\n  speed?: TTSVoiceSpeed | number;\n  emotion?: (TTSVoiceEmotion | string)[];\n  /**\n   * Volume of the speech. For sonic-3, the value is valid between 0.5 and 2.0.\n   * @see https://docs.cartesia.ai/api-reference/tts/bytes#body-generation-config-volume\n   */\n  volume?: number;\n  apiKey?: string;\n  language: string;\n  baseUrl: string;\n  apiVersion: string;\n\n  /**\n   * The timeout for the next chunk to be received from the Cartesia API.\n   */\n  chunkTimeout: number;\n\n  /**\n   * Whether to add word timestamps to the output. When enabled, the TTS will return\n   * timing information for each word in the transcript.\n   * @defaultValue true\n   */\n  wordTimestamps?: boolean;\n\n  pronunciationDictId?: string;\n}\n\nconst defaultTTSOptions: TTSOptions = {\n  model: 'sonic-3',\n  encoding: 'pcm_s16le',\n  sampleRate: 24000,\n  voice: TTSDefaultVoiceId,\n  apiKey: process.env.CARTESIA_API_KEY,\n  language: 'en',\n  baseUrl: 'https://api.cartesia.ai',\n  apiVersion: API_VERSION,\n  chunkTimeout: 5000,\n  wordTimestamps: true,\n};\n\nconst checkGenerationConfig = (opts: TTSOptions) => {\n  const logger = log();\n  if (isSonic3(opts.model)) {\n    if (opts.speed !== undefined && typeof opts.speed === 'number') {\n      if (opts.speed < 0.6 || opts.speed > 2.0) {\n        logger.warn('speed must be between 0.6 and 2.0 for sonic-3');\n      }\n    }\n    if (opts.volume !== undefined && (opts.volume < 0.5 || opts.volume > 2.0)) {\n      logger.warn('volume must be between 0.5 and 2.0 for sonic-3');\n    }\n  } else if (\n    opts.apiVersion !== API_VERSION_WITH_EXPERIMENTAL_CONTROLS ||\n    opts.model !== MODEL_WITH_EXPERIMENTAL_CONTROLS\n  ) {\n    if (opts.speed || opts.emotion) {\n      logger.warn(\n        { model: opts.model, speed: opts.speed, emotion: opts.emotion },\n        `speed and emotion controls are only supported for model '${MODEL_WITH_EXPERIMENTAL_CONTROLS}' ` +\n          `or sonic-3 models, see https://docs.cartesia.ai/developer-tools/changelog for details`,\n      );\n    }\n  }\n\n  if (opts.pronunciationDictId && !isSonic3(opts.model)) {\n    logger.warn(\n      { model: opts.model, pronunciationDictId: opts.pronunciationDictId },\n      'pronunciationDictId is only supported for sonic-3 models',\n    );\n  }\n};\n\nexport class TTS extends tts.TTS {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #closed = false;\n  label = 'cartesia.TTS';\n\n  get model(): string {\n    return this.#opts.model;\n  }\n\n  get provider(): string {\n    return 'Cartesia';\n  }\n\n  constructor(opts: Partial<TTSOptions> = {}) {\n    const resolvedOpts = {\n      ...defaultTTSOptions,\n      ...opts,\n    };\n\n    super(resolvedOpts.sampleRate || defaultTTSOptions.sampleRate, NUM_CHANNELS, {\n      streaming: true,\n      alignedTranscript: resolvedOpts.wordTimestamps ?? true,\n    });\n\n    this.#opts = resolvedOpts;\n    this.#opts.language = normalizeLanguage(this.#opts.language);\n\n    if (this.#opts.apiKey === undefined) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    // One socket, reused across generations. Cartesia recommends a single\n    // preconnected WebSocket for many generations because a fresh connection\n    // repays TCP/TLS setup on every turn:\n    // https://docs.cartesia.ai/use-the-api/compare-tts-endpoints\n    this.#pool = new ConnectionPool<WebSocket>({\n      connectCb: (timeoutMs) => this.#connectWebSocket(timeoutMs),\n      closeCb: async (ws) => safeCloseWebSocket(ws),\n      maxSessionDuration: MAX_SESSION_DURATION_MS,\n      markRefreshedOnGet: true,\n    });\n    connectionPools.set(this, this.#pool);\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    // Only these three fields reach Cartesia at WebSocket-handshake time (auth\n    // header, version header, host). Everything else (model, voice, encoding,\n    // sample rate, speed, emotion, volume, language) is sent in-band on each\n    // generation, so a pooled socket serves the new value without reconnecting.\n    // Reconnect only when one of the handshake inputs actually changes.\n    const handshakeChanged =\n      (opts.apiKey !== undefined && opts.apiKey !== this.#opts.apiKey) ||\n      (opts.apiVersion !== undefined && opts.apiVersion !== this.#opts.apiVersion) ||\n      (opts.baseUrl !== undefined && opts.baseUrl !== this.#opts.baseUrl);\n\n    this.#opts = { ...this.#opts, ...opts };\n    if (opts.language !== undefined) {\n      this.#opts.language = normalizeLanguage(opts.language);\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    if (handshakeChanged) {\n      this.#pool.invalidate();\n    }\n  }\n\n  synthesize(\n    text: string,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ): tts.ChunkedStream {\n    return new ChunkedStream(this, text, { ...this.#opts }, connOptions, abortSignal);\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream {\n    return new SynthesizeStream(this, { ...this.#opts }, options?.connOptions);\n  }\n\n  /**\n   * Open the pooled WebSocket ahead of the first generation so the first turn\n   * does not pay the connect. Safe to call more than once; it is a no-op when a\n   * connection already exists.\n   */\n  prewarm(): void {\n    this.#pool.prewarm();\n  }\n\n  override async close(): Promise<void> {\n    this.#closed = true;\n    await this.#pool.close();\n    await super.close();\n  }\n\n  async #connectWebSocket(timeoutMs: number): Promise<WebSocket> {\n    // Snapshot the handshake inputs. If a concurrent updateOptions() changes one\n    // of them while this connect is in flight, reconnect on the new value rather\n    // than pooling a socket built on stale credentials (mirrors the fishaudio\n    // plugin's model re-check).\n    const apiKey = this.#opts.apiKey!;\n    const apiVersion = this.#opts.apiVersion;\n    const baseUrl = this.#opts.baseUrl;\n    const url = `${baseUrl.replace(/^http/, 'ws')}/tts/websocket`;\n    const ws = await connectCartesiaWebSocket({\n      url,\n      headers: {\n        [AUTHORIZATION_HEADER]: apiKey,\n        [VERSION_HEADER]: apiVersion,\n      },\n      timeoutMs,\n    });\n    if (this.#closed) {\n      safeCloseWebSocket(ws);\n      throw new APIConnectionError({ message: 'Cartesia TTS is closed' });\n    }\n    if (\n      apiKey !== this.#opts.apiKey ||\n      apiVersion !== this.#opts.apiVersion ||\n      baseUrl !== this.#opts.baseUrl\n    ) {\n      safeCloseWebSocket(ws);\n      return await this.#connectWebSocket(timeoutMs);\n    }\n    // Drop a socket that closes (or errors) while idle in the pool. Between turns\n    // no generation listeners are attached, so without this the pool keeps a dead\n    // socket in `available` and the next turn spends a retry to discard it, or\n    // fails outright at maxRetry:0. A generation attaches its own listeners on top\n    // of these; the no-op error listener also stops an idle 'error' from crashing\n    // the process. Remove is a no-op once the socket is no longer pooled, so this\n    // is safe during an active generation and during close().\n    ws.on('error', () => {});\n    ws.on('close', () => this.#pool.remove(ws));\n    return ws;\n  }\n}\n\nexport class ChunkedStream extends tts.ChunkedStream {\n  label = 'cartesia.ChunkedStream';\n  #logger = log();\n  #opts: TTSOptions;\n  #text: string;\n\n  constructor(\n    tts: TTS,\n    text: string,\n    opts: TTSOptions,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ) {\n    super(text, tts, connOptions, abortSignal);\n    this.#text = text;\n    this.#opts = opts;\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n    const json = toCartesiaOptions(this.#opts);\n    json.transcript = this.#text;\n\n    const baseUrl = new URL(this.#opts.baseUrl);\n    const doneFut = new Future<void>();\n\n    const req = request(\n      {\n        hostname: baseUrl.hostname,\n        port: parseInt(baseUrl.port) || (baseUrl.protocol === 'https:' ? 443 : 80),\n        path: '/tts/bytes',\n        method: 'POST',\n        headers: {\n          [AUTHORIZATION_HEADER]: this.#opts.apiKey!,\n          [VERSION_HEADER]: this.#opts.apiVersion,\n        },\n        signal: this.abortSignal,\n      },\n      (res) => {\n        res.on('data', (chunk) => {\n          for (const frame of bstream.write(chunk)) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n        });\n        res.on('close', () => {\n          for (const frame of bstream.flush()) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n          this.queue.close();\n          if (!doneFut.done) doneFut.resolve();\n        });\n        res.on('error', (err) => {\n          if (err.message === 'aborted') return;\n          this.#logger.error({ err }, 'Cartesia TTS response error');\n          if (!doneFut.done) doneFut.reject(err);\n        });\n      },\n    );\n\n    req.on('error', (err) => {\n      if (err.name === 'AbortError') return;\n      this.#logger.error({ err }, 'Cartesia TTS request error');\n      if (!doneFut.done) doneFut.reject(err);\n    });\n    req.on('close', () => {\n      if (!doneFut.done) doneFut.resolve();\n    });\n    req.write(JSON.stringify(json));\n    req.end();\n\n    try {\n      await doneFut.await;\n    } catch (e) {\n      if (this.abortSignal.aborted) return;\n      if (!this.queue.closed) this.queue.close();\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nexport class SynthesizeStream extends tts.SynthesizeStream {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #logger = log();\n  #tokenizer = new tokenize.basic.SentenceTokenizer({\n    minSentenceLength: BUFFERED_WORDS_COUNT,\n  }).stream();\n  label = 'cartesia.SynthesizeStream';\n\n  constructor(tts: TTS, opts: TTSOptions, connOptions?: APIConnectOptions) {\n    super(tts, connOptions);\n    const pool = connectionPools.get(tts);\n    if (!pool) throw new Error('Cartesia connection pool is not initialized');\n    this.#pool = pool;\n    this.#opts = opts;\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    // Only finish the generation once both: 1) Cartesia returns done, AND 2) all sentences have been sent\n    let sentenceStreamClosed = false;\n\n    const sentenceStreamTask = async (ws: WebSocket) => {\n      const packet = toCartesiaOptions(this.#opts, true);\n      for await (const event of this.#tokenizer) {\n        const msg = {\n          ...packet,\n          context_id: requestId,\n          transcript: event.token + ' ',\n          continue: true,\n        };\n        this.markStarted();\n        ws.send(JSON.stringify(msg));\n      }\n\n      const endMsg = {\n        ...packet,\n        context_id: requestId,\n        transcript: ' ',\n        continue: false,\n      };\n      ws.send(JSON.stringify(endMsg));\n      // Mark sentence stream as closed\n      sentenceStreamClosed = true;\n    };\n\n    const inputTask = async () => {\n      for await (const data of this.input) {\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          this.#tokenizer.flush();\n          continue;\n        }\n        this.#tokenizer.pushText(data);\n      }\n      this.#tokenizer.endInput();\n      this.#tokenizer.close();\n    };\n\n    // Use event channel and set up listeners ONCE to avoid missing messages during listener re-registration\n    const recvTask = async (ws: WebSocket) => {\n      const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n\n      // Create event channel to buffer incoming messages\n      // This prevents message loss between listener re-registrations\n      const eventChannel = stream.createStreamChannel<RawData>();\n\n      let lastFrame: AudioFrame | undefined;\n      let pendingTimedTranscripts: TimedString[] = [];\n\n      const sendLastFrame = (segmentId: string, final: boolean) => {\n        if (lastFrame && !this.queue.closed) {\n          // Include timedTranscripts with the audio frame\n          this.queue.put({\n            requestId,\n            segmentId,\n            frame: lastFrame,\n            final,\n            timedTranscripts:\n              pendingTimedTranscripts.length > 0 ? pendingTimedTranscripts : undefined,\n          });\n          lastFrame = undefined;\n          pendingTimedTranscripts = [];\n        }\n      };\n\n      let timeout: NodeJS.Timeout | null = null;\n      // Set when the chunk watchdog fires: the socket is discarded, not pooled.\n      let timedOut = false;\n      // Set once this generation's `done` has been handled. Until then, a socket\n      // close or error is a mid-generation drop, not a normal end.\n      let completed = false;\n      // A socket close/error before completion. Thrown after the loop so the turn\n      // fails over (and the dead socket is discarded) instead of ending silently.\n      let streamError: Error | undefined;\n\n      const clearTTSChunkTimeout = () => {\n        if (timeout) {\n          clearTimeout(timeout);\n          timeout = null;\n        }\n      };\n\n      // Set up WebSocket listeners ONCE (not in a loop)\n      const onMessage = (data: RawData) => {\n        void eventChannel.write(data).catch((error: unknown) => {\n          this.#logger.debug({ error }, 'Failed writing Cartesia event to channel (likely closed)');\n        });\n      };\n\n      const onClose = (code: number, reason: Buffer) => {\n        // A close during an active generation is unexpected: the pool owns the\n        // socket lifecycle and does not close it between turns. If it happens\n        // before `done`, surface it so the turn retries rather than ending mid\n        // speech, and so withConnection discards the dead socket.\n        this.#logger.debug(`WebSocket closed with code ${code}: ${reason.toString()}`);\n        clearTTSChunkTimeout();\n        if (!completed && !timedOut && !streamError) {\n          streamError = new APIConnectionError({\n            message: `Cartesia WebSocket closed mid-generation (code=${code})`,\n          });\n        }\n        void eventChannel.close();\n      };\n\n      const onError = (err: Error) => {\n        this.#logger.error({ err }, 'Cartesia WebSocket error');\n        if (!completed && !timedOut && !streamError) {\n          streamError = err instanceof APIError ? err : toRetryableConnectionError(err);\n        }\n        void eventChannel.close();\n      };\n\n      // Attach listeners ONCE\n      ws.on('message', onMessage);\n      ws.on('close', onClose);\n      ws.on('error', onError);\n\n      try {\n        // Process messages from the channel\n        const reader = eventChannel.stream().getReader();\n\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (result.done) break;\n\n          const rawMsg = result.value;\n\n          // Parse message with Zod schema for type safety\n          let serverMsg: CartesiaServerMessage;\n          try {\n            const json = JSON.parse(rawMsg.toString());\n            serverMsg = cartesiaMessageSchema.parse(json);\n          } catch (parseErr) {\n            this.#logger.warn({ parseErr }, 'Failed to parse Cartesia message');\n            continue;\n          }\n\n          const segmentId = serverMsg.context_id;\n\n          // Handle error frames first. 4xx (e.g. empty-transcript on\n          // function-call turns) is non-fatal — log and fall through so an\n          // accompanying done:true still triggers the unified close path\n          // below. 5xx bubbles up so the base SynthesizeStream can retry.\n          if (isErrorMessage(serverMsg)) {\n            if (serverMsg.status_code >= 400 && serverMsg.status_code < 500) {\n              this.#logger.debug({ error: serverMsg.error }, 'Cartesia sent a non-fatal error');\n            } else {\n              this.#logger.error({ error: serverMsg.error }, 'Cartesia returned error');\n              throw new APIStatusError({\n                message: `Cartesia returned error: ${serverMsg.error}`,\n                options: { statusCode: serverMsg.status_code, retryable: true },\n              });\n            }\n          }\n\n          if (isChunkMessage(serverMsg)) {\n            const audioBuffer = Buffer.from(serverMsg.data, 'base64');\n            // Extract ArrayBuffer from Buffer for AudioByteStream compatibility\n            const audioData = audioBuffer.buffer.slice(\n              audioBuffer.byteOffset,\n              audioBuffer.byteOffset + audioBuffer.byteLength,\n            );\n            for (const frame of bstream.write(audioData)) {\n              sendLastFrame(segmentId, false);\n              lastFrame = frame;\n            }\n\n            // IMPORTANT: close WS if TTS chunk stream been stuck too long\n            // this allows unblock the current \"broken\" TTS node so that any future TTS nodes\n            // can continue to process the stream without been blocked by the stuck node\n            clearTTSChunkTimeout();\n            timeout = setTimeout(() => {\n              // cartesia chunk timeout quite often, so we make it a debug log\n              this.#logger.debug(\n                `Cartesia WebSocket TTS chunk stream timeout after ${this.#opts.chunkTimeout}ms`,\n              );\n              // The socket is stuck mid-generation, so it must not return to the\n              // pool. Poison it and unblock the reader; the post-loop check turns\n              // this into a retryable error so withConnection discards the socket.\n              timedOut = true;\n              safeCloseWebSocket(ws);\n              void eventChannel.close();\n            }, this.#opts.chunkTimeout);\n          } else if (this.#opts.wordTimestamps !== false && hasWordTimestamps(serverMsg)) {\n            const wordTimestamps = serverMsg.word_timestamps;\n            for (let i = 0; i < wordTimestamps.words.length; i++) {\n              const word = wordTimestamps.words[i];\n              const startTime = wordTimestamps.start[i];\n              const endTime = wordTimestamps.end[i];\n              if (word !== undefined && startTime !== undefined && endTime !== undefined) {\n                pendingTimedTranscripts.push(\n                  createTimedString({\n                    text: word + ' ', // Add space after word for consistency\n                    startTime,\n                    endTime,\n                  }),\n                );\n              }\n            }\n          } else if (isDoneMessage(serverMsg) || (isErrorMessage(serverMsg) && serverMsg.done)) {\n            // This ensures all sentences have been sent before closing\n            if (sentenceStreamClosed) {\n              for (const frame of bstream.flush()) {\n                sendLastFrame(segmentId, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(segmentId, true);\n              if (!this.queue.closed) {\n                this.queue.put(SynthesizeStream.END_OF_STREAM);\n              }\n\n              if (segmentId === requestId) {\n                clearTTSChunkTimeout();\n                completed = true;\n                // Leave the socket open so the pool reuses it on the next turn.\n                break; // Exit the loop\n              }\n            }\n            // If sentenceStreamClosed is false, continue receiving - more done messages will come\n          } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {\n            // flush_done is an ack with nothing to do; error frames without\n            // done:true were already logged above.\n            this.#logger.warn({ 'lk.pii.message': serverMsg }, 'Unknown Cartesia message');\n          }\n        }\n\n        if (timedOut) {\n          throw new APITimeoutError({\n            message: `Cartesia TTS chunk stream timed out after ${this.#opts.chunkTimeout}ms`,\n          });\n        }\n        if (streamError) {\n          throw streamError;\n        }\n      } catch (err) {\n        // Always propagate API errors so the base SynthesizeStream can retry\n        // and emit tts_error once retries are exhausted.\n        if (err instanceof APIError) throw err;\n        // skip log error for normal websocket close\n        if (err instanceof Error && !err.message.includes('WebSocket closed')) {\n          if (\n            err.message.includes('Queue is closed') ||\n            err.message.includes('Channel is closed')\n          ) {\n            this.#logger.warn(\n              { err },\n              'Channel closed during transcript processing (expected during disconnect)',\n            );\n          } else {\n            this.#logger.error({ err }, 'Error in recvTask from Cartesia WebSocket');\n          }\n        }\n      } finally {\n        // IMPORTANT: Remove listeners so connection can be reused\n        ws.off('message', onMessage);\n        ws.off('close', onClose);\n        ws.off('error', onError);\n        clearTTSChunkTimeout();\n      }\n    };\n\n    try {\n      // The pool hands back one live socket per call and reclaims it on success\n      // (put) or discards it on any thrown error (remove). A generation never\n      // closes the socket itself, so the next turn skips the handshake.\n      await this.#pool.withConnection(\n        async (ws) => {\n          if (ws.readyState !== WebSocket.OPEN) {\n            throw new APIConnectionError({ message: 'Cartesia pooled websocket is not open' });\n          }\n          await Promise.all([inputTask(), sentenceStreamTask(ws), recvTask(ws)]);\n        },\n        { timeout: this.connOptions.timeoutMs, signal: this.abortSignal },\n      );\n    } catch (e) {\n      if (this.abortSignal.aborted) {\n        return;\n      }\n      if (e instanceof APIError) throw e;\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nconst transientNetworkCodes = new Set([\n  'ETIMEDOUT',\n  'ECONNRESET',\n  'EAI_AGAIN',\n  'ENETUNREACH',\n  'ECONNREFUSED',\n  'EHOSTUNREACH',\n]);\n\nconst isRecord = (v: unknown): v is Record<string, unknown> => {\n  return v !== null && typeof v === 'object';\n};\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\nconst isAggregateErrorLike = (e: unknown): e is { errors: unknown[]; name?: string } => {\n  if (!isRecord(e)) return false;\n  return e.name === 'AggregateError' && Array.isArray(e.errors);\n};\n\nconst hasErrorCode = (e: unknown, code: string): boolean => {\n  if (isRecord(e) && e.code === code) return true;\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasErrorCode(inner, code));\n  }\n  return false;\n};\n\nconst hasAnyTransientCode = (e: unknown): boolean => {\n  if (isRecord(e) && typeof e.code === 'string') {\n    return transientNetworkCodes.has(e.code);\n  }\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasAnyTransientCode(inner));\n  }\n  return false;\n};\n\nconst toRetryableConnectionError = (e: unknown): APIConnectionError => {\n  const err = asError(e);\n  const isTimeout =\n    hasErrorCode(e, 'ETIMEDOUT') ||\n    (typeof err.message === 'string' && err.message.includes('ETIMEDOUT'));\n  const message = isTimeout\n    ? `Cartesia connection timed out`\n    : `Cartesia connection failed: ${err.message || 'unknown error'}`;\n  return isTimeout ? new APITimeoutError({ message }) : new APIConnectionError({ message });\n};\n\nconst waitForWsOpen = async ({\n  ws,\n  timeoutMs,\n  abortSignal,\n}: {\n  ws: WebSocket;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}) => {\n  if (abortSignal?.aborted) {\n    throw new Error('aborted');\n  }\n\n  const fut = new Future<void>();\n  let timeout: NodeJS.Timeout | undefined;\n\n  const cleanup = () => {\n    if (timeout) clearTimeout(timeout);\n    ws.off('open', onOpen);\n    ws.off('unexpected-response', onUnexpectedResponse);\n    ws.off('error', onError);\n    ws.off('close', onClose);\n    abortSignal?.removeEventListener('abort', onAbort);\n  };\n\n  const onOpen = () => fut.resolve();\n  const onError = (err: Error) => fut.reject(asError(err));\n  const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n    // Authentication headers can appear in WebSocket handshake errors.\n    const statusCode = response.statusCode ?? -1;\n    fut.reject(\n      new APIStatusError({\n        message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n        options: { statusCode },\n      }),\n    );\n  };\n  const onClose = (code: number, reason: Buffer) =>\n    fut.reject(\n      new Error(`WebSocket closed before open (code=${code}, reason=${reason.toString()})`),\n    );\n  const onAbort = () => fut.reject(new Error('aborted'));\n\n  ws.on('open', onOpen);\n  ws.on('unexpected-response', onUnexpectedResponse);\n  ws.on('error', onError);\n  ws.on('close', onClose);\n  abortSignal?.addEventListener('abort', onAbort, { once: true });\n\n  if (timeoutMs > 0) {\n    timeout = setTimeout(\n      () => fut.reject(new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' })),\n      timeoutMs,\n    );\n  }\n\n  try {\n    await fut.await;\n  } finally {\n    cleanup();\n  }\n};\n\nconst safeTerminateWebSocket = (ws: WebSocket) => {\n  // `ws` can emit an 'error' event during teardown (especially if CONNECTING).\n  // If there is no error listener at that moment, Node will treat it as unhandled and crash the process.\n  try {\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    // `terminate()` can throw if the socket was never established; `close()` is safer in CONNECTING.\n    if (ws.readyState === WebSocket.CONNECTING) {\n      ws.close();\n    } else {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\n// Graceful close used by the connection pool. A pooled socket is healthy when it\n// is retired (session age, option change, or TTS close), so a clean close frame\n// is preferable to an abrupt terminate; terminate remains the fallback for a\n// socket caught mid-handshake.\nconst safeCloseWebSocket = (ws: WebSocket) => {\n  try {\n    // `ws` can emit 'error' during teardown; without a listener Node treats it as\n    // unhandled and crashes the process.\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    if (ws.readyState === WebSocket.CONNECTING || ws.readyState === WebSocket.OPEN) {\n      ws.close();\n    } else if (ws.readyState !== WebSocket.CLOSED && ws.readyState !== WebSocket.CLOSING) {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\nconst connectCartesiaWebSocket = async ({\n  url,\n  headers,\n  timeoutMs,\n  abortSignal,\n}: {\n  url: string;\n  headers: Record<string, string>;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}): Promise<WebSocket> => {\n  const connectOnce = async (family?: number): Promise<WebSocket> => {\n    const ws = new WebSocket(url, { handshakeTimeout: timeoutMs, family, headers });\n    try {\n      await waitForWsOpen({ ws, timeoutMs, abortSignal });\n      return ws;\n    } catch (e) {\n      safeTerminateWebSocket(ws);\n      throw e;\n    }\n  };\n\n  let connectError: unknown;\n  try {\n    return await connectOnce();\n  } catch (e) {\n    connectError = e;\n    // Mitigation for Node.js dual-stack (IPv6/IPv4) connect flakiness (\"happy eyeballs\"):\n    // some environments surface `AggregateError` with nested `ETIMEDOUT` during the initial\n    // WebSocket open. In that case we do a one-off retry forcing IPv4 (`family: 4`) before\n    // letting the outer framework retry loop handle further attempts.\n    //\n    // If you still see `AggregateError`/`ETIMEDOUT`:\n    // - Increase the session TTS connect timeout (`connOptions.ttsConnOptions.timeoutMs`)\n    // - Or adjust Node's family autoselection behavior via `NODE_OPTIONS`, e.g.\n    //   `--network-family-autoselection-attempt-timeout=5000` (or disable it entirely).\n    if (!(e instanceof APIError) && (hasAnyTransientCode(e) || isAggregateErrorLike(e))) {\n      try {\n        return await connectOnce(4);\n      } catch (retryError) {\n        connectError = retryError;\n      }\n    }\n  }\n\n  if (connectError instanceof APIError) throw connectError;\n  const error = asError(connectError);\n  const isTimeout =\n    hasErrorCode(connectError, 'ETIMEDOUT') || /timed?\\s*out|timeout/i.test(error.message);\n  if (isTimeout) {\n    throw new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' });\n  }\n  // Transport errors can contain credentials in URLs.\n  throw new APIConnectionError({ message: sanitizedErrorName(error) });\n};\n\nconst toCartesiaOptions = (\n  opts: TTSOptions,\n  streaming: boolean = false,\n): { [id: string]: unknown } => {\n  const voice: { [id: string]: unknown } = {};\n  if (typeof opts.voice === 'string') {\n    voice.mode = 'id';\n    voice.id = opts.voice;\n  } else {\n    voice.mode = 'embedding';\n    voice.embedding = opts.voice;\n  }\n\n  if (opts.apiVersion === API_VERSION_WITH_EXPERIMENTAL_CONTROLS) {\n    const voiceControls: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      voiceControls.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      voiceControls.emotion = opts.emotion;\n    }\n    if (Object.keys(voiceControls).length) {\n      voice.__experimental_controls = voiceControls;\n    }\n  }\n\n  const result: { [id: string]: unknown } = {\n    model_id: opts.model,\n    voice,\n    output_format: {\n      container: 'raw',\n      encoding: opts.encoding,\n      sample_rate: opts.sampleRate,\n    },\n    language: getBaseLanguage(opts.language),\n    max_buffer_delay_ms: 0,\n  };\n\n  if (opts.pronunciationDictId) {\n    result.pronunciation_dict_id = opts.pronunciationDictId;\n  }\n\n  if (opts.apiVersion > API_VERSION_WITH_EXPERIMENTAL_CONTROLS && isSonic3(opts.model)) {\n    const generationConfig: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      generationConfig.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      generationConfig.emotion = opts.emotion[0];\n    }\n    if (opts.volume) {\n      generationConfig.volume = opts.volume;\n    }\n    if (Object.keys(generationConfig).length) {\n      result.generation_config = generationConfig;\n    }\n  }\n\n  if (streaming && opts.wordTimestamps !== false) {\n    result.add_timestamps = true;\n  }\n\n  return 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package/dist/tts.js

@@ -440,7 +440,7 @@ class SynthesizeStream extends tts.SynthesizeStream {
               }
             }
           } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {
-            this.#logger.warn({ message: serverMsg }, "Unknown Cartesia message");
+            this.#logger.warn({ "lk.pii.message": serverMsg }, "Unknown Cartesia message");
           }
         }
         if (timedOut) {

package/dist/tts.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  APIError,\n  APIStatusError,\n  APITimeoutError,\n  AudioByteStream,\n  ConnectionPool,\n  Future,\n  type TimedString,\n  asError,\n  createTimedString,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  shortuuid,\n  stream,\n  tokenize,\n  tts,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { request } from 'node:https';\nimport { type RawData, WebSocket } from 'ws';\nimport {\n  TTSDefaultVoiceId,\n  type TTSEncoding,\n  type TTSModels,\n  type TTSVoiceEmotion,\n  type TTSVoiceSpeed,\n  isSonic3,\n} from './models.js';\nimport {\n  type CartesiaServerMessage,\n  cartesiaMessageSchema,\n  hasWordTimestamps,\n  isChunkMessage,\n  isDoneMessage,\n  isErrorMessage,\n  isFlushDoneMessage,\n} from './types.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst API_VERSION = '2025-04-16';\nconst API_VERSION_WITH_EXPERIMENTAL_CONTROLS = '2024-11-13';\nconst MODEL_WITH_EXPERIMENTAL_CONTROLS = 'sonic-2-2025-03-07';\nconst NUM_CHANNELS = 1;\nconst BUFFERED_WORDS_COUNT = 8;\n// Cartesia refreshes a pooled socket after this long so a very long call cannot\n// keep one connection open indefinitely. Matches the Python plugin's 300s.\nconst MAX_SESSION_DURATION_MS = 300_000;\n\n// Lets each SynthesizeStream reach the pool owned by the TTS that created it,\n// without widening the constructor signature the base class fixes.\nconst connectionPools = new WeakMap<TTS, ConnectionPool<WebSocket>>();\n\nexport interface TTSOptions {\n  model: TTSModels | string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  voice: string | number[];\n  speed?: TTSVoiceSpeed | number;\n  emotion?: (TTSVoiceEmotion | string)[];\n  /**\n   * Volume of the speech. For sonic-3, the value is valid between 0.5 and 2.0.\n   * @see https://docs.cartesia.ai/api-reference/tts/bytes#body-generation-config-volume\n   */\n  volume?: number;\n  apiKey?: string;\n  language: string;\n  baseUrl: string;\n  apiVersion: string;\n\n  /**\n   * The timeout for the next chunk to be received from the Cartesia API.\n   */\n  chunkTimeout: number;\n\n  /**\n   * Whether to add word timestamps to the output. When enabled, the TTS will return\n   * timing information for each word in the transcript.\n   * @defaultValue true\n   */\n  wordTimestamps?: boolean;\n\n  pronunciationDictId?: string;\n}\n\nconst defaultTTSOptions: TTSOptions = {\n  model: 'sonic-3',\n  encoding: 'pcm_s16le',\n  sampleRate: 24000,\n  voice: TTSDefaultVoiceId,\n  apiKey: process.env.CARTESIA_API_KEY,\n  language: 'en',\n  baseUrl: 'https://api.cartesia.ai',\n  apiVersion: API_VERSION,\n  chunkTimeout: 5000,\n  wordTimestamps: true,\n};\n\nconst checkGenerationConfig = (opts: TTSOptions) => {\n  const logger = log();\n  if (isSonic3(opts.model)) {\n    if (opts.speed !== undefined && typeof opts.speed === 'number') {\n      if (opts.speed < 0.6 || opts.speed > 2.0) {\n        logger.warn('speed must be between 0.6 and 2.0 for sonic-3');\n      }\n    }\n    if (opts.volume !== undefined && (opts.volume < 0.5 || opts.volume > 2.0)) {\n      logger.warn('volume must be between 0.5 and 2.0 for sonic-3');\n    }\n  } else if (\n    opts.apiVersion !== API_VERSION_WITH_EXPERIMENTAL_CONTROLS ||\n    opts.model !== MODEL_WITH_EXPERIMENTAL_CONTROLS\n  ) {\n    if (opts.speed || opts.emotion) {\n      logger.warn(\n        { model: opts.model, speed: opts.speed, emotion: opts.emotion },\n        `speed and emotion controls are only supported for model '${MODEL_WITH_EXPERIMENTAL_CONTROLS}' ` +\n          `or sonic-3 models, see https://docs.cartesia.ai/developer-tools/changelog for details`,\n      );\n    }\n  }\n\n  if (opts.pronunciationDictId && !isSonic3(opts.model)) {\n    logger.warn(\n      { model: opts.model, pronunciationDictId: opts.pronunciationDictId },\n      'pronunciationDictId is only supported for sonic-3 models',\n    );\n  }\n};\n\nexport class TTS extends tts.TTS {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #closed = false;\n  label = 'cartesia.TTS';\n\n  get model(): string {\n    return this.#opts.model;\n  }\n\n  get provider(): string {\n    return 'Cartesia';\n  }\n\n  constructor(opts: Partial<TTSOptions> = {}) {\n    const resolvedOpts = {\n      ...defaultTTSOptions,\n      ...opts,\n    };\n\n    super(resolvedOpts.sampleRate || defaultTTSOptions.sampleRate, NUM_CHANNELS, {\n      streaming: true,\n      alignedTranscript: resolvedOpts.wordTimestamps ?? true,\n    });\n\n    this.#opts = resolvedOpts;\n    this.#opts.language = normalizeLanguage(this.#opts.language);\n\n    if (this.#opts.apiKey === undefined) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    // One socket, reused across generations. Cartesia recommends a single\n    // preconnected WebSocket for many generations because a fresh connection\n    // repays TCP/TLS setup on every turn:\n    // https://docs.cartesia.ai/use-the-api/compare-tts-endpoints\n    this.#pool = new ConnectionPool<WebSocket>({\n      connectCb: (timeoutMs) => this.#connectWebSocket(timeoutMs),\n      closeCb: async (ws) => safeCloseWebSocket(ws),\n      maxSessionDuration: MAX_SESSION_DURATION_MS,\n      markRefreshedOnGet: true,\n    });\n    connectionPools.set(this, this.#pool);\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    // Only these three fields reach Cartesia at WebSocket-handshake time (auth\n    // header, version header, host). Everything else (model, voice, encoding,\n    // sample rate, speed, emotion, volume, language) is sent in-band on each\n    // generation, so a pooled socket serves the new value without reconnecting.\n    // Reconnect only when one of the handshake inputs actually changes.\n    const handshakeChanged =\n      (opts.apiKey !== undefined && opts.apiKey !== this.#opts.apiKey) ||\n      (opts.apiVersion !== undefined && opts.apiVersion !== this.#opts.apiVersion) ||\n      (opts.baseUrl !== undefined && opts.baseUrl !== this.#opts.baseUrl);\n\n    this.#opts = { ...this.#opts, ...opts };\n    if (opts.language !== undefined) {\n      this.#opts.language = normalizeLanguage(opts.language);\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    if (handshakeChanged) {\n      this.#pool.invalidate();\n    }\n  }\n\n  synthesize(\n    text: string,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ): tts.ChunkedStream {\n    return new ChunkedStream(this, text, { ...this.#opts }, connOptions, abortSignal);\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream {\n    return new SynthesizeStream(this, { ...this.#opts }, options?.connOptions);\n  }\n\n  /**\n   * Open the pooled WebSocket ahead of the first generation so the first turn\n   * does not pay the connect. Safe to call more than once; it is a no-op when a\n   * connection already exists.\n   */\n  prewarm(): void {\n    this.#pool.prewarm();\n  }\n\n  override async close(): Promise<void> {\n    this.#closed = true;\n    await this.#pool.close();\n    await super.close();\n  }\n\n  async #connectWebSocket(timeoutMs: number): Promise<WebSocket> {\n    // Snapshot the handshake inputs. If a concurrent updateOptions() changes one\n    // of them while this connect is in flight, reconnect on the new value rather\n    // than pooling a socket built on stale credentials (mirrors the fishaudio\n    // plugin's model re-check).\n    const apiKey = this.#opts.apiKey!;\n    const apiVersion = this.#opts.apiVersion;\n    const baseUrl = this.#opts.baseUrl;\n    const url = `${baseUrl.replace(/^http/, 'ws')}/tts/websocket`;\n    const ws = await connectCartesiaWebSocket({\n      url,\n      headers: {\n        [AUTHORIZATION_HEADER]: apiKey,\n        [VERSION_HEADER]: apiVersion,\n      },\n      timeoutMs,\n    });\n    if (this.#closed) {\n      safeCloseWebSocket(ws);\n      throw new APIConnectionError({ message: 'Cartesia TTS is closed' });\n    }\n    if (\n      apiKey !== this.#opts.apiKey ||\n      apiVersion !== this.#opts.apiVersion ||\n      baseUrl !== this.#opts.baseUrl\n    ) {\n      safeCloseWebSocket(ws);\n      return await this.#connectWebSocket(timeoutMs);\n    }\n    // Drop a socket that closes (or errors) while idle in the pool. Between turns\n    // no generation listeners are attached, so without this the pool keeps a dead\n    // socket in `available` and the next turn spends a retry to discard it, or\n    // fails outright at maxRetry:0. A generation attaches its own listeners on top\n    // of these; the no-op error listener also stops an idle 'error' from crashing\n    // the process. Remove is a no-op once the socket is no longer pooled, so this\n    // is safe during an active generation and during close().\n    ws.on('error', () => {});\n    ws.on('close', () => this.#pool.remove(ws));\n    return ws;\n  }\n}\n\nexport class ChunkedStream extends tts.ChunkedStream {\n  label = 'cartesia.ChunkedStream';\n  #logger = log();\n  #opts: TTSOptions;\n  #text: string;\n\n  constructor(\n    tts: TTS,\n    text: string,\n    opts: TTSOptions,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ) {\n    super(text, tts, connOptions, abortSignal);\n    this.#text = text;\n    this.#opts = opts;\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n    const json = toCartesiaOptions(this.#opts);\n    json.transcript = this.#text;\n\n    const baseUrl = new URL(this.#opts.baseUrl);\n    const doneFut = new Future<void>();\n\n    const req = request(\n      {\n        hostname: baseUrl.hostname,\n        port: parseInt(baseUrl.port) || (baseUrl.protocol === 'https:' ? 443 : 80),\n        path: '/tts/bytes',\n        method: 'POST',\n        headers: {\n          [AUTHORIZATION_HEADER]: this.#opts.apiKey!,\n          [VERSION_HEADER]: this.#opts.apiVersion,\n        },\n        signal: this.abortSignal,\n      },\n      (res) => {\n        res.on('data', (chunk) => {\n          for (const frame of bstream.write(chunk)) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n        });\n        res.on('close', () => {\n          for (const frame of bstream.flush()) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n          this.queue.close();\n          if (!doneFut.done) doneFut.resolve();\n        });\n        res.on('error', (err) => {\n          if (err.message === 'aborted') return;\n          this.#logger.error({ err }, 'Cartesia TTS response error');\n          if (!doneFut.done) doneFut.reject(err);\n        });\n      },\n    );\n\n    req.on('error', (err) => {\n      if (err.name === 'AbortError') return;\n      this.#logger.error({ err }, 'Cartesia TTS request error');\n      if (!doneFut.done) doneFut.reject(err);\n    });\n    req.on('close', () => {\n      if (!doneFut.done) doneFut.resolve();\n    });\n    req.write(JSON.stringify(json));\n    req.end();\n\n    try {\n      await doneFut.await;\n    } catch (e) {\n      if (this.abortSignal.aborted) return;\n      if (!this.queue.closed) this.queue.close();\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nexport class SynthesizeStream extends tts.SynthesizeStream {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #logger = log();\n  #tokenizer = new tokenize.basic.SentenceTokenizer({\n    minSentenceLength: BUFFERED_WORDS_COUNT,\n  }).stream();\n  label = 'cartesia.SynthesizeStream';\n\n  constructor(tts: TTS, opts: TTSOptions, connOptions?: APIConnectOptions) {\n    super(tts, connOptions);\n    const pool = connectionPools.get(tts);\n    if (!pool) throw new Error('Cartesia connection pool is not initialized');\n    this.#pool = pool;\n    this.#opts = opts;\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    // Only finish the generation once both: 1) Cartesia returns done, AND 2) all sentences have been sent\n    let sentenceStreamClosed = false;\n\n    const sentenceStreamTask = async (ws: WebSocket) => {\n      const packet = toCartesiaOptions(this.#opts, true);\n      for await (const event of this.#tokenizer) {\n        const msg = {\n          ...packet,\n          context_id: requestId,\n          transcript: event.token + ' ',\n          continue: true,\n        };\n        this.markStarted();\n        ws.send(JSON.stringify(msg));\n      }\n\n      const endMsg = {\n        ...packet,\n        context_id: requestId,\n        transcript: ' ',\n        continue: false,\n      };\n      ws.send(JSON.stringify(endMsg));\n      // Mark sentence stream as closed\n      sentenceStreamClosed = true;\n    };\n\n    const inputTask = async () => {\n      for await (const data of this.input) {\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          this.#tokenizer.flush();\n          continue;\n        }\n        this.#tokenizer.pushText(data);\n      }\n      this.#tokenizer.endInput();\n      this.#tokenizer.close();\n    };\n\n    // Use event channel and set up listeners ONCE to avoid missing messages during listener re-registration\n    const recvTask = async (ws: WebSocket) => {\n      const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n\n      // Create event channel to buffer incoming messages\n      // This prevents message loss between listener re-registrations\n      const eventChannel = stream.createStreamChannel<RawData>();\n\n      let lastFrame: AudioFrame | undefined;\n      let pendingTimedTranscripts: TimedString[] = [];\n\n      const sendLastFrame = (segmentId: string, final: boolean) => {\n        if (lastFrame && !this.queue.closed) {\n          // Include timedTranscripts with the audio frame\n          this.queue.put({\n            requestId,\n            segmentId,\n            frame: lastFrame,\n            final,\n            timedTranscripts:\n              pendingTimedTranscripts.length > 0 ? pendingTimedTranscripts : undefined,\n          });\n          lastFrame = undefined;\n          pendingTimedTranscripts = [];\n        }\n      };\n\n      let timeout: NodeJS.Timeout | null = null;\n      // Set when the chunk watchdog fires: the socket is discarded, not pooled.\n      let timedOut = false;\n      // Set once this generation's `done` has been handled. Until then, a socket\n      // close or error is a mid-generation drop, not a normal end.\n      let completed = false;\n      // A socket close/error before completion. Thrown after the loop so the turn\n      // fails over (and the dead socket is discarded) instead of ending silently.\n      let streamError: Error | undefined;\n\n      const clearTTSChunkTimeout = () => {\n        if (timeout) {\n          clearTimeout(timeout);\n          timeout = null;\n        }\n      };\n\n      // Set up WebSocket listeners ONCE (not in a loop)\n      const onMessage = (data: RawData) => {\n        void eventChannel.write(data).catch((error: unknown) => {\n          this.#logger.debug({ error }, 'Failed writing Cartesia event to channel (likely closed)');\n        });\n      };\n\n      const onClose = (code: number, reason: Buffer) => {\n        // A close during an active generation is unexpected: the pool owns the\n        // socket lifecycle and does not close it between turns. If it happens\n        // before `done`, surface it so the turn retries rather than ending mid\n        // speech, and so withConnection discards the dead socket.\n        this.#logger.debug(`WebSocket closed with code ${code}: ${reason.toString()}`);\n        clearTTSChunkTimeout();\n        if (!completed && !timedOut && !streamError) {\n          streamError = new APIConnectionError({\n            message: `Cartesia WebSocket closed mid-generation (code=${code})`,\n          });\n        }\n        void eventChannel.close();\n      };\n\n      const onError = (err: Error) => {\n        this.#logger.error({ err }, 'Cartesia WebSocket error');\n        if (!completed && !timedOut && !streamError) {\n          streamError = err instanceof APIError ? err : toRetryableConnectionError(err);\n        }\n        void eventChannel.close();\n      };\n\n      // Attach listeners ONCE\n      ws.on('message', onMessage);\n      ws.on('close', onClose);\n      ws.on('error', onError);\n\n      try {\n        // Process messages from the channel\n        const reader = eventChannel.stream().getReader();\n\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (result.done) break;\n\n          const rawMsg = result.value;\n\n          // Parse message with Zod schema for type safety\n          let serverMsg: CartesiaServerMessage;\n          try {\n            const json = JSON.parse(rawMsg.toString());\n            serverMsg = cartesiaMessageSchema.parse(json);\n          } catch (parseErr) {\n            this.#logger.warn({ parseErr }, 'Failed to parse Cartesia message');\n            continue;\n          }\n\n          const segmentId = serverMsg.context_id;\n\n          // Handle error frames first. 4xx (e.g. empty-transcript on\n          // function-call turns) is non-fatal — log and fall through so an\n          // accompanying done:true still triggers the unified close path\n          // below. 5xx bubbles up so the base SynthesizeStream can retry.\n          if (isErrorMessage(serverMsg)) {\n            if (serverMsg.status_code >= 400 && serverMsg.status_code < 500) {\n              this.#logger.debug({ error: serverMsg.error }, 'Cartesia sent a non-fatal error');\n            } else {\n              this.#logger.error({ error: serverMsg.error }, 'Cartesia returned error');\n              throw new APIStatusError({\n                message: `Cartesia returned error: ${serverMsg.error}`,\n                options: { statusCode: serverMsg.status_code, retryable: true },\n              });\n            }\n          }\n\n          if (isChunkMessage(serverMsg)) {\n            const audioBuffer = Buffer.from(serverMsg.data, 'base64');\n            // Extract ArrayBuffer from Buffer for AudioByteStream compatibility\n            const audioData = audioBuffer.buffer.slice(\n              audioBuffer.byteOffset,\n              audioBuffer.byteOffset + audioBuffer.byteLength,\n            );\n            for (const frame of bstream.write(audioData)) {\n              sendLastFrame(segmentId, false);\n              lastFrame = frame;\n            }\n\n            // IMPORTANT: close WS if TTS chunk stream been stuck too long\n            // this allows unblock the current \"broken\" TTS node so that any future TTS nodes\n            // can continue to process the stream without been blocked by the stuck node\n            clearTTSChunkTimeout();\n            timeout = setTimeout(() => {\n              // cartesia chunk timeout quite often, so we make it a debug log\n              this.#logger.debug(\n                `Cartesia WebSocket TTS chunk stream timeout after ${this.#opts.chunkTimeout}ms`,\n              );\n              // The socket is stuck mid-generation, so it must not return to the\n              // pool. Poison it and unblock the reader; the post-loop check turns\n              // this into a retryable error so withConnection discards the socket.\n              timedOut = true;\n              safeCloseWebSocket(ws);\n              void eventChannel.close();\n            }, this.#opts.chunkTimeout);\n          } else if (this.#opts.wordTimestamps !== false && hasWordTimestamps(serverMsg)) {\n            const wordTimestamps = serverMsg.word_timestamps;\n            for (let i = 0; i < wordTimestamps.words.length; i++) {\n              const word = wordTimestamps.words[i];\n              const startTime = wordTimestamps.start[i];\n              const endTime = wordTimestamps.end[i];\n              if (word !== undefined && startTime !== undefined && endTime !== undefined) {\n                pendingTimedTranscripts.push(\n                  createTimedString({\n                    text: word + ' ', // Add space after word for consistency\n                    startTime,\n                    endTime,\n                  }),\n                );\n              }\n            }\n          } else if (isDoneMessage(serverMsg) || (isErrorMessage(serverMsg) && serverMsg.done)) {\n            // This ensures all sentences have been sent before closing\n            if (sentenceStreamClosed) {\n              for (const frame of bstream.flush()) {\n                sendLastFrame(segmentId, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(segmentId, true);\n              if (!this.queue.closed) {\n                this.queue.put(SynthesizeStream.END_OF_STREAM);\n              }\n\n              if (segmentId === requestId) {\n                clearTTSChunkTimeout();\n                completed = true;\n                // Leave the socket open so the pool reuses it on the next turn.\n                break; // Exit the loop\n              }\n            }\n            // If sentenceStreamClosed is false, continue receiving - more done messages will come\n          } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {\n            // flush_done is an ack with nothing to do; error frames without\n            // done:true were already logged above.\n            this.#logger.warn({ message: serverMsg }, 'Unknown Cartesia message');\n          }\n        }\n\n        if (timedOut) {\n          throw new APITimeoutError({\n            message: `Cartesia TTS chunk stream timed out after ${this.#opts.chunkTimeout}ms`,\n          });\n        }\n        if (streamError) {\n          throw streamError;\n        }\n      } catch (err) {\n        // Always propagate API errors so the base SynthesizeStream can retry\n        // and emit tts_error once retries are exhausted.\n        if (err instanceof APIError) throw err;\n        // skip log error for normal websocket close\n        if (err instanceof Error && !err.message.includes('WebSocket closed')) {\n          if (\n            err.message.includes('Queue is closed') ||\n            err.message.includes('Channel is closed')\n          ) {\n            this.#logger.warn(\n              { err },\n              'Channel closed during transcript processing (expected during disconnect)',\n            );\n          } else {\n            this.#logger.error({ err }, 'Error in recvTask from Cartesia WebSocket');\n          }\n        }\n      } finally {\n        // IMPORTANT: Remove listeners so connection can be reused\n        ws.off('message', onMessage);\n        ws.off('close', onClose);\n        ws.off('error', onError);\n        clearTTSChunkTimeout();\n      }\n    };\n\n    try {\n      // The pool hands back one live socket per call and reclaims it on success\n      // (put) or discards it on any thrown error (remove). A generation never\n      // closes the socket itself, so the next turn skips the handshake.\n      await this.#pool.withConnection(\n        async (ws) => {\n          if (ws.readyState !== WebSocket.OPEN) {\n            throw new APIConnectionError({ message: 'Cartesia pooled websocket is not open' });\n          }\n          await Promise.all([inputTask(), sentenceStreamTask(ws), recvTask(ws)]);\n        },\n        { timeout: this.connOptions.timeoutMs, signal: this.abortSignal },\n      );\n    } catch (e) {\n      if (this.abortSignal.aborted) {\n        return;\n      }\n      if (e instanceof APIError) throw e;\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nconst transientNetworkCodes = new Set([\n  'ETIMEDOUT',\n  'ECONNRESET',\n  'EAI_AGAIN',\n  'ENETUNREACH',\n  'ECONNREFUSED',\n  'EHOSTUNREACH',\n]);\n\nconst isRecord = (v: unknown): v is Record<string, unknown> => {\n  return v !== null && typeof v === 'object';\n};\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\nconst isAggregateErrorLike = (e: unknown): e is { errors: unknown[]; name?: string } => {\n  if (!isRecord(e)) return false;\n  return e.name === 'AggregateError' && Array.isArray(e.errors);\n};\n\nconst hasErrorCode = (e: unknown, code: string): boolean => {\n  if (isRecord(e) && e.code === code) return true;\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasErrorCode(inner, code));\n  }\n  return false;\n};\n\nconst hasAnyTransientCode = (e: unknown): boolean => {\n  if (isRecord(e) && typeof e.code === 'string') {\n    return transientNetworkCodes.has(e.code);\n  }\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasAnyTransientCode(inner));\n  }\n  return false;\n};\n\nconst toRetryableConnectionError = (e: unknown): APIConnectionError => {\n  const err = asError(e);\n  const isTimeout =\n    hasErrorCode(e, 'ETIMEDOUT') ||\n    (typeof err.message === 'string' && err.message.includes('ETIMEDOUT'));\n  const message = isTimeout\n    ? `Cartesia connection timed out`\n    : `Cartesia connection failed: ${err.message || 'unknown error'}`;\n  return isTimeout ? new APITimeoutError({ message }) : new APIConnectionError({ message });\n};\n\nconst waitForWsOpen = async ({\n  ws,\n  timeoutMs,\n  abortSignal,\n}: {\n  ws: WebSocket;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}) => {\n  if (abortSignal?.aborted) {\n    throw new Error('aborted');\n  }\n\n  const fut = new Future<void>();\n  let timeout: NodeJS.Timeout | undefined;\n\n  const cleanup = () => {\n    if (timeout) clearTimeout(timeout);\n    ws.off('open', onOpen);\n    ws.off('unexpected-response', onUnexpectedResponse);\n    ws.off('error', onError);\n    ws.off('close', onClose);\n    abortSignal?.removeEventListener('abort', onAbort);\n  };\n\n  const onOpen = () => fut.resolve();\n  const onError = (err: Error) => fut.reject(asError(err));\n  const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n    // Authentication headers can appear in WebSocket handshake errors.\n    const statusCode = response.statusCode ?? -1;\n    fut.reject(\n      new APIStatusError({\n        message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n        options: { statusCode },\n      }),\n    );\n  };\n  const onClose = (code: number, reason: Buffer) =>\n    fut.reject(\n      new Error(`WebSocket closed before open (code=${code}, reason=${reason.toString()})`),\n    );\n  const onAbort = () => fut.reject(new Error('aborted'));\n\n  ws.on('open', onOpen);\n  ws.on('unexpected-response', onUnexpectedResponse);\n  ws.on('error', onError);\n  ws.on('close', onClose);\n  abortSignal?.addEventListener('abort', onAbort, { once: true });\n\n  if (timeoutMs > 0) {\n    timeout = setTimeout(\n      () => fut.reject(new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' })),\n      timeoutMs,\n    );\n  }\n\n  try {\n    await fut.await;\n  } finally {\n    cleanup();\n  }\n};\n\nconst safeTerminateWebSocket = (ws: WebSocket) => {\n  // `ws` can emit an 'error' event during teardown (especially if CONNECTING).\n  // If there is no error listener at that moment, Node will treat it as unhandled and crash the process.\n  try {\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    // `terminate()` can throw if the socket was never established; `close()` is safer in CONNECTING.\n    if (ws.readyState === WebSocket.CONNECTING) {\n      ws.close();\n    } else {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\n// Graceful close used by the connection pool. A pooled socket is healthy when it\n// is retired (session age, option change, or TTS close), so a clean close frame\n// is preferable to an abrupt terminate; terminate remains the fallback for a\n// socket caught mid-handshake.\nconst safeCloseWebSocket = (ws: WebSocket) => {\n  try {\n    // `ws` can emit 'error' during teardown; without a listener Node treats it as\n    // unhandled and crashes the process.\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    if (ws.readyState === WebSocket.CONNECTING || ws.readyState === WebSocket.OPEN) {\n      ws.close();\n    } else if (ws.readyState !== WebSocket.CLOSED && ws.readyState !== WebSocket.CLOSING) {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\nconst connectCartesiaWebSocket = async ({\n  url,\n  headers,\n  timeoutMs,\n  abortSignal,\n}: {\n  url: string;\n  headers: Record<string, string>;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}): Promise<WebSocket> => {\n  const connectOnce = async (family?: number): Promise<WebSocket> => {\n    const ws = new WebSocket(url, { handshakeTimeout: timeoutMs, family, headers });\n    try {\n      await waitForWsOpen({ ws, timeoutMs, abortSignal });\n      return ws;\n    } catch (e) {\n      safeTerminateWebSocket(ws);\n      throw e;\n    }\n  };\n\n  let connectError: unknown;\n  try {\n    return await connectOnce();\n  } catch (e) {\n    connectError = e;\n    // Mitigation for Node.js dual-stack (IPv6/IPv4) connect flakiness (\"happy eyeballs\"):\n    // some environments surface `AggregateError` with nested `ETIMEDOUT` during the initial\n    // WebSocket open. In that case we do a one-off retry forcing IPv4 (`family: 4`) before\n    // letting the outer framework retry loop handle further attempts.\n    //\n    // If you still see `AggregateError`/`ETIMEDOUT`:\n    // - Increase the session TTS connect timeout (`connOptions.ttsConnOptions.timeoutMs`)\n    // - Or adjust Node's family autoselection behavior via `NODE_OPTIONS`, e.g.\n    //   `--network-family-autoselection-attempt-timeout=5000` (or disable it entirely).\n    if (!(e instanceof APIError) && (hasAnyTransientCode(e) || isAggregateErrorLike(e))) {\n      try {\n        return await connectOnce(4);\n      } catch (retryError) {\n        connectError = retryError;\n      }\n    }\n  }\n\n  if (connectError instanceof APIError) throw connectError;\n  const error = asError(connectError);\n  const isTimeout =\n    hasErrorCode(connectError, 'ETIMEDOUT') || /timed?\\s*out|timeout/i.test(error.message);\n  if (isTimeout) {\n    throw new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' });\n  }\n  // Transport errors can contain credentials in URLs.\n  throw new APIConnectionError({ message: sanitizedErrorName(error) });\n};\n\nconst toCartesiaOptions = (\n  opts: TTSOptions,\n  streaming: boolean = false,\n): { [id: string]: unknown } => {\n  const voice: { [id: string]: unknown } = {};\n  if (typeof opts.voice === 'string') {\n    voice.mode = 'id';\n    voice.id = opts.voice;\n  } else {\n    voice.mode = 'embedding';\n    voice.embedding = opts.voice;\n  }\n\n  if (opts.apiVersion === API_VERSION_WITH_EXPERIMENTAL_CONTROLS) {\n    const voiceControls: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      voiceControls.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      voiceControls.emotion = opts.emotion;\n    }\n    if (Object.keys(voiceControls).length) {\n      voice.__experimental_controls = voiceControls;\n    }\n  }\n\n  const result: { [id: string]: unknown } = {\n    model_id: opts.model,\n    voice,\n    output_format: {\n      container: 'raw',\n      encoding: opts.encoding,\n      sample_rate: opts.sampleRate,\n    },\n    language: getBaseLanguage(opts.language),\n    max_buffer_delay_ms: 0,\n  };\n\n  if (opts.pronunciationDictId) {\n    result.pronunciation_dict_id = opts.pronunciationDictId;\n  }\n\n  if (opts.apiVersion > API_VERSION_WITH_EXPERIMENTAL_CONTROLS && isSonic3(opts.model)) {\n    const generationConfig: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      generationConfig.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      generationConfig.emotion = opts.emotion[0];\n    }\n    if (opts.volume) {\n      generationConfig.volume = opts.volume;\n    }\n    if (Object.keys(generationConfig).length) {\n      result.generation_config = generationConfig;\n    }\n  }\n\n  if (streaming && opts.wordTimestamps !== false) {\n    result.add_timestamps = true;\n  }\n\n  return 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+{"version":3,"sources":["../src/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  APIError,\n  APIStatusError,\n  APITimeoutError,\n  AudioByteStream,\n  ConnectionPool,\n  Future,\n  type TimedString,\n  asError,\n  createTimedString,\n  getBaseLanguage,\n  log,\n  normalizeLanguage,\n  shortuuid,\n  stream,\n  tokenize,\n  tts,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { request } from 'node:https';\nimport { type RawData, WebSocket } from 'ws';\nimport {\n  TTSDefaultVoiceId,\n  type TTSEncoding,\n  type TTSModels,\n  type TTSVoiceEmotion,\n  type TTSVoiceSpeed,\n  isSonic3,\n} from './models.js';\nimport {\n  type CartesiaServerMessage,\n  cartesiaMessageSchema,\n  hasWordTimestamps,\n  isChunkMessage,\n  isDoneMessage,\n  isErrorMessage,\n  isFlushDoneMessage,\n} from './types.js';\n\nconst AUTHORIZATION_HEADER = 'X-API-Key';\nconst VERSION_HEADER = 'Cartesia-Version';\nconst API_VERSION = '2025-04-16';\nconst API_VERSION_WITH_EXPERIMENTAL_CONTROLS = '2024-11-13';\nconst MODEL_WITH_EXPERIMENTAL_CONTROLS = 'sonic-2-2025-03-07';\nconst NUM_CHANNELS = 1;\nconst BUFFERED_WORDS_COUNT = 8;\n// Cartesia refreshes a pooled socket after this long so a very long call cannot\n// keep one connection open indefinitely. Matches the Python plugin's 300s.\nconst MAX_SESSION_DURATION_MS = 300_000;\n\n// Lets each SynthesizeStream reach the pool owned by the TTS that created it,\n// without widening the constructor signature the base class fixes.\nconst connectionPools = new WeakMap<TTS, ConnectionPool<WebSocket>>();\n\nexport interface TTSOptions {\n  model: TTSModels | string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  voice: string | number[];\n  speed?: TTSVoiceSpeed | number;\n  emotion?: (TTSVoiceEmotion | string)[];\n  /**\n   * Volume of the speech. For sonic-3, the value is valid between 0.5 and 2.0.\n   * @see https://docs.cartesia.ai/api-reference/tts/bytes#body-generation-config-volume\n   */\n  volume?: number;\n  apiKey?: string;\n  language: string;\n  baseUrl: string;\n  apiVersion: string;\n\n  /**\n   * The timeout for the next chunk to be received from the Cartesia API.\n   */\n  chunkTimeout: number;\n\n  /**\n   * Whether to add word timestamps to the output. When enabled, the TTS will return\n   * timing information for each word in the transcript.\n   * @defaultValue true\n   */\n  wordTimestamps?: boolean;\n\n  pronunciationDictId?: string;\n}\n\nconst defaultTTSOptions: TTSOptions = {\n  model: 'sonic-3',\n  encoding: 'pcm_s16le',\n  sampleRate: 24000,\n  voice: TTSDefaultVoiceId,\n  apiKey: process.env.CARTESIA_API_KEY,\n  language: 'en',\n  baseUrl: 'https://api.cartesia.ai',\n  apiVersion: API_VERSION,\n  chunkTimeout: 5000,\n  wordTimestamps: true,\n};\n\nconst checkGenerationConfig = (opts: TTSOptions) => {\n  const logger = log();\n  if (isSonic3(opts.model)) {\n    if (opts.speed !== undefined && typeof opts.speed === 'number') {\n      if (opts.speed < 0.6 || opts.speed > 2.0) {\n        logger.warn('speed must be between 0.6 and 2.0 for sonic-3');\n      }\n    }\n    if (opts.volume !== undefined && (opts.volume < 0.5 || opts.volume > 2.0)) {\n      logger.warn('volume must be between 0.5 and 2.0 for sonic-3');\n    }\n  } else if (\n    opts.apiVersion !== API_VERSION_WITH_EXPERIMENTAL_CONTROLS ||\n    opts.model !== MODEL_WITH_EXPERIMENTAL_CONTROLS\n  ) {\n    if (opts.speed || opts.emotion) {\n      logger.warn(\n        { model: opts.model, speed: opts.speed, emotion: opts.emotion },\n        `speed and emotion controls are only supported for model '${MODEL_WITH_EXPERIMENTAL_CONTROLS}' ` +\n          `or sonic-3 models, see https://docs.cartesia.ai/developer-tools/changelog for details`,\n      );\n    }\n  }\n\n  if (opts.pronunciationDictId && !isSonic3(opts.model)) {\n    logger.warn(\n      { model: opts.model, pronunciationDictId: opts.pronunciationDictId },\n      'pronunciationDictId is only supported for sonic-3 models',\n    );\n  }\n};\n\nexport class TTS extends tts.TTS {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #closed = false;\n  label = 'cartesia.TTS';\n\n  get model(): string {\n    return this.#opts.model;\n  }\n\n  get provider(): string {\n    return 'Cartesia';\n  }\n\n  constructor(opts: Partial<TTSOptions> = {}) {\n    const resolvedOpts = {\n      ...defaultTTSOptions,\n      ...opts,\n    };\n\n    super(resolvedOpts.sampleRate || defaultTTSOptions.sampleRate, NUM_CHANNELS, {\n      streaming: true,\n      alignedTranscript: resolvedOpts.wordTimestamps ?? true,\n    });\n\n    this.#opts = resolvedOpts;\n    this.#opts.language = normalizeLanguage(this.#opts.language);\n\n    if (this.#opts.apiKey === undefined) {\n      throw new Error(\n        'Cartesia API key is required, whether as an argument or as $CARTESIA_API_KEY',\n      );\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    // One socket, reused across generations. Cartesia recommends a single\n    // preconnected WebSocket for many generations because a fresh connection\n    // repays TCP/TLS setup on every turn:\n    // https://docs.cartesia.ai/use-the-api/compare-tts-endpoints\n    this.#pool = new ConnectionPool<WebSocket>({\n      connectCb: (timeoutMs) => this.#connectWebSocket(timeoutMs),\n      closeCb: async (ws) => safeCloseWebSocket(ws),\n      maxSessionDuration: MAX_SESSION_DURATION_MS,\n      markRefreshedOnGet: true,\n    });\n    connectionPools.set(this, this.#pool);\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    // Only these three fields reach Cartesia at WebSocket-handshake time (auth\n    // header, version header, host). Everything else (model, voice, encoding,\n    // sample rate, speed, emotion, volume, language) is sent in-band on each\n    // generation, so a pooled socket serves the new value without reconnecting.\n    // Reconnect only when one of the handshake inputs actually changes.\n    const handshakeChanged =\n      (opts.apiKey !== undefined && opts.apiKey !== this.#opts.apiKey) ||\n      (opts.apiVersion !== undefined && opts.apiVersion !== this.#opts.apiVersion) ||\n      (opts.baseUrl !== undefined && opts.baseUrl !== this.#opts.baseUrl);\n\n    this.#opts = { ...this.#opts, ...opts };\n    if (opts.language !== undefined) {\n      this.#opts.language = normalizeLanguage(opts.language);\n    }\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n\n    if (handshakeChanged) {\n      this.#pool.invalidate();\n    }\n  }\n\n  synthesize(\n    text: string,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ): tts.ChunkedStream {\n    return new ChunkedStream(this, text, { ...this.#opts }, connOptions, abortSignal);\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream {\n    return new SynthesizeStream(this, { ...this.#opts }, options?.connOptions);\n  }\n\n  /**\n   * Open the pooled WebSocket ahead of the first generation so the first turn\n   * does not pay the connect. Safe to call more than once; it is a no-op when a\n   * connection already exists.\n   */\n  prewarm(): void {\n    this.#pool.prewarm();\n  }\n\n  override async close(): Promise<void> {\n    this.#closed = true;\n    await this.#pool.close();\n    await super.close();\n  }\n\n  async #connectWebSocket(timeoutMs: number): Promise<WebSocket> {\n    // Snapshot the handshake inputs. If a concurrent updateOptions() changes one\n    // of them while this connect is in flight, reconnect on the new value rather\n    // than pooling a socket built on stale credentials (mirrors the fishaudio\n    // plugin's model re-check).\n    const apiKey = this.#opts.apiKey!;\n    const apiVersion = this.#opts.apiVersion;\n    const baseUrl = this.#opts.baseUrl;\n    const url = `${baseUrl.replace(/^http/, 'ws')}/tts/websocket`;\n    const ws = await connectCartesiaWebSocket({\n      url,\n      headers: {\n        [AUTHORIZATION_HEADER]: apiKey,\n        [VERSION_HEADER]: apiVersion,\n      },\n      timeoutMs,\n    });\n    if (this.#closed) {\n      safeCloseWebSocket(ws);\n      throw new APIConnectionError({ message: 'Cartesia TTS is closed' });\n    }\n    if (\n      apiKey !== this.#opts.apiKey ||\n      apiVersion !== this.#opts.apiVersion ||\n      baseUrl !== this.#opts.baseUrl\n    ) {\n      safeCloseWebSocket(ws);\n      return await this.#connectWebSocket(timeoutMs);\n    }\n    // Drop a socket that closes (or errors) while idle in the pool. Between turns\n    // no generation listeners are attached, so without this the pool keeps a dead\n    // socket in `available` and the next turn spends a retry to discard it, or\n    // fails outright at maxRetry:0. A generation attaches its own listeners on top\n    // of these; the no-op error listener also stops an idle 'error' from crashing\n    // the process. Remove is a no-op once the socket is no longer pooled, so this\n    // is safe during an active generation and during close().\n    ws.on('error', () => {});\n    ws.on('close', () => this.#pool.remove(ws));\n    return ws;\n  }\n}\n\nexport class ChunkedStream extends tts.ChunkedStream {\n  label = 'cartesia.ChunkedStream';\n  #logger = log();\n  #opts: TTSOptions;\n  #text: string;\n\n  constructor(\n    tts: TTS,\n    text: string,\n    opts: TTSOptions,\n    connOptions?: APIConnectOptions,\n    abortSignal?: AbortSignal,\n  ) {\n    super(text, tts, connOptions, abortSignal);\n    this.#text = text;\n    this.#opts = opts;\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n    const json = toCartesiaOptions(this.#opts);\n    json.transcript = this.#text;\n\n    const baseUrl = new URL(this.#opts.baseUrl);\n    const doneFut = new Future<void>();\n\n    const req = request(\n      {\n        hostname: baseUrl.hostname,\n        port: parseInt(baseUrl.port) || (baseUrl.protocol === 'https:' ? 443 : 80),\n        path: '/tts/bytes',\n        method: 'POST',\n        headers: {\n          [AUTHORIZATION_HEADER]: this.#opts.apiKey!,\n          [VERSION_HEADER]: this.#opts.apiVersion,\n        },\n        signal: this.abortSignal,\n      },\n      (res) => {\n        res.on('data', (chunk) => {\n          for (const frame of bstream.write(chunk)) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n        });\n        res.on('close', () => {\n          for (const frame of bstream.flush()) {\n            this.queue.put({\n              requestId,\n              frame,\n              final: false,\n              segmentId: requestId,\n            });\n          }\n          this.queue.close();\n          if (!doneFut.done) doneFut.resolve();\n        });\n        res.on('error', (err) => {\n          if (err.message === 'aborted') return;\n          this.#logger.error({ err }, 'Cartesia TTS response error');\n          if (!doneFut.done) doneFut.reject(err);\n        });\n      },\n    );\n\n    req.on('error', (err) => {\n      if (err.name === 'AbortError') return;\n      this.#logger.error({ err }, 'Cartesia TTS request error');\n      if (!doneFut.done) doneFut.reject(err);\n    });\n    req.on('close', () => {\n      if (!doneFut.done) doneFut.resolve();\n    });\n    req.write(JSON.stringify(json));\n    req.end();\n\n    try {\n      await doneFut.await;\n    } catch (e) {\n      if (this.abortSignal.aborted) return;\n      if (!this.queue.closed) this.queue.close();\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nexport class SynthesizeStream extends tts.SynthesizeStream {\n  #opts: TTSOptions;\n  #pool: ConnectionPool<WebSocket>;\n  #logger = log();\n  #tokenizer = new tokenize.basic.SentenceTokenizer({\n    minSentenceLength: BUFFERED_WORDS_COUNT,\n  }).stream();\n  label = 'cartesia.SynthesizeStream';\n\n  constructor(tts: TTS, opts: TTSOptions, connOptions?: APIConnectOptions) {\n    super(tts, connOptions);\n    const pool = connectionPools.get(tts);\n    if (!pool) throw new Error('Cartesia connection pool is not initialized');\n    this.#pool = pool;\n    this.#opts = opts;\n  }\n\n  updateOptions(opts: Partial<TTSOptions>) {\n    this.#opts = { ...this.#opts, ...opts };\n\n    if (\n      this.#opts.speed ||\n      this.#opts.emotion ||\n      this.#opts.volume ||\n      this.#opts.pronunciationDictId\n    ) {\n      checkGenerationConfig(this.#opts);\n    }\n  }\n\n  protected async run() {\n    const requestId = shortuuid();\n    // Only finish the generation once both: 1) Cartesia returns done, AND 2) all sentences have been sent\n    let sentenceStreamClosed = false;\n\n    const sentenceStreamTask = async (ws: WebSocket) => {\n      const packet = toCartesiaOptions(this.#opts, true);\n      for await (const event of this.#tokenizer) {\n        const msg = {\n          ...packet,\n          context_id: requestId,\n          transcript: event.token + ' ',\n          continue: true,\n        };\n        this.markStarted();\n        ws.send(JSON.stringify(msg));\n      }\n\n      const endMsg = {\n        ...packet,\n        context_id: requestId,\n        transcript: ' ',\n        continue: false,\n      };\n      ws.send(JSON.stringify(endMsg));\n      // Mark sentence stream as closed\n      sentenceStreamClosed = true;\n    };\n\n    const inputTask = async () => {\n      for await (const data of this.input) {\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          this.#tokenizer.flush();\n          continue;\n        }\n        this.#tokenizer.pushText(data);\n      }\n      this.#tokenizer.endInput();\n      this.#tokenizer.close();\n    };\n\n    // Use event channel and set up listeners ONCE to avoid missing messages during listener re-registration\n    const recvTask = async (ws: WebSocket) => {\n      const bstream = new AudioByteStream(this.#opts.sampleRate, NUM_CHANNELS);\n\n      // Create event channel to buffer incoming messages\n      // This prevents message loss between listener re-registrations\n      const eventChannel = stream.createStreamChannel<RawData>();\n\n      let lastFrame: AudioFrame | undefined;\n      let pendingTimedTranscripts: TimedString[] = [];\n\n      const sendLastFrame = (segmentId: string, final: boolean) => {\n        if (lastFrame && !this.queue.closed) {\n          // Include timedTranscripts with the audio frame\n          this.queue.put({\n            requestId,\n            segmentId,\n            frame: lastFrame,\n            final,\n            timedTranscripts:\n              pendingTimedTranscripts.length > 0 ? pendingTimedTranscripts : undefined,\n          });\n          lastFrame = undefined;\n          pendingTimedTranscripts = [];\n        }\n      };\n\n      let timeout: NodeJS.Timeout | null = null;\n      // Set when the chunk watchdog fires: the socket is discarded, not pooled.\n      let timedOut = false;\n      // Set once this generation's `done` has been handled. Until then, a socket\n      // close or error is a mid-generation drop, not a normal end.\n      let completed = false;\n      // A socket close/error before completion. Thrown after the loop so the turn\n      // fails over (and the dead socket is discarded) instead of ending silently.\n      let streamError: Error | undefined;\n\n      const clearTTSChunkTimeout = () => {\n        if (timeout) {\n          clearTimeout(timeout);\n          timeout = null;\n        }\n      };\n\n      // Set up WebSocket listeners ONCE (not in a loop)\n      const onMessage = (data: RawData) => {\n        void eventChannel.write(data).catch((error: unknown) => {\n          this.#logger.debug({ error }, 'Failed writing Cartesia event to channel (likely closed)');\n        });\n      };\n\n      const onClose = (code: number, reason: Buffer) => {\n        // A close during an active generation is unexpected: the pool owns the\n        // socket lifecycle and does not close it between turns. If it happens\n        // before `done`, surface it so the turn retries rather than ending mid\n        // speech, and so withConnection discards the dead socket.\n        this.#logger.debug(`WebSocket closed with code ${code}: ${reason.toString()}`);\n        clearTTSChunkTimeout();\n        if (!completed && !timedOut && !streamError) {\n          streamError = new APIConnectionError({\n            message: `Cartesia WebSocket closed mid-generation (code=${code})`,\n          });\n        }\n        void eventChannel.close();\n      };\n\n      const onError = (err: Error) => {\n        this.#logger.error({ err }, 'Cartesia WebSocket error');\n        if (!completed && !timedOut && !streamError) {\n          streamError = err instanceof APIError ? err : toRetryableConnectionError(err);\n        }\n        void eventChannel.close();\n      };\n\n      // Attach listeners ONCE\n      ws.on('message', onMessage);\n      ws.on('close', onClose);\n      ws.on('error', onError);\n\n      try {\n        // Process messages from the channel\n        const reader = eventChannel.stream().getReader();\n\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (result.done) break;\n\n          const rawMsg = result.value;\n\n          // Parse message with Zod schema for type safety\n          let serverMsg: CartesiaServerMessage;\n          try {\n            const json = JSON.parse(rawMsg.toString());\n            serverMsg = cartesiaMessageSchema.parse(json);\n          } catch (parseErr) {\n            this.#logger.warn({ parseErr }, 'Failed to parse Cartesia message');\n            continue;\n          }\n\n          const segmentId = serverMsg.context_id;\n\n          // Handle error frames first. 4xx (e.g. empty-transcript on\n          // function-call turns) is non-fatal — log and fall through so an\n          // accompanying done:true still triggers the unified close path\n          // below. 5xx bubbles up so the base SynthesizeStream can retry.\n          if (isErrorMessage(serverMsg)) {\n            if (serverMsg.status_code >= 400 && serverMsg.status_code < 500) {\n              this.#logger.debug({ error: serverMsg.error }, 'Cartesia sent a non-fatal error');\n            } else {\n              this.#logger.error({ error: serverMsg.error }, 'Cartesia returned error');\n              throw new APIStatusError({\n                message: `Cartesia returned error: ${serverMsg.error}`,\n                options: { statusCode: serverMsg.status_code, retryable: true },\n              });\n            }\n          }\n\n          if (isChunkMessage(serverMsg)) {\n            const audioBuffer = Buffer.from(serverMsg.data, 'base64');\n            // Extract ArrayBuffer from Buffer for AudioByteStream compatibility\n            const audioData = audioBuffer.buffer.slice(\n              audioBuffer.byteOffset,\n              audioBuffer.byteOffset + audioBuffer.byteLength,\n            );\n            for (const frame of bstream.write(audioData)) {\n              sendLastFrame(segmentId, false);\n              lastFrame = frame;\n            }\n\n            // IMPORTANT: close WS if TTS chunk stream been stuck too long\n            // this allows unblock the current \"broken\" TTS node so that any future TTS nodes\n            // can continue to process the stream without been blocked by the stuck node\n            clearTTSChunkTimeout();\n            timeout = setTimeout(() => {\n              // cartesia chunk timeout quite often, so we make it a debug log\n              this.#logger.debug(\n                `Cartesia WebSocket TTS chunk stream timeout after ${this.#opts.chunkTimeout}ms`,\n              );\n              // The socket is stuck mid-generation, so it must not return to the\n              // pool. Poison it and unblock the reader; the post-loop check turns\n              // this into a retryable error so withConnection discards the socket.\n              timedOut = true;\n              safeCloseWebSocket(ws);\n              void eventChannel.close();\n            }, this.#opts.chunkTimeout);\n          } else if (this.#opts.wordTimestamps !== false && hasWordTimestamps(serverMsg)) {\n            const wordTimestamps = serverMsg.word_timestamps;\n            for (let i = 0; i < wordTimestamps.words.length; i++) {\n              const word = wordTimestamps.words[i];\n              const startTime = wordTimestamps.start[i];\n              const endTime = wordTimestamps.end[i];\n              if (word !== undefined && startTime !== undefined && endTime !== undefined) {\n                pendingTimedTranscripts.push(\n                  createTimedString({\n                    text: word + ' ', // Add space after word for consistency\n                    startTime,\n                    endTime,\n                  }),\n                );\n              }\n            }\n          } else if (isDoneMessage(serverMsg) || (isErrorMessage(serverMsg) && serverMsg.done)) {\n            // This ensures all sentences have been sent before closing\n            if (sentenceStreamClosed) {\n              for (const frame of bstream.flush()) {\n                sendLastFrame(segmentId, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(segmentId, true);\n              if (!this.queue.closed) {\n                this.queue.put(SynthesizeStream.END_OF_STREAM);\n              }\n\n              if (segmentId === requestId) {\n                clearTTSChunkTimeout();\n                completed = true;\n                // Leave the socket open so the pool reuses it on the next turn.\n                break; // Exit the loop\n              }\n            }\n            // If sentenceStreamClosed is false, continue receiving - more done messages will come\n          } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {\n            // flush_done is an ack with nothing to do; error frames without\n            // done:true were already logged above.\n            this.#logger.warn({ 'lk.pii.message': serverMsg }, 'Unknown Cartesia message');\n          }\n        }\n\n        if (timedOut) {\n          throw new APITimeoutError({\n            message: `Cartesia TTS chunk stream timed out after ${this.#opts.chunkTimeout}ms`,\n          });\n        }\n        if (streamError) {\n          throw streamError;\n        }\n      } catch (err) {\n        // Always propagate API errors so the base SynthesizeStream can retry\n        // and emit tts_error once retries are exhausted.\n        if (err instanceof APIError) throw err;\n        // skip log error for normal websocket close\n        if (err instanceof Error && !err.message.includes('WebSocket closed')) {\n          if (\n            err.message.includes('Queue is closed') ||\n            err.message.includes('Channel is closed')\n          ) {\n            this.#logger.warn(\n              { err },\n              'Channel closed during transcript processing (expected during disconnect)',\n            );\n          } else {\n            this.#logger.error({ err }, 'Error in recvTask from Cartesia WebSocket');\n          }\n        }\n      } finally {\n        // IMPORTANT: Remove listeners so connection can be reused\n        ws.off('message', onMessage);\n        ws.off('close', onClose);\n        ws.off('error', onError);\n        clearTTSChunkTimeout();\n      }\n    };\n\n    try {\n      // The pool hands back one live socket per call and reclaims it on success\n      // (put) or discards it on any thrown error (remove). A generation never\n      // closes the socket itself, so the next turn skips the handshake.\n      await this.#pool.withConnection(\n        async (ws) => {\n          if (ws.readyState !== WebSocket.OPEN) {\n            throw new APIConnectionError({ message: 'Cartesia pooled websocket is not open' });\n          }\n          await Promise.all([inputTask(), sentenceStreamTask(ws), recvTask(ws)]);\n        },\n        { timeout: this.connOptions.timeoutMs, signal: this.abortSignal },\n      );\n    } catch (e) {\n      if (this.abortSignal.aborted) {\n        return;\n      }\n      if (e instanceof APIError) throw e;\n      throw toRetryableConnectionError(e);\n    }\n  }\n}\n\nconst transientNetworkCodes = new Set([\n  'ETIMEDOUT',\n  'ECONNRESET',\n  'EAI_AGAIN',\n  'ENETUNREACH',\n  'ECONNREFUSED',\n  'EHOSTUNREACH',\n]);\n\nconst isRecord = (v: unknown): v is Record<string, unknown> => {\n  return v !== null && typeof v === 'object';\n};\n\nconst sanitizedErrorName = (error: Error): string => {\n  if (error instanceof SyntaxError) return 'SyntaxError';\n  if (error instanceof TypeError) return 'TypeError';\n  if (error instanceof RangeError) return 'RangeError';\n  if (error instanceof AggregateError) return 'AggregateError';\n  return 'Error';\n};\n\nconst isAggregateErrorLike = (e: unknown): e is { errors: unknown[]; name?: string } => {\n  if (!isRecord(e)) return false;\n  return e.name === 'AggregateError' && Array.isArray(e.errors);\n};\n\nconst hasErrorCode = (e: unknown, code: string): boolean => {\n  if (isRecord(e) && e.code === code) return true;\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasErrorCode(inner, code));\n  }\n  return false;\n};\n\nconst hasAnyTransientCode = (e: unknown): boolean => {\n  if (isRecord(e) && typeof e.code === 'string') {\n    return transientNetworkCodes.has(e.code);\n  }\n  if (isAggregateErrorLike(e)) {\n    return e.errors.some((inner) => hasAnyTransientCode(inner));\n  }\n  return false;\n};\n\nconst toRetryableConnectionError = (e: unknown): APIConnectionError => {\n  const err = asError(e);\n  const isTimeout =\n    hasErrorCode(e, 'ETIMEDOUT') ||\n    (typeof err.message === 'string' && err.message.includes('ETIMEDOUT'));\n  const message = isTimeout\n    ? `Cartesia connection timed out`\n    : `Cartesia connection failed: ${err.message || 'unknown error'}`;\n  return isTimeout ? new APITimeoutError({ message }) : new APIConnectionError({ message });\n};\n\nconst waitForWsOpen = async ({\n  ws,\n  timeoutMs,\n  abortSignal,\n}: {\n  ws: WebSocket;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}) => {\n  if (abortSignal?.aborted) {\n    throw new Error('aborted');\n  }\n\n  const fut = new Future<void>();\n  let timeout: NodeJS.Timeout | undefined;\n\n  const cleanup = () => {\n    if (timeout) clearTimeout(timeout);\n    ws.off('open', onOpen);\n    ws.off('unexpected-response', onUnexpectedResponse);\n    ws.off('error', onError);\n    ws.off('close', onClose);\n    abortSignal?.removeEventListener('abort', onAbort);\n  };\n\n  const onOpen = () => fut.resolve();\n  const onError = (err: Error) => fut.reject(asError(err));\n  const onUnexpectedResponse = (_request: unknown, response: { statusCode?: number }) => {\n    // Authentication headers can appear in WebSocket handshake errors.\n    const statusCode = response.statusCode ?? -1;\n    fut.reject(\n      new APIStatusError({\n        message: `Cartesia WebSocket connection rejected with status ${statusCode}`,\n        options: { statusCode },\n      }),\n    );\n  };\n  const onClose = (code: number, reason: Buffer) =>\n    fut.reject(\n      new Error(`WebSocket closed before open (code=${code}, reason=${reason.toString()})`),\n    );\n  const onAbort = () => fut.reject(new Error('aborted'));\n\n  ws.on('open', onOpen);\n  ws.on('unexpected-response', onUnexpectedResponse);\n  ws.on('error', onError);\n  ws.on('close', onClose);\n  abortSignal?.addEventListener('abort', onAbort, { once: true });\n\n  if (timeoutMs > 0) {\n    timeout = setTimeout(\n      () => fut.reject(new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' })),\n      timeoutMs,\n    );\n  }\n\n  try {\n    await fut.await;\n  } finally {\n    cleanup();\n  }\n};\n\nconst safeTerminateWebSocket = (ws: WebSocket) => {\n  // `ws` can emit an 'error' event during teardown (especially if CONNECTING).\n  // If there is no error listener at that moment, Node will treat it as unhandled and crash the process.\n  try {\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    // `terminate()` can throw if the socket was never established; `close()` is safer in CONNECTING.\n    if (ws.readyState === WebSocket.CONNECTING) {\n      ws.close();\n    } else {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\n// Graceful close used by the connection pool. A pooled socket is healthy when it\n// is retired (session age, option change, or TTS close), so a clean close frame\n// is preferable to an abrupt terminate; terminate remains the fallback for a\n// socket caught mid-handshake.\nconst safeCloseWebSocket = (ws: WebSocket) => {\n  try {\n    // `ws` can emit 'error' during teardown; without a listener Node treats it as\n    // unhandled and crashes the process.\n    ws.on('error', () => {});\n  } catch {\n    // ignore\n  }\n\n  try {\n    if (ws.readyState === WebSocket.CONNECTING || ws.readyState === WebSocket.OPEN) {\n      ws.close();\n    } else if (ws.readyState !== WebSocket.CLOSED && ws.readyState !== WebSocket.CLOSING) {\n      ws.terminate();\n    }\n  } catch {\n    // ignore\n  }\n};\n\nconst connectCartesiaWebSocket = async ({\n  url,\n  headers,\n  timeoutMs,\n  abortSignal,\n}: {\n  url: string;\n  headers: Record<string, string>;\n  timeoutMs: number;\n  abortSignal?: AbortSignal;\n}): Promise<WebSocket> => {\n  const connectOnce = async (family?: number): Promise<WebSocket> => {\n    const ws = new WebSocket(url, { handshakeTimeout: timeoutMs, family, headers });\n    try {\n      await waitForWsOpen({ ws, timeoutMs, abortSignal });\n      return ws;\n    } catch (e) {\n      safeTerminateWebSocket(ws);\n      throw e;\n    }\n  };\n\n  let connectError: unknown;\n  try {\n    return await connectOnce();\n  } catch (e) {\n    connectError = e;\n    // Mitigation for Node.js dual-stack (IPv6/IPv4) connect flakiness (\"happy eyeballs\"):\n    // some environments surface `AggregateError` with nested `ETIMEDOUT` during the initial\n    // WebSocket open. In that case we do a one-off retry forcing IPv4 (`family: 4`) before\n    // letting the outer framework retry loop handle further attempts.\n    //\n    // If you still see `AggregateError`/`ETIMEDOUT`:\n    // - Increase the session TTS connect timeout (`connOptions.ttsConnOptions.timeoutMs`)\n    // - Or adjust Node's family autoselection behavior via `NODE_OPTIONS`, e.g.\n    //   `--network-family-autoselection-attempt-timeout=5000` (or disable it entirely).\n    if (!(e instanceof APIError) && (hasAnyTransientCode(e) || isAggregateErrorLike(e))) {\n      try {\n        return await connectOnce(4);\n      } catch (retryError) {\n        connectError = retryError;\n      }\n    }\n  }\n\n  if (connectError instanceof APIError) throw connectError;\n  const error = asError(connectError);\n  const isTimeout =\n    hasErrorCode(connectError, 'ETIMEDOUT') || /timed?\\s*out|timeout/i.test(error.message);\n  if (isTimeout) {\n    throw new APITimeoutError({ message: 'Cartesia WebSocket connection timed out' });\n  }\n  // Transport errors can contain credentials in URLs.\n  throw new APIConnectionError({ message: sanitizedErrorName(error) });\n};\n\nconst toCartesiaOptions = (\n  opts: TTSOptions,\n  streaming: boolean = false,\n): { [id: string]: unknown } => {\n  const voice: { [id: string]: unknown } = {};\n  if (typeof opts.voice === 'string') {\n    voice.mode = 'id';\n    voice.id = opts.voice;\n  } else {\n    voice.mode = 'embedding';\n    voice.embedding = opts.voice;\n  }\n\n  if (opts.apiVersion === API_VERSION_WITH_EXPERIMENTAL_CONTROLS) {\n    const voiceControls: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      voiceControls.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      voiceControls.emotion = opts.emotion;\n    }\n    if (Object.keys(voiceControls).length) {\n      voice.__experimental_controls = voiceControls;\n    }\n  }\n\n  const result: { [id: string]: unknown } = {\n    model_id: opts.model,\n    voice,\n    output_format: {\n      container: 'raw',\n      encoding: opts.encoding,\n      sample_rate: opts.sampleRate,\n    },\n    language: getBaseLanguage(opts.language),\n    max_buffer_delay_ms: 0,\n  };\n\n  if (opts.pronunciationDictId) {\n    result.pronunciation_dict_id = opts.pronunciationDictId;\n  }\n\n  if (opts.apiVersion > API_VERSION_WITH_EXPERIMENTAL_CONTROLS && isSonic3(opts.model)) {\n    const generationConfig: { [id: string]: unknown } = {};\n    if (opts.speed) {\n      generationConfig.speed = opts.speed;\n    }\n    if (opts.emotion) {\n      generationConfig.emotion = opts.emotion[0];\n    }\n    if (opts.volume) {\n      generationConfig.volume = opts.volume;\n    }\n    if (Object.keys(generationConfig).length) {\n      result.generation_config = generationConfig;\n    }\n  }\n\n  if (streaming && opts.wordTimestamps !== false) {\n    result.add_timestamps = true;\n  }\n\n  return 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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@livekit/agents-plugin-cartesia",
-  "version": "1.6.4",
+  "version": "1.7.0",
   "description": "Cartesia plugin for LiveKit Node Agents",
   "main": "dist/index.js",
   "require": "dist/index.cjs",
@@ -28,23 +28,23 @@
     "README.md"
   ],
   "devDependencies": {
-    "@livekit/rtc-node": "^0.13.33",
-    "@microsoft/api-extractor": "^7.35.0",
+    "@livekit/rtc-node": "^0.13.34",
+    "@microsoft/api-extractor": "^7.58.12",
     "@types/ws": "^8.5.10",
     "tsup": "^8.3.5",
     "typescript": "^5.0.0",
-    "@livekit/agents": "1.6.4",
-    "@livekit/agents-plugin-openai": "1.6.4",
-    "@livekit/agents-plugin-silero": "1.6.4",
-    "@livekit/agents-plugins-test": "1.6.4"
+    "@livekit/agents": "1.7.0",
+    "@livekit/agents-plugin-openai": "1.7.0",
+    "@livekit/agents-plugin-silero": "1.7.0",
+    "@livekit/agents-plugins-test": "1.7.0"
   },
   "dependencies": {
     "ws": "^8.21.0"
   },
   "peerDependencies": {
-    "@livekit/rtc-node": "^0.13.33",
+    "@livekit/rtc-node": "^0.13.34",
     "zod": "^3.25.76 || ^4.1.8",
-    "@livekit/agents": "1.6.4"
+    "@livekit/agents": "1.7.0"
   },
   "scripts": {
     "build": "tsup --onSuccess \"pnpm build:types\"",

package/src/stt.ts

@@ -499,7 +499,7 @@ export class SpeechStream extends stt.SpeechStream {
         }
         try {
           if (msg.type === 'error') {
-            this.#logger.error('Cartesia sent an error', msg);
+            this.#logger.error({ 'lk.pii.message': msg }, 'Cartesia sent an error');
 
             // do not close the websocket on bad requests since that may be caused by invalid messages
             if (msg.status_code === undefined || msg.status_code >= 500) {
@@ -657,7 +657,7 @@ export class SpeechStream extends stt.SpeechStream {
       }
 
       default:
-        this.#logger.warn('received unexpected message from Cartesia STT', { data });
+        this.#logger.warn({ 'lk.pii.data': data }, 'received unexpected message from Cartesia STT');
     }
   }
 

package/src/tts.ts

@@ -635,7 +635,7 @@ export class SynthesizeStream extends tts.SynthesizeStream {
           } else if (!isFlushDoneMessage(serverMsg) && !isErrorMessage(serverMsg)) {
             // flush_done is an ack with nothing to do; error frames without
             // done:true were already logged above.
-            this.#logger.warn({ message: serverMsg }, 'Unknown Cartesia message');
+            this.#logger.warn({ 'lk.pii.message': serverMsg }, 'Unknown Cartesia message');
           }
         }