@livekit/agents-plugin-cartesia 1.6.3 → 1.6.4

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). 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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({ 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.test.cjs

@@ -85,6 +85,55 @@ async function synthesizeTurn(cartesia, text, connOptions) {
   }
 }
 (0, import_vitest.describe)("Cartesia streaming pool", () => {
+  (0, import_vitest.it)("redacts API keys from WebSocket handshake errors", async () => {
+    const secret = "cartesia-secret-api-key-do-not-log";
+    const wss = new import_ws.WebSocketServer({
+      host: "127.0.0.1",
+      port: 0,
+      verifyClient: (_info, done) => done(false, 401, "Unauthorized")
+    });
+    await (0, import_node_events.once)(wss, "listening");
+    const address = wss.address();
+    const cartesia = new import_tts.TTS({ apiKey: secret, baseUrl: `http://127.0.0.1:${address.port}` });
+    const errorEvent = (0, import_node_events.once)(cartesia, "error");
+    try {
+      const stream = cartesia.stream({
+        connOptions: { ...import_agents.DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }
+      });
+      stream.pushText("test");
+      stream.endInput();
+      const [{ error }] = await errorEvent;
+      (0, import_vitest.expect)(error).toBeInstanceOf(import_agents.APIStatusError);
+      (0, import_vitest.expect)(error.statusCode).toBe(401);
+      (0, import_vitest.expect)(error.message).not.toContain(secret);
+      (0, import_vitest.expect)(error.toString()).not.toContain(secret);
+      stream.close();
+    } finally {
+      await cartesia.close();
+      await closeWebSocketServer(wss);
+    }
+  });
+  (0, import_vitest.it)("does not retain generic WebSocket connection errors", async () => {
+    const secret = "cartesia-secret-api-key-do-not-log";
+    const cartesia = new import_tts.TTS({ apiKey: secret, baseUrl: `http://[${secret}` });
+    const errorEvent = (0, import_node_events.once)(cartesia, "error");
+    try {
+      const stream = cartesia.stream({
+        connOptions: { ...import_agents.DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }
+      });
+      stream.pushText("test");
+      stream.endInput();
+      const [{ error }] = await errorEvent;
+      (0, import_vitest.expect)(error).toBeInstanceOf(import_agents.APIConnectionError);
+      (0, import_vitest.expect)(error.message).toBe("SyntaxError");
+      (0, import_vitest.expect)(error.message).not.toContain(secret);
+      (0, import_vitest.expect)(error.toString()).not.toContain(secret);
+      (0, import_vitest.expect)(error.cause).toBeUndefined();
+      stream.close();
+    } finally {
+      await cartesia.close();
+    }
+  });
   (0, import_vitest.it)("reuses one websocket across sequential turns", async () => {
     const { wss, baseURL } = await startWebSocketServer();
     const server = serveCartesia(wss);

package/dist/tts.test.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/tts.test.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS, tts } from '@livekit/agents';\nimport { STT } from '@livekit/agents-plugin-openai';\nimport { tts as testTts } from '@livekit/agents-plugins-test';\nimport { once } from 'node:events';\nimport type { AddressInfo } from 'node:net';\nimport { describe, expect, it } from 'vitest';\nimport { type WebSocket, WebSocketServer } from 'ws';\nimport { TTS } from './tts.js';\n\nconst hasCartesiaConfig = Boolean(process.env.CARTESIA_API_KEY && process.env.OPENAI_API_KEY);\n\nif (hasCartesiaConfig) {\n  describe('Cartesia', async () => {\n    await testTts(new TTS(), new STT());\n  });\n} else {\n  describe('Cartesia', () => {\n    it.skip('requires CARTESIA_API_KEY and OPENAI_API_KEY', () => {});\n  });\n}\n\n// A single 24 kHz mono s16le frame's worth of silence, base64-encoded the way\n// Cartesia sends audio chunks.\nconst CHUNK_BASE64 = Buffer.alloc(4800).toString('base64');\n\nasync function startWebSocketServer() {\n  const wss = new WebSocketServer({ host: '127.0.0.1', port: 0 });\n  await once(wss, 'listening');\n  const address = wss.address() as AddressInfo;\n  return { wss, baseURL: `http://127.0.0.1:${address.port}` };\n}\n\nasync function closeWebSocketServer(wss: WebSocketServer): Promise<void> {\n  for (const client of wss.clients) {\n    client.close();\n  }\n  await new Promise<void>((resolve) => wss.close(() => resolve()));\n}\n\nasync function waitFor<T>(promise: Promise<T>, timeoutMs = 1000): Promise<T> {\n  let timeout: ReturnType<typeof setTimeout> | undefined;\n  try {\n    return await Promise.race([\n      promise,\n      new Promise<never>((_, reject) => {\n        timeout = setTimeout(() => reject(new Error('timed out waiting for promise')), timeoutMs);\n      }),\n    ]);\n  } finally {\n    if (timeout) clearTimeout(timeout);\n  }\n}\n\n// A minimal Cartesia TTS WebSocket server: for every generation it replies with\n// one audio chunk and a done message, echoing the caller's context_id. `onStop`\n// lets a test override the reply (e.g. to simulate a provider failure); return\n// false to suppress the normal chunk/done reply.\nfunction serveCartesia(\n  wss: WebSocketServer,\n  onStop?: (ws: WebSocket, contextId: string, connectionNumber: number) => boolean,\n): { connectionCount: () => number } {\n  let connectionCount = 0;\n  wss.on('connection', (ws) => {\n    connectionCount++;\n    const connectionNumber = connectionCount;\n    ws.on('message', (raw) => {\n      const message = JSON.parse(raw.toString()) as { context_id: string; continue?: boolean };\n      if (message.continue !== false) return; // only reply once the turn is closed\n      const contextId = message.context_id;\n      if (onStop && !onStop(ws, contextId, connectionNumber)) return;\n      ws.send(\n        JSON.stringify({\n          type: 'chunk',\n          data: CHUNK_BASE64,\n          done: false,\n          status_code: 200,\n          step_time: 0,\n          context_id: contextId,\n        }),\n      );\n      ws.send(\n        JSON.stringify({ type: 'done', done: true, status_code: 200, context_id: contextId }),\n      );\n    });\n  });\n  return { connectionCount: () => connectionCount };\n}\n\nasync function synthesizeTurn(\n  cartesia: TTS,\n  text: string,\n  connOptions?: APIConnectOptions,\n): Promise<tts.SynthesizedAudio[]> {\n  const stream = cartesia.stream({ connOptions });\n  stream.pushText(text);\n  stream.endInput();\n\n  try {\n    const events: tts.SynthesizedAudio[] = [];\n    for await (const event of stream) {\n      if (event !== tts.SynthesizeStream.END_OF_STREAM) events.push(event);\n    }\n    return events;\n  } finally {\n    stream.close();\n  }\n}\n\ndescribe('Cartesia streaming pool', () => {\n  it('reuses one websocket across sequential turns', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'second turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('prewarms and reuses the ready websocket', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n    const connected = once(wss, 'connection');\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      cartesia.prewarm();\n      await waitFor(connected);\n      expect(await synthesizeTurn(cartesia, 'prewarmed turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('discards a poisoned websocket after a failure', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // The first connection drops the turn; the second serves it normally.\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.close(1011, 'provider failure');\n        return false;\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(\n        await synthesizeTurn(cartesia, 'failing turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('fails over when the socket drops mid-generation instead of ending silently', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // Connection 1 emits one audio chunk, then drops WITHOUT a done message,\n    // i.e. mid-speech. Connection 2 serves the recovery turn normally.\n    const server = serveCartesia(wss, (ws, contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.send(\n          JSON.stringify({\n            type: 'chunk',\n            data: CHUNK_BASE64,\n            done: false,\n            status_code: 200,\n            step_time: 0,\n            context_id: contextId,\n          }),\n        );\n        setTimeout(() => ws.close(1011, 'mid-speech drop'), 5);\n        return false; // suppress the normal chunk/done reply\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      // The dropped turn does not complete successfully (it fails over rather\n      // than silently ending); at maxRetry: 0 that surfaces as no audio.\n      expect(\n        await synthesizeTurn(cartesia, 'dropping turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      // The dead socket is discarded, so the next turn opens a fresh one.\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('replaces a websocket that closed while idle', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    let firstConnectionClosed: (() => void) | undefined;\n    const firstClosed = new Promise<void>((resolve) => {\n      firstConnectionClosed = resolve;\n    });\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.on('close', () => firstConnectionClosed?.());\n        // Serve the turn, then drop the idle socket so the next turn reconnects.\n        setTimeout(() => ws.close(), 10);\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      await waitFor(firstClosed);\n      // Let the client observe the close so the idle handler removes the socket\n      // before the next checkout, making the maxRetry: 0 assertion deterministic.\n      await new Promise((resolve) => setTimeout(resolve, 100));\n      // maxRetry: 0 proves the idle-closed socket was dropped from the pool, not\n      // handed back to burn the turn's only attempt.\n      expect(\n        await waitFor(\n          synthesizeTurn(cartesia, 'second turn.', { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }),\n        ),\n      ).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('closes the pooled websocket when the TTS closes', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      await synthesizeTurn(cartesia, 'closing turn.');\n      await cartesia.close();\n      // close() drains the pooled socket; give the close frame a beat to land.\n      await waitFor(\n        (async () => {\n          while (wss.clients.size > 0) await new Promise((r) => setTimeout(r, 5));\n        })(),\n      );\n      expect(wss.clients.size).toBe(0);\n    } finally {\n      await closeWebSocketServer(wss);\n    }\n  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finally {\n    if (timeout) clearTimeout(timeout);\n  }\n}\n\n// A minimal Cartesia TTS WebSocket server: for every generation it replies with\n// one audio chunk and a done message, echoing the caller's context_id. `onStop`\n// lets a test override the reply (e.g. to simulate a provider failure); return\n// false to suppress the normal chunk/done reply.\nfunction serveCartesia(\n  wss: WebSocketServer,\n  onStop?: (ws: WebSocket, contextId: string, connectionNumber: number) => boolean,\n): { connectionCount: () => number } {\n  let connectionCount = 0;\n  wss.on('connection', (ws) => {\n    connectionCount++;\n    const connectionNumber = connectionCount;\n    ws.on('message', (raw) => {\n      const message = JSON.parse(raw.toString()) as { context_id: string; continue?: boolean };\n      if (message.continue !== false) return; // only reply once the turn is closed\n      const contextId = message.context_id;\n      if (onStop && !onStop(ws, contextId, connectionNumber)) return;\n      ws.send(\n        JSON.stringify({\n          type: 'chunk',\n          data: CHUNK_BASE64,\n          done: false,\n          status_code: 200,\n          step_time: 0,\n          context_id: contextId,\n        }),\n      );\n      ws.send(\n        JSON.stringify({ type: 'done', done: true, status_code: 200, context_id: contextId }),\n      );\n    });\n  });\n  return { connectionCount: () => connectionCount };\n}\n\nasync function synthesizeTurn(\n  cartesia: TTS,\n  text: string,\n  connOptions?: APIConnectOptions,\n): Promise<tts.SynthesizedAudio[]> {\n  const stream = cartesia.stream({ connOptions });\n  stream.pushText(text);\n  stream.endInput();\n\n  try {\n    const events: tts.SynthesizedAudio[] = [];\n    for await (const event of stream) {\n      if (event !== tts.SynthesizeStream.END_OF_STREAM) events.push(event);\n    }\n    return events;\n  } finally {\n    stream.close();\n  }\n}\n\ndescribe('Cartesia streaming pool', () => {\n  it('redacts API keys from WebSocket handshake errors', async () => {\n    const secret = 'cartesia-secret-api-key-do-not-log';\n    const wss = new WebSocketServer({\n      host: '127.0.0.1',\n      port: 0,\n      verifyClient: (_info, done) => done(false, 401, 'Unauthorized'),\n    });\n    await once(wss, 'listening');\n    const address = wss.address() as AddressInfo;\n    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://127.0.0.1:${address.port}` });\n    const errorEvent = once(cartesia, 'error') as Promise<Parameters<tts.TTSCallbacks['error']>>;\n\n    try {\n      const stream = cartesia.stream({\n        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 },\n      });\n      stream.pushText('test');\n      stream.endInput();\n\n      const [{ error }] = await errorEvent;\n      expect(error).toBeInstanceOf(APIStatusError);\n      expect((error as APIStatusError).statusCode).toBe(401);\n      expect(error.message).not.toContain(secret);\n      expect(error.toString()).not.toContain(secret);\n      stream.close();\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('does not retain generic WebSocket connection errors', async () => {\n    const secret = 'cartesia-secret-api-key-do-not-log';\n    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://[${secret}` });\n    const errorEvent = once(cartesia, 'error') as Promise<Parameters<tts.TTSCallbacks['error']>>;\n\n    try {\n      const stream = cartesia.stream({\n        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 },\n      });\n      stream.pushText('test');\n      stream.endInput();\n\n      const [{ error }] = await errorEvent;\n      expect(error).toBeInstanceOf(APIConnectionError);\n      expect(error.message).toBe('SyntaxError');\n      expect(error.message).not.toContain(secret);\n      expect(error.toString()).not.toContain(secret);\n      expect((error as Error & { cause?: unknown }).cause).toBeUndefined();\n      stream.close();\n    } finally {\n      await cartesia.close();\n    }\n  });\n\n  it('reuses one websocket across sequential turns', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'second turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('prewarms and reuses the ready websocket', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n    const connected = once(wss, 'connection');\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      cartesia.prewarm();\n      await waitFor(connected);\n      expect(await synthesizeTurn(cartesia, 'prewarmed turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('discards a poisoned websocket after a failure', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // The first connection drops the turn; the second serves it normally.\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.close(1011, 'provider failure');\n        return false;\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(\n        await synthesizeTurn(cartesia, 'failing turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('fails over when the socket drops mid-generation instead of ending silently', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // Connection 1 emits one audio chunk, then drops WITHOUT a done message,\n    // i.e. mid-speech. Connection 2 serves the recovery turn normally.\n    const server = serveCartesia(wss, (ws, contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.send(\n          JSON.stringify({\n            type: 'chunk',\n            data: CHUNK_BASE64,\n            done: false,\n            status_code: 200,\n            step_time: 0,\n            context_id: contextId,\n          }),\n        );\n        setTimeout(() => ws.close(1011, 'mid-speech drop'), 5);\n        return false; // suppress the normal chunk/done reply\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      // The dropped turn does not complete successfully (it fails over rather\n      // than silently ending); at maxRetry: 0 that surfaces as no audio.\n      expect(\n        await synthesizeTurn(cartesia, 'dropping turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      // The dead socket is discarded, so the next turn opens a fresh one.\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('replaces a websocket that closed while idle', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    let firstConnectionClosed: (() => void) | undefined;\n    const firstClosed = new Promise<void>((resolve) => {\n      firstConnectionClosed = resolve;\n    });\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.on('close', () => firstConnectionClosed?.());\n        // Serve the turn, then drop the idle socket so the next turn reconnects.\n        setTimeout(() => ws.close(), 10);\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      await waitFor(firstClosed);\n      // Let the client observe the close so the idle handler removes the socket\n      // before the next checkout, making the maxRetry: 0 assertion deterministic.\n      await new Promise((resolve) => setTimeout(resolve, 100));\n      // maxRetry: 0 proves the idle-closed socket was dropped from the pool, not\n      // handed back to burn the turn's only attempt.\n      expect(\n        await waitFor(\n          synthesizeTurn(cartesia, 'second turn.', { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }),\n        ),\n      ).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('closes the pooled websocket when the TTS closes', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      await synthesizeTurn(cartesia, 'closing turn.');\n      await cartesia.close();\n      // close() drains the pooled socket; give the close frame a beat to land.\n      await waitFor(\n        (async () => {\n          while (wss.clients.size > 0) await new Promise((r) => setTimeout(r, 5));\n        })(),\n      );\n      expect(wss.clients.size).toBe(0);\n    } finally {\n      await closeWebSocketServer(wss);\n    }\n  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package/dist/tts.test.js

@@ -1,4 +1,9 @@
-import { DEFAULT_API_CONNECT_OPTIONS, tts } from "@livekit/agents";
+import {
+  APIConnectionError,
+  APIStatusError,
+  DEFAULT_API_CONNECT_OPTIONS,
+  tts
+} from "@livekit/agents";
 import { STT } from "@livekit/agents-plugin-openai";
 import { tts as testTts } from "@livekit/agents-plugins-test";
 import { once } from "node:events";
@@ -84,6 +89,55 @@ async function synthesizeTurn(cartesia, text, connOptions) {
   }
 }
 describe("Cartesia streaming pool", () => {
+  it("redacts API keys from WebSocket handshake errors", async () => {
+    const secret = "cartesia-secret-api-key-do-not-log";
+    const wss = new WebSocketServer({
+      host: "127.0.0.1",
+      port: 0,
+      verifyClient: (_info, done) => done(false, 401, "Unauthorized")
+    });
+    await once(wss, "listening");
+    const address = wss.address();
+    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://127.0.0.1:${address.port}` });
+    const errorEvent = once(cartesia, "error");
+    try {
+      const stream = cartesia.stream({
+        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }
+      });
+      stream.pushText("test");
+      stream.endInput();
+      const [{ error }] = await errorEvent;
+      expect(error).toBeInstanceOf(APIStatusError);
+      expect(error.statusCode).toBe(401);
+      expect(error.message).not.toContain(secret);
+      expect(error.toString()).not.toContain(secret);
+      stream.close();
+    } finally {
+      await cartesia.close();
+      await closeWebSocketServer(wss);
+    }
+  });
+  it("does not retain generic WebSocket connection errors", async () => {
+    const secret = "cartesia-secret-api-key-do-not-log";
+    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://[${secret}` });
+    const errorEvent = once(cartesia, "error");
+    try {
+      const stream = cartesia.stream({
+        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }
+      });
+      stream.pushText("test");
+      stream.endInput();
+      const [{ error }] = await errorEvent;
+      expect(error).toBeInstanceOf(APIConnectionError);
+      expect(error.message).toBe("SyntaxError");
+      expect(error.message).not.toContain(secret);
+      expect(error.toString()).not.toContain(secret);
+      expect(error.cause).toBeUndefined();
+      stream.close();
+    } finally {
+      await cartesia.close();
+    }
+  });
   it("reuses one websocket across sequential turns", async () => {
     const { wss, baseURL } = await startWebSocketServer();
     const server = serveCartesia(wss);

package/dist/tts.test.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/tts.test.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS, tts } from '@livekit/agents';\nimport { STT } from '@livekit/agents-plugin-openai';\nimport { tts as testTts } from '@livekit/agents-plugins-test';\nimport { once } from 'node:events';\nimport type { AddressInfo } from 'node:net';\nimport { describe, expect, it } from 'vitest';\nimport { type WebSocket, WebSocketServer } from 'ws';\nimport { TTS } from './tts.js';\n\nconst hasCartesiaConfig = Boolean(process.env.CARTESIA_API_KEY && process.env.OPENAI_API_KEY);\n\nif (hasCartesiaConfig) {\n  describe('Cartesia', async () => {\n    await testTts(new TTS(), new STT());\n  });\n} else {\n  describe('Cartesia', () => {\n    it.skip('requires CARTESIA_API_KEY and OPENAI_API_KEY', () => {});\n  });\n}\n\n// A single 24 kHz mono s16le frame's worth of silence, base64-encoded the way\n// Cartesia sends audio chunks.\nconst CHUNK_BASE64 = Buffer.alloc(4800).toString('base64');\n\nasync function startWebSocketServer() {\n  const wss = new WebSocketServer({ host: '127.0.0.1', port: 0 });\n  await once(wss, 'listening');\n  const address = wss.address() as AddressInfo;\n  return { wss, baseURL: `http://127.0.0.1:${address.port}` };\n}\n\nasync function closeWebSocketServer(wss: WebSocketServer): Promise<void> {\n  for (const client of wss.clients) {\n    client.close();\n  }\n  await new Promise<void>((resolve) => wss.close(() => resolve()));\n}\n\nasync function waitFor<T>(promise: Promise<T>, timeoutMs = 1000): Promise<T> {\n  let timeout: ReturnType<typeof setTimeout> | undefined;\n  try {\n    return await Promise.race([\n      promise,\n      new Promise<never>((_, reject) => {\n        timeout = setTimeout(() => reject(new Error('timed out waiting for promise')), timeoutMs);\n      }),\n    ]);\n  } finally {\n    if (timeout) clearTimeout(timeout);\n  }\n}\n\n// A minimal Cartesia TTS WebSocket server: for every generation it replies with\n// one audio chunk and a done message, echoing the caller's context_id. `onStop`\n// lets a test override the reply (e.g. to simulate a provider failure); return\n// false to suppress the normal chunk/done reply.\nfunction serveCartesia(\n  wss: WebSocketServer,\n  onStop?: (ws: WebSocket, contextId: string, connectionNumber: number) => boolean,\n): { connectionCount: () => number } {\n  let connectionCount = 0;\n  wss.on('connection', (ws) => {\n    connectionCount++;\n    const connectionNumber = connectionCount;\n    ws.on('message', (raw) => {\n      const message = JSON.parse(raw.toString()) as { context_id: string; continue?: boolean };\n      if (message.continue !== false) return; // only reply once the turn is closed\n      const contextId = message.context_id;\n      if (onStop && !onStop(ws, contextId, connectionNumber)) return;\n      ws.send(\n        JSON.stringify({\n          type: 'chunk',\n          data: CHUNK_BASE64,\n          done: false,\n          status_code: 200,\n          step_time: 0,\n          context_id: contextId,\n        }),\n      );\n      ws.send(\n        JSON.stringify({ type: 'done', done: true, status_code: 200, context_id: contextId }),\n      );\n    });\n  });\n  return { connectionCount: () => connectionCount };\n}\n\nasync function synthesizeTurn(\n  cartesia: TTS,\n  text: string,\n  connOptions?: APIConnectOptions,\n): Promise<tts.SynthesizedAudio[]> {\n  const stream = cartesia.stream({ connOptions });\n  stream.pushText(text);\n  stream.endInput();\n\n  try {\n    const events: tts.SynthesizedAudio[] = [];\n    for await (const event of stream) {\n      if (event !== tts.SynthesizeStream.END_OF_STREAM) events.push(event);\n    }\n    return events;\n  } finally {\n    stream.close();\n  }\n}\n\ndescribe('Cartesia streaming pool', () => {\n  it('reuses one websocket across sequential turns', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'second turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('prewarms and reuses the ready websocket', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n    const connected = once(wss, 'connection');\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      cartesia.prewarm();\n      await waitFor(connected);\n      expect(await synthesizeTurn(cartesia, 'prewarmed turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('discards a poisoned websocket after a failure', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // The first connection drops the turn; the second serves it normally.\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.close(1011, 'provider failure');\n        return false;\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(\n        await synthesizeTurn(cartesia, 'failing turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('fails over when the socket drops mid-generation instead of ending silently', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // Connection 1 emits one audio chunk, then drops WITHOUT a done message,\n    // i.e. mid-speech. Connection 2 serves the recovery turn normally.\n    const server = serveCartesia(wss, (ws, contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.send(\n          JSON.stringify({\n            type: 'chunk',\n            data: CHUNK_BASE64,\n            done: false,\n            status_code: 200,\n            step_time: 0,\n            context_id: contextId,\n          }),\n        );\n        setTimeout(() => ws.close(1011, 'mid-speech drop'), 5);\n        return false; // suppress the normal chunk/done reply\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      // The dropped turn does not complete successfully (it fails over rather\n      // than silently ending); at maxRetry: 0 that surfaces as no audio.\n      expect(\n        await synthesizeTurn(cartesia, 'dropping turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      // The dead socket is discarded, so the next turn opens a fresh one.\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('replaces a websocket that closed while idle', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    let firstConnectionClosed: (() => void) | undefined;\n    const firstClosed = new Promise<void>((resolve) => {\n      firstConnectionClosed = resolve;\n    });\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.on('close', () => firstConnectionClosed?.());\n        // Serve the turn, then drop the idle socket so the next turn reconnects.\n        setTimeout(() => ws.close(), 10);\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      await waitFor(firstClosed);\n      // Let the client observe the close so the idle handler removes the socket\n      // before the next checkout, making the maxRetry: 0 assertion deterministic.\n      await new Promise((resolve) => setTimeout(resolve, 100));\n      // maxRetry: 0 proves the idle-closed socket was dropped from the pool, not\n      // handed back to burn the turn's only attempt.\n      expect(\n        await waitFor(\n          synthesizeTurn(cartesia, 'second turn.', { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }),\n        ),\n      ).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('closes the pooled websocket when the TTS closes', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      await synthesizeTurn(cartesia, 'closing turn.');\n      await cartesia.close();\n      // close() drains the pooled socket; give the close frame a beat to land.\n      await waitFor(\n        (async () => {\n          while (wss.clients.size > 0) await new Promise((r) => setTimeout(r, 5));\n        })(),\n      );\n      expect(wss.clients.size).toBe(0);\n    } finally {\n      await closeWebSocketServer(wss);\n    }\n  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+{"version":3,"sources":["../src/tts.test.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport {\n  type APIConnectOptions,\n  APIConnectionError,\n  APIStatusError,\n  DEFAULT_API_CONNECT_OPTIONS,\n  tts,\n} from '@livekit/agents';\nimport { STT } from '@livekit/agents-plugin-openai';\nimport { tts as testTts } from '@livekit/agents-plugins-test';\nimport { once } from 'node:events';\nimport type { AddressInfo } from 'node:net';\nimport { describe, expect, it } from 'vitest';\nimport { type WebSocket, WebSocketServer } from 'ws';\nimport { TTS } from './tts.js';\n\nconst hasCartesiaConfig = Boolean(process.env.CARTESIA_API_KEY && process.env.OPENAI_API_KEY);\n\nif (hasCartesiaConfig) {\n  describe('Cartesia', async () => {\n    await testTts(new TTS(), new STT());\n  });\n} else {\n  describe('Cartesia', () => {\n    it.skip('requires CARTESIA_API_KEY and OPENAI_API_KEY', () => {});\n  });\n}\n\n// A single 24 kHz mono s16le frame's worth of silence, base64-encoded the way\n// Cartesia sends audio chunks.\nconst CHUNK_BASE64 = Buffer.alloc(4800).toString('base64');\n\nasync function startWebSocketServer() {\n  const wss = new WebSocketServer({ host: '127.0.0.1', port: 0 });\n  await once(wss, 'listening');\n  const address = wss.address() as AddressInfo;\n  return { wss, baseURL: `http://127.0.0.1:${address.port}` };\n}\n\nasync function closeWebSocketServer(wss: WebSocketServer): Promise<void> {\n  for (const client of wss.clients) {\n    client.close();\n  }\n  await new Promise<void>((resolve) => wss.close(() => resolve()));\n}\n\nasync function waitFor<T>(promise: Promise<T>, timeoutMs = 1000): Promise<T> {\n  let timeout: ReturnType<typeof setTimeout> | undefined;\n  try {\n    return await Promise.race([\n      promise,\n      new Promise<never>((_, reject) => {\n        timeout = setTimeout(() => reject(new Error('timed out waiting for promise')), timeoutMs);\n      }),\n    ]);\n  } finally {\n    if (timeout) clearTimeout(timeout);\n  }\n}\n\n// A minimal Cartesia TTS WebSocket server: for every generation it replies with\n// one audio chunk and a done message, echoing the caller's context_id. `onStop`\n// lets a test override the reply (e.g. to simulate a provider failure); return\n// false to suppress the normal chunk/done reply.\nfunction serveCartesia(\n  wss: WebSocketServer,\n  onStop?: (ws: WebSocket, contextId: string, connectionNumber: number) => boolean,\n): { connectionCount: () => number } {\n  let connectionCount = 0;\n  wss.on('connection', (ws) => {\n    connectionCount++;\n    const connectionNumber = connectionCount;\n    ws.on('message', (raw) => {\n      const message = JSON.parse(raw.toString()) as { context_id: string; continue?: boolean };\n      if (message.continue !== false) return; // only reply once the turn is closed\n      const contextId = message.context_id;\n      if (onStop && !onStop(ws, contextId, connectionNumber)) return;\n      ws.send(\n        JSON.stringify({\n          type: 'chunk',\n          data: CHUNK_BASE64,\n          done: false,\n          status_code: 200,\n          step_time: 0,\n          context_id: contextId,\n        }),\n      );\n      ws.send(\n        JSON.stringify({ type: 'done', done: true, status_code: 200, context_id: contextId }),\n      );\n    });\n  });\n  return { connectionCount: () => connectionCount };\n}\n\nasync function synthesizeTurn(\n  cartesia: TTS,\n  text: string,\n  connOptions?: APIConnectOptions,\n): Promise<tts.SynthesizedAudio[]> {\n  const stream = cartesia.stream({ connOptions });\n  stream.pushText(text);\n  stream.endInput();\n\n  try {\n    const events: tts.SynthesizedAudio[] = [];\n    for await (const event of stream) {\n      if (event !== tts.SynthesizeStream.END_OF_STREAM) events.push(event);\n    }\n    return events;\n  } finally {\n    stream.close();\n  }\n}\n\ndescribe('Cartesia streaming pool', () => {\n  it('redacts API keys from WebSocket handshake errors', async () => {\n    const secret = 'cartesia-secret-api-key-do-not-log';\n    const wss = new WebSocketServer({\n      host: '127.0.0.1',\n      port: 0,\n      verifyClient: (_info, done) => done(false, 401, 'Unauthorized'),\n    });\n    await once(wss, 'listening');\n    const address = wss.address() as AddressInfo;\n    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://127.0.0.1:${address.port}` });\n    const errorEvent = once(cartesia, 'error') as Promise<Parameters<tts.TTSCallbacks['error']>>;\n\n    try {\n      const stream = cartesia.stream({\n        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 },\n      });\n      stream.pushText('test');\n      stream.endInput();\n\n      const [{ error }] = await errorEvent;\n      expect(error).toBeInstanceOf(APIStatusError);\n      expect((error as APIStatusError).statusCode).toBe(401);\n      expect(error.message).not.toContain(secret);\n      expect(error.toString()).not.toContain(secret);\n      stream.close();\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('does not retain generic WebSocket connection errors', async () => {\n    const secret = 'cartesia-secret-api-key-do-not-log';\n    const cartesia = new TTS({ apiKey: secret, baseUrl: `http://[${secret}` });\n    const errorEvent = once(cartesia, 'error') as Promise<Parameters<tts.TTSCallbacks['error']>>;\n\n    try {\n      const stream = cartesia.stream({\n        connOptions: { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 },\n      });\n      stream.pushText('test');\n      stream.endInput();\n\n      const [{ error }] = await errorEvent;\n      expect(error).toBeInstanceOf(APIConnectionError);\n      expect(error.message).toBe('SyntaxError');\n      expect(error.message).not.toContain(secret);\n      expect(error.toString()).not.toContain(secret);\n      expect((error as Error & { cause?: unknown }).cause).toBeUndefined();\n      stream.close();\n    } finally {\n      await cartesia.close();\n    }\n  });\n\n  it('reuses one websocket across sequential turns', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'second turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('prewarms and reuses the ready websocket', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    const server = serveCartesia(wss);\n    const connected = once(wss, 'connection');\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      cartesia.prewarm();\n      await waitFor(connected);\n      expect(await synthesizeTurn(cartesia, 'prewarmed turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(1);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('discards a poisoned websocket after a failure', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // The first connection drops the turn; the second serves it normally.\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.close(1011, 'provider failure');\n        return false;\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(\n        await synthesizeTurn(cartesia, 'failing turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('fails over when the socket drops mid-generation instead of ending silently', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    // Connection 1 emits one audio chunk, then drops WITHOUT a done message,\n    // i.e. mid-speech. Connection 2 serves the recovery turn normally.\n    const server = serveCartesia(wss, (ws, contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.send(\n          JSON.stringify({\n            type: 'chunk',\n            data: CHUNK_BASE64,\n            done: false,\n            status_code: 200,\n            step_time: 0,\n            context_id: contextId,\n          }),\n        );\n        setTimeout(() => ws.close(1011, 'mid-speech drop'), 5);\n        return false; // suppress the normal chunk/done reply\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      // The dropped turn does not complete successfully (it fails over rather\n      // than silently ending); at maxRetry: 0 that surfaces as no audio.\n      expect(\n        await synthesizeTurn(cartesia, 'dropping turn.', {\n          ...DEFAULT_API_CONNECT_OPTIONS,\n          maxRetry: 0,\n        }),\n      ).toHaveLength(0);\n      // The dead socket is discarded, so the next turn opens a fresh one.\n      expect(await synthesizeTurn(cartesia, 'recovery turn.')).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('replaces a websocket that closed while idle', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    let firstConnectionClosed: (() => void) | undefined;\n    const firstClosed = new Promise<void>((resolve) => {\n      firstConnectionClosed = resolve;\n    });\n    const server = serveCartesia(wss, (ws, _contextId, connectionNumber) => {\n      if (connectionNumber === 1) {\n        ws.on('close', () => firstConnectionClosed?.());\n        // Serve the turn, then drop the idle socket so the next turn reconnects.\n        setTimeout(() => ws.close(), 10);\n      }\n      return true;\n    });\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      expect(await synthesizeTurn(cartesia, 'first turn.')).not.toHaveLength(0);\n      await waitFor(firstClosed);\n      // Let the client observe the close so the idle handler removes the socket\n      // before the next checkout, making the maxRetry: 0 assertion deterministic.\n      await new Promise((resolve) => setTimeout(resolve, 100));\n      // maxRetry: 0 proves the idle-closed socket was dropped from the pool, not\n      // handed back to burn the turn's only attempt.\n      expect(\n        await waitFor(\n          synthesizeTurn(cartesia, 'second turn.', { ...DEFAULT_API_CONNECT_OPTIONS, maxRetry: 0 }),\n        ),\n      ).not.toHaveLength(0);\n      expect(server.connectionCount()).toBe(2);\n    } finally {\n      await cartesia.close();\n      await closeWebSocketServer(wss);\n    }\n  });\n\n  it('closes the pooled websocket when the TTS closes', async () => {\n    const { wss, baseURL } = await startWebSocketServer();\n    serveCartesia(wss);\n\n    const cartesia = new TTS({ apiKey: 'test-key', baseUrl: baseURL });\n    try {\n      await synthesizeTurn(cartesia, 'closing turn.');\n      await cartesia.close();\n      // close() drains the pooled socket; give the close frame a beat to land.\n      await waitFor(\n        (async () => {\n          while (wss.clients.size > 0) await new Promise((r) => setTimeout(r, 5));\n        })(),\n      );\n      expect(wss.clients.size).toBe(0);\n    } finally {\n      await closeWebSocketServer(wss);\n    }\n  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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@livekit/agents-plugin-cartesia",
-  "version": "1.6.3",
+  "version": "1.6.4",
   "description": "Cartesia plugin for LiveKit Node Agents",
   "main": "dist/index.js",
   "require": "dist/index.cjs",
@@ -33,10 +33,10 @@
     "@types/ws": "^8.5.10",
     "tsup": "^8.3.5",
     "typescript": "^5.0.0",
-    "@livekit/agents": "1.6.3",
-    "@livekit/agents-plugin-openai": "1.6.3",
-    "@livekit/agents-plugin-silero": "1.6.3",
-    "@livekit/agents-plugins-test": "1.6.3"
+    "@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"
   },
   "dependencies": {
     "ws": "^8.21.0"
@@ -44,7 +44,7 @@
   "peerDependencies": {
     "@livekit/rtc-node": "^0.13.33",
     "zod": "^3.25.76 || ^4.1.8",
-    "@livekit/agents": "1.6.3"
+    "@livekit/agents": "1.6.4"
   },
   "scripts": {
     "build": "tsup --onSuccess \"pnpm build:types\"",