@livekit/agents 1.0.20 → 1.0.22

package/dist/inference/tts.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/inference/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport { basic as tokenizeBasic } from '../tokenize/index.js';\nimport type { ChunkedStream } from '../tts/index.js';\nimport { SynthesizeStream as BaseSynthesizeStream, TTS as BaseTTS } from '../tts/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { shortuuid } from '../utils.js';\nimport {\n  type TtsClientEvent,\n  type TtsServerEvent,\n  type TtsSessionCreateEvent,\n  ttsClientEventSchema,\n  ttsServerEventSchema,\n} from './api_protos.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type CartesiaModels =\n  | 'cartesia'\n  | 'cartesia/sonic'\n  | 'cartesia/sonic-2'\n  | 'cartesia/sonic-turbo';\n\nexport type ElevenlabsModels =\n  | 'elevenlabs'\n  | 'elevenlabs/eleven_flash_v2'\n  | 'elevenlabs/eleven_flash_v2_5'\n  | 'elevenlabs/eleven_turbo_v2'\n  | 'elevenlabs/eleven_turbo_v2_5'\n  | 'elevenlabs/eleven_multilingual_v2';\n\nexport type RimeModels = 'rime' | 'rime/mist' | 'rime/mistv2' | 'rime/arcana';\n\nexport type InworldModels = 'inworld' | 'inworld/inworld-tts-1';\n\nexport interface CartesiaOptions {\n  duration?: number; // max duration of audio in seconds\n  speed?: 'slow' | 'normal' | 'fast'; // default: not specified\n}\n\nexport interface ElevenlabsOptions {\n  inactivity_timeout?: number; // default: 60\n  apply_text_normalization?: 'auto' | 'off' | 'on'; // default: \"auto\"\n}\n\nexport interface RimeOptions {}\n\nexport interface InworldOptions {}\n\ntype _TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels;\n\nexport type TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels | AnyString;\n\nexport type ModelWithVoice = `${_TTSModels}:${string}` | TTSModels;\n\nexport type TTSOptions<TModel extends TTSModels> = TModel extends CartesiaModels\n  ? CartesiaOptions\n  : TModel extends ElevenlabsModels\n    ? ElevenlabsOptions\n    : TModel extends RimeOptions\n      ? RimeOptions\n      : TModel extends InworldOptions\n        ? InworldOptions\n        : Record<string, unknown>;\n\ntype TTSEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: TTSEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'https://agent-gateway.livekit.cloud/v1';\nconst NUM_CHANNELS = 1;\nconst DEFAULT_LANGUAGE = 'en';\n\nexport interface InferenceTTSOptions<TModel extends TTSModels> {\n  model?: TModel;\n  voice?: string;\n  language?: string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: TTSOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference TTS\n */\nexport class TTS<TModel extends TTSModels> extends BaseTTS {\n  private opts: InferenceTTSOptions<TModel>;\n  private streams: Set<SynthesizeStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts: {\n    model: TModel;\n    voice?: string;\n    language?: string;\n    baseURL?: string;\n    encoding?: TTSEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: TTSOptions<TModel>;\n  }) {\n    const sampleRate = opts?.sampleRate ?? DEFAULT_SAMPLE_RATE;\n    super(sampleRate, 1, { streaming: true });\n\n    const {\n      model,\n      voice,\n      language = DEFAULT_LANGUAGE,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as TTSOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    // read voice id from the model if provided: \"provider/model:voice_id\"\n    let nextModel = model;\n    let nextVoice = voice;\n    if (typeof nextModel === 'string') {\n      const idx = nextModel.lastIndexOf(':');\n      if (idx !== -1) {\n        const voiceFromModel = nextModel.slice(idx + 1);\n        if (nextVoice && nextVoice !== voiceFromModel) {\n          this.#logger.warn(\n            '`voice` is provided via both argument and model, using the one from the argument',\n            { voice: nextVoice, model: nextModel },\n          );\n        } else {\n          nextVoice = voiceFromModel;\n        }\n        nextModel = nextModel.slice(0, idx) as TModel;\n      }\n    }\n\n    this.opts = {\n      model: nextModel,\n      voice: nextVoice,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label() {\n    return 'inference.TTS';\n  }\n\n  static fromModelString(modelString: string): TTS<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, voice] = modelString.split(':') as [TTSModels, string];\n      return new TTS({ model, voice });\n    }\n    return new TTS({ model: modelString });\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  synthesize(_: string): ChunkedStream {\n    throw new Error('ChunkedStream is not implemented');\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream<TModel> {\n    const { connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const stream = new SynthesizeStream(this, { ...this.opts }, connOptions);\n    this.streams.add(stream);\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/tts`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const params = {\n      type: 'session.create',\n      sample_rate: String(this.opts.sampleRate),\n      encoding: this.opts.encoding,\n      extra: this.opts.modelOptions,\n    } as TtsSessionCreateEvent;\n\n    if (this.opts.voice) params.voice = this.opts.voice;\n    if (this.opts.model) params.model = this.opts.model;\n    if (this.opts.language) params.language = this.opts.language;\n\n    const socket = await connectWs(url, headers, timeout);\n    socket.send(JSON.stringify(params));\n    return socket;\n  }\n\n  async closeWs(ws: WebSocket) {\n    await ws.close();\n  }\n\n  async close() {\n    for (const stream of this.streams) {\n      await stream.close();\n    }\n    this.streams.clear();\n  }\n}\n\nexport class SynthesizeStream<TModel extends TTSModels> extends BaseSynthesizeStream {\n  private opts: InferenceTTSOptions<TModel>;\n  private tts: TTS<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(tts: TTS<TModel>, opts: InferenceTTSOptions<TModel>, connOptions: APIConnectOptions) {\n    super(tts, connOptions);\n    this.opts = opts;\n    this.tts = tts;\n    this.connOptions = connOptions;\n  }\n\n  get label() {\n    return 'inference.SynthesizeStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n  }\n\n  protected async run(): Promise<void> {\n    let ws: WebSocket | null = null;\n    let closing = false;\n    let finalReceived = false;\n    let lastFrame: AudioFrame | undefined;\n\n    const sendTokenizerStream = new tokenizeBasic.SentenceTokenizer().stream();\n    const eventChannel = createStreamChannel<TtsServerEvent>();\n    const requestId = shortuuid('tts_request_');\n\n    const resourceCleanup = () => {\n      if (closing) return;\n      closing = true;\n      sendTokenizerStream.close();\n      eventChannel.close();\n      ws?.removeAllListeners();\n      ws?.close();\n    };\n\n    const sendClientEvent = async (event: TtsClientEvent) => {\n      const validatedEvent = await ttsClientEventSchema.parseAsync(event);\n      if (!ws || ws.readyState !== WebSocket.OPEN) {\n        this.#logger.warn('Trying to send client TTS event to a closed WebSocket');\n        return;\n      }\n      ws.send(JSON.stringify(validatedEvent));\n    };\n\n    const sendLastFrame = (segmentId: string, final: boolean) => {\n      if (lastFrame) {\n        this.queue.put({ requestId, segmentId, frame: lastFrame, final });\n        lastFrame = undefined;\n      }\n    };\n\n    const createInputTask = async () => {\n      for await (const data of this.input) {\n        if (this.abortController.signal.aborted) break;\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          sendTokenizerStream.flush();\n          continue;\n        }\n        sendTokenizerStream.pushText(data);\n      }\n      sendTokenizerStream.endInput();\n    };\n\n    const createSentenceStreamTask = async () => {\n      for await (const ev of sendTokenizerStream) {\n        if (this.abortController.signal.aborted) break;\n\n        sendClientEvent({\n          type: 'input_transcript',\n          transcript: ev.token + ' ',\n        });\n      }\n\n      sendClientEvent({ type: 'session.flush' });\n    };\n\n    const createWsListenerTask = async (ws: WebSocket) => {\n      return new Promise<void>((resolve, reject) => {\n        this.abortController.signal.addEventListener('abort', () => {\n          resourceCleanup();\n          reject(new Error('WebSocket connection aborted'));\n        });\n\n        ws.on('message', async (data) => {\n          const eventJson = JSON.parse(data.toString()) as Record<string, unknown>;\n          const validatedEvent = ttsServerEventSchema.parse(eventJson);\n          eventChannel.write(validatedEvent);\n        });\n\n        ws.on('error', (e) => {\n          this.#logger.error({ error: e }, 'WebSocket error');\n          resourceCleanup();\n          reject(e);\n        });\n\n        ws.on('close', () => {\n          resourceCleanup();\n\n          if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n          if (finalReceived) return resolve();\n\n          reject(\n            new APIStatusError({\n              message: 'Gateway connection closed unexpectedly',\n              options: { requestId },\n            }),\n          );\n        });\n      });\n    };\n\n    const createRecvTask = async () => {\n      let currentSessionId: string | null = null;\n\n      const bstream = new AudioByteStream(this.opts.sampleRate, NUM_CHANNELS);\n      const serverEventStream = eventChannel.stream();\n      const reader = serverEventStream.getReader();\n\n      try {\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (this.abortController.signal.aborted) return;\n          if (result.done) return;\n\n          const serverEvent = result.value;\n          switch (serverEvent.type) {\n            case 'session.created':\n              currentSessionId = serverEvent.session_id;\n              break;\n            case 'output_audio':\n              const base64Data = new Int8Array(Buffer.from(serverEvent.audio, 'base64'));\n              for (const frame of bstream.write(base64Data.buffer)) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              break;\n            case 'done':\n              finalReceived = true;\n              for (const frame of bstream.flush()) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(currentSessionId!, true);\n              this.queue.put(SynthesizeStream.END_OF_STREAM);\n              break;\n            case 'session.closed':\n              resourceCleanup();\n              break;\n            case 'error':\n              this.#logger.error(\n                { serverEvent },\n                'Received error message from LiveKit TTS WebSocket',\n              );\n              resourceCleanup();\n              throw new APIError(`LiveKit TTS returned error: ${serverEvent.message}`);\n            default:\n              this.#logger.warn('Unexpected message %s', serverEvent);\n              break;\n          }\n        }\n      } finally {\n        reader.releaseLock();\n        try {\n          await serverEventStream.cancel();\n        } catch (e) {\n          this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n        }\n      }\n    };\n\n    try {\n      ws = await this.tts.connectWs(this.connOptions.timeoutMs);\n\n      await Promise.all([\n        createInputTask(),\n        createSentenceStreamTask(),\n        createWsListenerTask(ws),\n        createRecvTask(),\n      ]);\n    } catch (e) {\n      this.#logger.error({ error: e }, 'Error in SynthesizeStream');\n    } finally {\n      resourceCleanup();\n    }\n  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+{"version":3,"sources":["../../src/inference/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport { basic as tokenizeBasic } from '../tokenize/index.js';\nimport type { ChunkedStream } from '../tts/index.js';\nimport { SynthesizeStream as BaseSynthesizeStream, TTS as BaseTTS } from '../tts/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { shortuuid } from '../utils.js';\nimport {\n  type TtsClientEvent,\n  type TtsServerEvent,\n  type TtsSessionCreateEvent,\n  ttsClientEventSchema,\n  ttsServerEventSchema,\n} from './api_protos.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type CartesiaModels =\n  | 'cartesia'\n  | 'cartesia/sonic'\n  | 'cartesia/sonic-2'\n  | 'cartesia/sonic-turbo';\n\nexport type ElevenlabsModels =\n  | 'elevenlabs'\n  | 'elevenlabs/eleven_flash_v2'\n  | 'elevenlabs/eleven_flash_v2_5'\n  | 'elevenlabs/eleven_turbo_v2'\n  | 'elevenlabs/eleven_turbo_v2_5'\n  | 'elevenlabs/eleven_multilingual_v2';\n\nexport type RimeModels = 'rime' | 'rime/mist' | 'rime/mistv2' | 'rime/arcana';\n\nexport type InworldModels = 'inworld' | 'inworld/inworld-tts-1';\n\nexport interface CartesiaOptions {\n  duration?: number; // max duration of audio in seconds\n  speed?: 'slow' | 'normal' | 'fast'; // default: not specified\n}\n\nexport interface ElevenlabsOptions {\n  inactivity_timeout?: number; // default: 60\n  apply_text_normalization?: 'auto' | 'off' | 'on'; // default: \"auto\"\n}\n\nexport interface RimeOptions {}\n\nexport interface InworldOptions {}\n\ntype _TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels;\n\nexport type TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels | AnyString;\n\nexport type ModelWithVoice = `${_TTSModels}:${string}` | TTSModels;\n\nexport type TTSOptions<TModel extends TTSModels> = TModel extends CartesiaModels\n  ? CartesiaOptions\n  : TModel extends ElevenlabsModels\n    ? ElevenlabsOptions\n    : TModel extends RimeOptions\n      ? RimeOptions\n      : TModel extends InworldOptions\n        ? InworldOptions\n        : Record<string, unknown>;\n\ntype TTSEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: TTSEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'https://agent-gateway.livekit.cloud/v1';\nconst NUM_CHANNELS = 1;\nconst DEFAULT_LANGUAGE = 'en';\n\nexport interface InferenceTTSOptions<TModel extends TTSModels> {\n  model?: TModel;\n  voice?: string;\n  language?: string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: TTSOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference TTS\n */\nexport class TTS<TModel extends TTSModels> extends BaseTTS {\n  private opts: InferenceTTSOptions<TModel>;\n  private streams: Set<SynthesizeStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts: {\n    model: TModel;\n    voice?: string;\n    language?: string;\n    baseURL?: string;\n    encoding?: TTSEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: TTSOptions<TModel>;\n  }) {\n    const sampleRate = opts?.sampleRate ?? DEFAULT_SAMPLE_RATE;\n    super(sampleRate, 1, { streaming: true });\n\n    const {\n      model,\n      voice,\n      language = DEFAULT_LANGUAGE,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as TTSOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    // read voice id from the model if provided: \"provider/model:voice_id\"\n    let nextModel = model;\n    let nextVoice = voice;\n    if (typeof nextModel === 'string') {\n      const idx = nextModel.lastIndexOf(':');\n      if (idx !== -1) {\n        const voiceFromModel = nextModel.slice(idx + 1);\n        if (nextVoice && nextVoice !== voiceFromModel) {\n          this.#logger.warn(\n            '`voice` is provided via both argument and model, using the one from the argument',\n            { voice: nextVoice, model: nextModel },\n          );\n        } else {\n          nextVoice = voiceFromModel;\n        }\n        nextModel = nextModel.slice(0, idx) as TModel;\n      }\n    }\n\n    this.opts = {\n      model: nextModel,\n      voice: nextVoice,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label() {\n    return 'inference.TTS';\n  }\n\n  static fromModelString(modelString: string): TTS<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, voice] = modelString.split(':') as [TTSModels, string];\n      return new TTS({ model, voice });\n    }\n    return new TTS({ model: modelString });\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  synthesize(_: string): ChunkedStream {\n    throw new Error('ChunkedStream is not implemented');\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream<TModel> {\n    const { connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const stream = new SynthesizeStream(this, { ...this.opts }, connOptions);\n    this.streams.add(stream);\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/tts`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const params = {\n      type: 'session.create',\n      sample_rate: String(this.opts.sampleRate),\n      encoding: this.opts.encoding,\n      extra: this.opts.modelOptions,\n    } as TtsSessionCreateEvent;\n\n    if (this.opts.voice) params.voice = this.opts.voice;\n    if (this.opts.model) params.model = this.opts.model;\n    if (this.opts.language) params.language = this.opts.language;\n\n    const socket = await connectWs(url, headers, timeout);\n    socket.send(JSON.stringify(params));\n    return socket;\n  }\n\n  async closeWs(ws: WebSocket) {\n    await ws.close();\n  }\n\n  async close() {\n    for (const stream of this.streams) {\n      await stream.close();\n    }\n    this.streams.clear();\n  }\n}\n\nexport class SynthesizeStream<TModel extends TTSModels> extends BaseSynthesizeStream {\n  private opts: InferenceTTSOptions<TModel>;\n  private tts: TTS<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(tts: TTS<TModel>, opts: InferenceTTSOptions<TModel>, connOptions: APIConnectOptions) {\n    super(tts, connOptions);\n    this.opts = opts;\n    this.tts = tts;\n    this.connOptions = connOptions;\n  }\n\n  get label() {\n    return 'inference.SynthesizeStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n  }\n\n  protected async run(): Promise<void> {\n    let ws: WebSocket | null = null;\n    let closing = false;\n    let finalReceived = false;\n    let lastFrame: AudioFrame | undefined;\n\n    const sendTokenizerStream = new tokenizeBasic.SentenceTokenizer().stream();\n    const eventChannel = createStreamChannel<TtsServerEvent>();\n    const requestId = shortuuid('tts_request_');\n\n    const resourceCleanup = () => {\n      if (closing) return;\n      closing = true;\n      sendTokenizerStream.close();\n      eventChannel.close();\n      ws?.removeAllListeners();\n      ws?.close();\n    };\n\n    const sendClientEvent = async (event: TtsClientEvent) => {\n      // Don't send events to a closed WebSocket or aborted controller\n      if (this.abortController.signal.aborted || closing) return;\n\n      const validatedEvent = await ttsClientEventSchema.parseAsync(event);\n      if (!ws || ws.readyState !== WebSocket.OPEN) {\n        this.#logger.warn('Trying to send client TTS event to a closed WebSocket');\n        return;\n      }\n      ws.send(JSON.stringify(validatedEvent));\n    };\n\n    const sendLastFrame = (segmentId: string, final: boolean) => {\n      if (lastFrame) {\n        this.queue.put({ requestId, segmentId, frame: lastFrame, final });\n        lastFrame = undefined;\n      }\n    };\n\n    const createInputTask = 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shutdown, not an error\n        });\n\n        ws.on('message', async (data) => {\n          const eventJson = JSON.parse(data.toString()) as Record<string, unknown>;\n          const validatedEvent = ttsServerEventSchema.parse(eventJson);\n          eventChannel.write(validatedEvent);\n        });\n\n        ws.on('error', (e) => {\n          this.#logger.error({ error: e }, 'WebSocket error');\n          resourceCleanup();\n          reject(e);\n        });\n\n        ws.on('close', () => {\n          resourceCleanup();\n\n          if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n          if (finalReceived) return resolve();\n\n          reject(\n            new APIStatusError({\n              message: 'Gateway connection closed unexpectedly',\n              options: { requestId },\n            }),\n          );\n        });\n      });\n    };\n\n    const createRecvTask = async () => {\n      let currentSessionId: string | null = null;\n\n      const bstream = new AudioByteStream(this.opts.sampleRate, NUM_CHANNELS);\n      const serverEventStream = eventChannel.stream();\n      const reader = serverEventStream.getReader();\n\n      try {\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (this.abortController.signal.aborted) return;\n          if (result.done) return;\n\n          const serverEvent = result.value;\n          switch (serverEvent.type) {\n            case 'session.created':\n              currentSessionId = serverEvent.session_id;\n              break;\n            case 'output_audio':\n              const base64Data = new Int8Array(Buffer.from(serverEvent.audio, 'base64'));\n              for (const frame of bstream.write(base64Data.buffer)) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              break;\n            case 'done':\n              finalReceived = true;\n              for (const frame of bstream.flush()) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(currentSessionId!, true);\n              this.queue.put(SynthesizeStream.END_OF_STREAM);\n              break;\n            case 'session.closed':\n              resourceCleanup();\n              break;\n            case 'error':\n              this.#logger.error(\n                { serverEvent },\n                'Received error message from LiveKit TTS WebSocket',\n              );\n              resourceCleanup();\n              throw new APIError(`LiveKit TTS returned error: ${serverEvent.message}`);\n            default:\n              this.#logger.warn('Unexpected message %s', serverEvent);\n              break;\n          }\n        }\n      } finally {\n        reader.releaseLock();\n        try {\n          await serverEventStream.cancel();\n        } catch (e) {\n          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package/dist/inference/tts.d.ts.map

@@ -1 +1 @@
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package/dist/inference/tts.js

@@ -162,6 +162,7 @@ class SynthesizeStream extends BaseSynthesizeStream {
       ws == null ? void 0 : ws.close();
     };
     const sendClientEvent = async (event) => {
+      if (this.abortController.signal.aborted || closing) return;
       const validatedEvent = await ttsClientEventSchema.parseAsync(event);
       if (!ws || ws.readyState !== WebSocket.OPEN) {
         this.#logger.warn("Trying to send client TTS event to a closed WebSocket");
@@ -200,7 +201,7 @@ class SynthesizeStream extends BaseSynthesizeStream {
       return new Promise((resolve, reject) => {
         this.abortController.signal.addEventListener("abort", () => {
           resourceCleanup();
-          reject(new Error("WebSocket connection aborted"));
+          resolve();
         });
         ws2.on("message", async (data) => {
           const eventJson = JSON.parse(data.toString());
@@ -288,8 +289,6 @@ class SynthesizeStream extends BaseSynthesizeStream {
         createWsListenerTask(ws),
         createRecvTask()
       ]);
-    } catch (e) {
-      this.#logger.error({ error: e }, "Error in SynthesizeStream");
     } finally {
       resourceCleanup();
     }

package/dist/inference/tts.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/inference/tts.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { WebSocket } from 'ws';\nimport { APIError, APIStatusError } from '../_exceptions.js';\nimport { AudioByteStream } from '../audio.js';\nimport { log } from '../log.js';\nimport { createStreamChannel } from '../stream/stream_channel.js';\nimport { basic as tokenizeBasic } from '../tokenize/index.js';\nimport type { ChunkedStream } from '../tts/index.js';\nimport { SynthesizeStream as BaseSynthesizeStream, TTS as BaseTTS } from '../tts/index.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport { shortuuid } from '../utils.js';\nimport {\n  type TtsClientEvent,\n  type TtsServerEvent,\n  type TtsSessionCreateEvent,\n  ttsClientEventSchema,\n  ttsServerEventSchema,\n} from './api_protos.js';\nimport { type AnyString, connectWs, createAccessToken } from './utils.js';\n\nexport type CartesiaModels =\n  | 'cartesia'\n  | 'cartesia/sonic'\n  | 'cartesia/sonic-2'\n  | 'cartesia/sonic-turbo';\n\nexport type ElevenlabsModels =\n  | 'elevenlabs'\n  | 'elevenlabs/eleven_flash_v2'\n  | 'elevenlabs/eleven_flash_v2_5'\n  | 'elevenlabs/eleven_turbo_v2'\n  | 'elevenlabs/eleven_turbo_v2_5'\n  | 'elevenlabs/eleven_multilingual_v2';\n\nexport type RimeModels = 'rime' | 'rime/mist' | 'rime/mistv2' | 'rime/arcana';\n\nexport type InworldModels = 'inworld' | 'inworld/inworld-tts-1';\n\nexport interface CartesiaOptions {\n  duration?: number; // max duration of audio in seconds\n  speed?: 'slow' | 'normal' | 'fast'; // default: not specified\n}\n\nexport interface ElevenlabsOptions {\n  inactivity_timeout?: number; // default: 60\n  apply_text_normalization?: 'auto' | 'off' | 'on'; // default: \"auto\"\n}\n\nexport interface RimeOptions {}\n\nexport interface InworldOptions {}\n\ntype _TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels;\n\nexport type TTSModels = CartesiaModels | ElevenlabsModels | RimeModels | InworldModels | AnyString;\n\nexport type ModelWithVoice = `${_TTSModels}:${string}` | TTSModels;\n\nexport type TTSOptions<TModel extends TTSModels> = TModel extends CartesiaModels\n  ? CartesiaOptions\n  : TModel extends ElevenlabsModels\n    ? ElevenlabsOptions\n    : TModel extends RimeOptions\n      ? RimeOptions\n      : TModel extends InworldOptions\n        ? InworldOptions\n        : Record<string, unknown>;\n\ntype TTSEncoding = 'pcm_s16le';\n\nconst DEFAULT_ENCODING: TTSEncoding = 'pcm_s16le';\nconst DEFAULT_SAMPLE_RATE = 16000;\nconst DEFAULT_BASE_URL = 'https://agent-gateway.livekit.cloud/v1';\nconst NUM_CHANNELS = 1;\nconst DEFAULT_LANGUAGE = 'en';\n\nexport interface InferenceTTSOptions<TModel extends TTSModels> {\n  model?: TModel;\n  voice?: string;\n  language?: string;\n  encoding: TTSEncoding;\n  sampleRate: number;\n  baseURL: string;\n  apiKey: string;\n  apiSecret: string;\n  modelOptions: TTSOptions<TModel>;\n}\n\n/**\n * Livekit Cloud Inference TTS\n */\nexport class TTS<TModel extends TTSModels> extends BaseTTS {\n  private opts: InferenceTTSOptions<TModel>;\n  private streams: Set<SynthesizeStream<TModel>> = new Set();\n\n  #logger = log();\n\n  constructor(opts: {\n    model: TModel;\n    voice?: string;\n    language?: string;\n    baseURL?: string;\n    encoding?: TTSEncoding;\n    sampleRate?: number;\n    apiKey?: string;\n    apiSecret?: string;\n    modelOptions?: TTSOptions<TModel>;\n  }) {\n    const sampleRate = opts?.sampleRate ?? DEFAULT_SAMPLE_RATE;\n    super(sampleRate, 1, { streaming: true });\n\n    const {\n      model,\n      voice,\n      language = DEFAULT_LANGUAGE,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as TTSOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    // read voice id from the model if provided: \"provider/model:voice_id\"\n    let nextModel = model;\n    let nextVoice = voice;\n    if (typeof nextModel === 'string') {\n      const idx = nextModel.lastIndexOf(':');\n      if (idx !== -1) {\n        const voiceFromModel = nextModel.slice(idx + 1);\n        if (nextVoice && nextVoice !== voiceFromModel) {\n          this.#logger.warn(\n            '`voice` is provided via both argument and model, using the one from the argument',\n            { voice: nextVoice, model: nextModel },\n          );\n        } else {\n          nextVoice = voiceFromModel;\n        }\n        nextModel = nextModel.slice(0, idx) as TModel;\n      }\n    }\n\n    this.opts = {\n      model: nextModel,\n      voice: nextVoice,\n      language,\n      encoding,\n      sampleRate,\n      baseURL: lkBaseURL,\n      apiKey: lkApiKey,\n      apiSecret: lkApiSecret,\n      modelOptions,\n    };\n  }\n\n  get label() {\n    return 'inference.TTS';\n  }\n\n  static fromModelString(modelString: string): TTS<AnyString> {\n    if (modelString.includes(':')) {\n      const [model, voice] = modelString.split(':') as [TTSModels, string];\n      return new TTS({ model, voice });\n    }\n    return new TTS({ model: modelString });\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  synthesize(_: string): ChunkedStream {\n    throw new Error('ChunkedStream is not implemented');\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream<TModel> {\n    const { connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const stream = new SynthesizeStream(this, { ...this.opts }, connOptions);\n    this.streams.add(stream);\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 'ws');\n    }\n\n    const token = await createAccessToken(this.opts.apiKey, this.opts.apiSecret);\n    const url = `${baseURL}/tts`;\n    const headers = { Authorization: `Bearer ${token}` } as Record<string, string>;\n\n    const params = {\n      type: 'session.create',\n      sample_rate: String(this.opts.sampleRate),\n      encoding: this.opts.encoding,\n      extra: this.opts.modelOptions,\n    } as TtsSessionCreateEvent;\n\n    if (this.opts.voice) params.voice = this.opts.voice;\n    if (this.opts.model) params.model = this.opts.model;\n    if (this.opts.language) params.language = this.opts.language;\n\n    const socket = await connectWs(url, headers, timeout);\n    socket.send(JSON.stringify(params));\n    return socket;\n  }\n\n  async closeWs(ws: WebSocket) {\n    await ws.close();\n  }\n\n  async close() {\n    for (const stream of this.streams) {\n      await stream.close();\n    }\n    this.streams.clear();\n  }\n}\n\nexport class SynthesizeStream<TModel extends TTSModels> extends BaseSynthesizeStream {\n  private opts: InferenceTTSOptions<TModel>;\n  private tts: TTS<TModel>;\n  private connOptions: APIConnectOptions;\n\n  #logger = log();\n\n  constructor(tts: TTS<TModel>, opts: InferenceTTSOptions<TModel>, connOptions: APIConnectOptions) {\n    super(tts, connOptions);\n    this.opts = opts;\n    this.tts = tts;\n    this.connOptions = connOptions;\n  }\n\n  get label() {\n    return 'inference.SynthesizeStream';\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n  }\n\n  protected async run(): Promise<void> {\n    let ws: WebSocket | null = null;\n    let closing = false;\n    let finalReceived = false;\n    let lastFrame: AudioFrame | undefined;\n\n    const sendTokenizerStream = new tokenizeBasic.SentenceTokenizer().stream();\n    const eventChannel = createStreamChannel<TtsServerEvent>();\n    const requestId = shortuuid('tts_request_');\n\n    const resourceCleanup = () => {\n      if (closing) return;\n      closing = true;\n      sendTokenizerStream.close();\n      eventChannel.close();\n      ws?.removeAllListeners();\n      ws?.close();\n    };\n\n    const sendClientEvent = async (event: TtsClientEvent) => {\n      const validatedEvent = await ttsClientEventSchema.parseAsync(event);\n      if (!ws || ws.readyState !== WebSocket.OPEN) {\n        this.#logger.warn('Trying to send client TTS event to a closed WebSocket');\n        return;\n      }\n      ws.send(JSON.stringify(validatedEvent));\n    };\n\n    const sendLastFrame = (segmentId: string, final: boolean) => {\n      if (lastFrame) {\n        this.queue.put({ requestId, segmentId, frame: lastFrame, final });\n        lastFrame = undefined;\n      }\n    };\n\n    const createInputTask = async () => {\n      for await (const data of this.input) {\n        if (this.abortController.signal.aborted) break;\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          sendTokenizerStream.flush();\n          continue;\n        }\n        sendTokenizerStream.pushText(data);\n      }\n      sendTokenizerStream.endInput();\n    };\n\n    const createSentenceStreamTask = async () => {\n      for await (const ev of sendTokenizerStream) {\n        if (this.abortController.signal.aborted) break;\n\n        sendClientEvent({\n          type: 'input_transcript',\n          transcript: ev.token + ' ',\n        });\n      }\n\n      sendClientEvent({ type: 'session.flush' });\n    };\n\n    const createWsListenerTask = async (ws: WebSocket) => {\n      return new Promise<void>((resolve, reject) => {\n        this.abortController.signal.addEventListener('abort', () => {\n          resourceCleanup();\n          reject(new Error('WebSocket connection aborted'));\n        });\n\n        ws.on('message', async (data) => {\n          const eventJson = JSON.parse(data.toString()) as Record<string, unknown>;\n          const validatedEvent = ttsServerEventSchema.parse(eventJson);\n          eventChannel.write(validatedEvent);\n        });\n\n        ws.on('error', (e) => {\n          this.#logger.error({ error: e }, 'WebSocket error');\n          resourceCleanup();\n          reject(e);\n        });\n\n        ws.on('close', () => {\n          resourceCleanup();\n\n          if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n          if (finalReceived) return resolve();\n\n          reject(\n            new APIStatusError({\n              message: 'Gateway connection closed unexpectedly',\n              options: { requestId },\n            }),\n          );\n        });\n      });\n    };\n\n    const createRecvTask = async () => {\n      let currentSessionId: string | null = null;\n\n      const bstream = new AudioByteStream(this.opts.sampleRate, NUM_CHANNELS);\n      const serverEventStream = eventChannel.stream();\n      const reader = serverEventStream.getReader();\n\n    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CartesiaOptions\n  : TModel extends ElevenlabsModels\n    ? ElevenlabsOptions\n    : TModel extends RimeOptions\n      ? RimeOptions\n      : TModel extends InworldOptions\n        ? 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DEFAULT_SAMPLE_RATE;\n    super(sampleRate, 1, { streaming: true });\n\n    const {\n      model,\n      voice,\n      language = DEFAULT_LANGUAGE,\n      baseURL,\n      encoding = DEFAULT_ENCODING,\n      apiKey,\n      apiSecret,\n      modelOptions = {} as TTSOptions<TModel>,\n    } = opts || {};\n\n    const lkBaseURL = baseURL || process.env.LIVEKIT_INFERENCE_URL || DEFAULT_BASE_URL;\n    const lkApiKey = apiKey || process.env.LIVEKIT_INFERENCE_API_KEY || process.env.LIVEKIT_API_KEY;\n    if (!lkApiKey) {\n      throw new Error('apiKey is required: pass apiKey or set LIVEKIT_API_KEY');\n    }\n\n    const lkApiSecret =\n      apiSecret || process.env.LIVEKIT_INFERENCE_API_SECRET || process.env.LIVEKIT_API_SECRET;\n    if (!lkApiSecret) {\n      throw new Error('apiSecret is required: pass apiSecret or set LIVEKIT_API_SECRET');\n    }\n\n    // read voice id from the model if provided: \"provider/model:voice_id\"\n    let nextModel = model;\n    let nextVoice = voice;\n    if 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modelString.split(':') as [TTSModels, string];\n      return new TTS({ model, voice });\n    }\n    return new TTS({ model: modelString });\n  }\n\n  updateOptions(opts: Partial<Pick<InferenceTTSOptions<TModel>, 'model' | 'voice' | 'language'>>) {\n    this.opts = { ...this.opts, ...opts };\n    for (const stream of this.streams) {\n      stream.updateOptions(opts);\n    }\n  }\n\n  synthesize(_: string): ChunkedStream {\n    throw new Error('ChunkedStream is not implemented');\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): SynthesizeStream<TModel> {\n    const { connOptions = DEFAULT_API_CONNECT_OPTIONS } = options || {};\n    const stream = new SynthesizeStream(this, { ...this.opts }, connOptions);\n    this.streams.add(stream);\n    return stream;\n  }\n\n  async connectWs(timeout: number): Promise<WebSocket> {\n    let baseURL = this.opts.baseURL;\n    if (baseURL.startsWith('http://') || baseURL.startsWith('https://')) {\n      baseURL = baseURL.replace('http', 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shortuuid('tts_request_');\n\n    const resourceCleanup = () => {\n      if (closing) return;\n      closing = true;\n      sendTokenizerStream.close();\n      eventChannel.close();\n      ws?.removeAllListeners();\n      ws?.close();\n    };\n\n    const sendClientEvent = async (event: TtsClientEvent) => {\n      // Don't send events to a closed WebSocket or aborted controller\n      if (this.abortController.signal.aborted || closing) return;\n\n      const validatedEvent = await ttsClientEventSchema.parseAsync(event);\n      if (!ws || ws.readyState !== WebSocket.OPEN) {\n        this.#logger.warn('Trying to send client TTS event to a closed WebSocket');\n        return;\n      }\n      ws.send(JSON.stringify(validatedEvent));\n    };\n\n    const sendLastFrame = (segmentId: string, final: boolean) => {\n      if (lastFrame) {\n        this.queue.put({ requestId, segmentId, frame: lastFrame, final });\n        lastFrame = undefined;\n      }\n    };\n\n    const createInputTask = async () => {\n      for await (const data of this.input) {\n        if (this.abortController.signal.aborted) break;\n        if (data === SynthesizeStream.FLUSH_SENTINEL) {\n          sendTokenizerStream.flush();\n          continue;\n        }\n        sendTokenizerStream.pushText(data);\n      }\n      sendTokenizerStream.endInput();\n    };\n\n    const createSentenceStreamTask = async () => {\n      for await (const ev of sendTokenizerStream) {\n        if (this.abortController.signal.aborted) break;\n\n        sendClientEvent({\n          type: 'input_transcript',\n          transcript: ev.token + ' ',\n        });\n      }\n\n      sendClientEvent({ type: 'session.flush' });\n    };\n\n    const createWsListenerTask = async (ws: WebSocket) => {\n      return new Promise<void>((resolve, reject) => {\n        this.abortController.signal.addEventListener('abort', () => {\n          resourceCleanup();\n          resolve(); // Abort is triggered by close(), which is a normal shutdown, not an error\n        });\n\n        ws.on('message', async (data) => {\n          const eventJson = JSON.parse(data.toString()) as Record<string, unknown>;\n          const validatedEvent = ttsServerEventSchema.parse(eventJson);\n          eventChannel.write(validatedEvent);\n        });\n\n        ws.on('error', (e) => {\n          this.#logger.error({ error: e }, 'WebSocket error');\n          resourceCleanup();\n          reject(e);\n        });\n\n        ws.on('close', () => {\n          resourceCleanup();\n\n          if (!closing) return this.#logger.error('WebSocket closed unexpectedly');\n          if (finalReceived) return resolve();\n\n          reject(\n            new APIStatusError({\n              message: 'Gateway connection closed unexpectedly',\n              options: { requestId },\n            }),\n          );\n        });\n      });\n    };\n\n    const createRecvTask = async () => {\n      let currentSessionId: string | null = null;\n\n      const bstream = new AudioByteStream(this.opts.sampleRate, NUM_CHANNELS);\n      const serverEventStream = eventChannel.stream();\n      const reader = serverEventStream.getReader();\n\n      try {\n        while (!this.closed && !this.abortController.signal.aborted) {\n          const result = await reader.read();\n          if (this.abortController.signal.aborted) return;\n          if (result.done) return;\n\n          const serverEvent = result.value;\n          switch (serverEvent.type) {\n            case 'session.created':\n              currentSessionId = serverEvent.session_id;\n              break;\n            case 'output_audio':\n              const base64Data = new Int8Array(Buffer.from(serverEvent.audio, 'base64'));\n              for (const frame of bstream.write(base64Data.buffer)) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              break;\n            case 'done':\n              finalReceived = true;\n              for (const frame of bstream.flush()) {\n                sendLastFrame(currentSessionId!, false);\n                lastFrame = frame;\n              }\n              sendLastFrame(currentSessionId!, true);\n              this.queue.put(SynthesizeStream.END_OF_STREAM);\n              break;\n            case 'session.closed':\n              resourceCleanup();\n              break;\n            case 'error':\n              this.#logger.error(\n                { serverEvent },\n                'Received error message from LiveKit TTS WebSocket',\n              );\n              resourceCleanup();\n              throw new APIError(`LiveKit TTS returned error: ${serverEvent.message}`);\n            default:\n              this.#logger.warn('Unexpected message %s', serverEvent);\n              break;\n          }\n        }\n      } finally {\n        reader.releaseLock();\n        try {\n          await serverEventStream.cancel();\n        } catch (e) {\n          this.#logger.debug('Error cancelling serverEventStream (may already be cancelled):', e);\n        }\n      }\n    };\n\n    try {\n      ws = await this.tts.connectWs(this.connOptions.timeoutMs);\n\n      await Promise.all([\n        createInputTask(),\n        createSentenceStreamTask(),\n        createWsListenerTask(ws),\n        createRecvTask(),\n      ]);\n    } finally {\n      resourceCleanup();\n    }\n  }\n}\n"],"mappings":"AAIA,SAAS,iBAAiB;AAC1B,SAAS,UAAU,sBAAsB;AACzC,SAAS,uBAAuB;AAChC,SAAS,WAAW;AACpB,SAAS,2BAA2B;AACpC,SAAS,SAAS,qBAAqB;AAEvC,SAAS,oBAAoB,sBAAsB,OAAO,eAAe;AACzE,SAAiC,mCAAmC;AACpE,SAAS,iBAAiB;AAC1B;AAAA,EAIE;AAAA,EACA;AAAA,OACK;AACP,SAAyB,WAAW,yBAAyB;AAoD7D,MAAM,mBAAgC;AACtC,MAAM,sBAAsB;AAC5B,MAAM,mBAAmB;AACzB,MAAM,eAAe;AACrB,MAAM,mBAAmB;AAiBlB,MAAM,YAAsC,QAAQ;AAAA,EACjD;AAAA,EACA,UAAyC,oBAAI,IAAI;AAAA,EAEzD,UAAU,IAAI;AAAA,EAEd,YAAY,MAUT;AACD,UAAM,cAAa,6BAAM,eAAc;AACvC,UAAM,YAAY,GAAG,EAAE,WAAW,KAAK,CAAC;AAExC,UAAM;AAAA,MACJ;AAAA,MACA;AAAA,MACA,WAAW;AAAA,MACX;AAAA,MACA,WAAW;AAAA,MACX;AAAA,MACA;AAAA,MACA,eAAe,CAAC;AAAA,IAClB,IAAI,QAAQ,CAAC;AAEb,UAAM,YAAY,WAAW,QAAQ,IAAI,yBAAyB;AAClE,UAAM,WAAW,UAAU,QAAQ,IAAI,6BAA6B,QAAQ,IAAI;AAChF,QAAI,CAAC,UAAU;AACb,YAAM,IAAI,MAAM,wDAAwD;AAAA,IAC1E;AAEA,UAAM,cACJ,aAAa,QAAQ,IAAI,gCAAgC,QAAQ,IAAI;AACvE,QAAI,CAAC,aAAa;AAChB,YAAM,IAAI,MAAM,iEAAiE;AAAA,IACnF;AAGA,QAAI,YAAY;AAChB,QAAI,YAAY;AAChB,QAAI,OAAO,cAAc,UAAU;AACjC,YAAM,MAAM,UAAU,YAAY,GAAG;AACrC,UAAI,QAAQ,IAAI;AACd,cAAM,iBAAiB,UAAU,MAAM,MAAM,CAAC;AAC9C,YAAI,aAAa,cAAc,gBAAgB;AAC7C,eAAK,QAAQ;AAAA,YACX;AAAA,YACA,EAAE,OAAO,WAAW,OAAO,UAAU;AAAA,UACvC;AAAA,QACF,OAAO;AACL,sBAAY;AAAA,QACd;AACA,oBAAY,UAAU,MAAM,GAAG,GAAG;AAAA,MACpC;AAAA,IACF;AAEA,SAAK,OAAO;AAAA,MACV,OAAO;AAAA,MACP,OAAO;AAAA,MACP;AAAA,MACA;AAAA,MACA;AAAA,MACA,SAAS;AAAA,MACT,QAAQ;AAAA,MACR,WAAW;AAAA,MACX;AAAA,IACF;AAAA,EACF;AAAA,EAEA,IAAI,QAAQ;AACV,WAAO;AAAA,EACT;AAAA,EAEA,OAAO,gBAAgB,aAAqC;AAC1D,QAAI,YAAY,SAAS,GAAG,GAAG;AAC7B,YAAM,CAAC,OAAO,KAAK,IAAI,YAAY,MAAM,GAAG;AAC5C,aAAO,IAAI,IAAI,EAAE,OAAO,MAAM,CAAC;AAAA,IACjC;AACA,WAAO,IAAI,IAAI,EAAE,OAAO,YAAY,CAAC;AAAA,EACvC;AAAA,EAEA,cAAc,MAAkF;AAC9F,SAAK,OAAO,EAAE,GAAG,KAAK,MAAM,GAAG,KAAK;AACpC,eAAW,UAAU,KAAK,SAAS;AACjC,aAAO,cAAc,IAAI;AAAA,IAC3B;AAAA,EACF;AAAA,EAEA,WAAW,GAA0B;AACnC,UAAM,IAAI,MAAM,kCAAkC;AAAA,EACpD;AAAA,EAEA,OAAO,SAAyE;AAC9E,UAAM,EAAE,cAAc,4BAA4B,IAAI,WAAW,CAAC;AAClE,UAAM,SAAS,IAAI,iBAAiB,MAAM,EAAE,GAAG,KAAK,KAAK,GAAG,WAAW;AACvE,SAAK,QAAQ,IAAI,MAAM;AACvB,WAAO;AAAA,EACT;AAAA,EAEA,MAAM,UAAU,SAAqC;AACnD,QAAI,UAAU,KAAK,KAAK;AACxB,QAAI,QAAQ,WAAW,SAAS,KAAK,QAAQ,WAAW,UAAU,GAAG;AACnE,gBAAU,QAAQ,QAAQ,QAAQ,IAAI;AAAA,IACxC;AAEA,UAAM,QAAQ,MAAM,kBAAkB,KAAK,KAAK,QAAQ,KAAK,KAAK,SAAS;AAC3E,UAAM,MAAM,GAAG,OAAO;AACtB,UAAM,UAAU,EAAE,eAAe,UAAU,KAAK,GAAG;AAEnD,UAAM,SAAS;AAAA,MACb,MAAM;AAAA,MACN,aAAa,OAAO,KAAK,KAAK,UAAU;AAAA,MACxC,UAAU,KAAK,KAAK;AAAA,MACpB,OAAO,KAAK,KAAK;AAAA,IACnB;AAEA,QAAI,KAAK,KAAK,MAAO,QAAO,QAAQ,KAAK,KAAK;AAC9C,QAAI,KAAK,KAAK,MAAO,QAAO,QAAQ,KAAK,KAAK;AAC9C,QAAI,KAAK,KAAK,SAAU,QAAO,WAAW,KAAK,KAAK;AAEpD,UAAM,SAAS,MAAM,UAAU,KAAK,SAAS,OAAO;AACpD,WAAO,KAAK,KAAK,UAAU,MAAM,CAAC;AAClC,WAAO;AAAA,EACT;AAAA,EAEA,MAAM,QAAQ,IAAe;AAC3B,UAAM,GAAG,MAAM;AAAA,EACjB;AAAA,EAEA,MAAM,QAAQ;AACZ,eAAW,UAAU,KAAK,SAAS;AACjC,YAAM,OAAO,MAAM;AAAA,IACrB;AACA,SAAK,QAAQ,MAAM;AAAA,EACrB;AACF;AAEO,MAAM,yBAAmD,qBAAqB;AAAA,EAC3E;AAAA,EACA;AAAA,EACA;AAAA,EAER,UAAU,IAAI;AAAA,EAEd,YAAY,KAAkB,MAAmC,aAAgC;AAC/F,UAAM,KAAK,WAAW;AACtB,SAAK,OAAO;AACZ,SAAK,MAAM;AACX,SAAK,cAAc;AAAA,EACrB;AAAA,EAEA,IAAI,QAAQ;AACV,WAAO;AAAA,EACT;AAAA,EAEA,cAAc,MAAkF;AAC9F,SAAK,OAAO,EAAE,GAAG,KAAK,MAAM,GAAG,KAAK;AAAA,EACtC;AAAA,EAEA,MAAgB,MAAqB;AACnC,QAAI,KAAuB;AAC3B,QAAI,UAAU;AACd,QAAI,gBAAgB;AACpB,QAAI;AAEJ,UAAM,sBAAsB,IAAI,cAAc,kBAAkB,EAAE,OAAO;AACzE,UAAM,eAAe,oBAAoC;AACzD,UAAM,YAAY,UAAU,cAAc;AAE1C,UAAM,kBAAkB,MAAM;AAC5B,UAAI,QAAS;AACb,gBAAU;AACV,0BAAoB,MAAM;AAC1B,mBAAa,MAAM;AACnB,+BAAI;AACJ,+BAAI;AAAA,IACN;AAEA,UAAM,kBAAkB,OAAO,UAA0B;AAEvD,UAAI,KAAK,gBAAgB,OAAO,WAAW,QAAS;AAEpD,YAAM,iBAAiB,MAAM,qBAAqB,WAAW,KAAK;AAClE,UAAI,CAAC,MAAM,GAAG,eAAe,UAAU,MAAM;AAC3C,aAAK,QAAQ,KAAK,uDAAuD;AACzE;AAAA,MACF;AACA,SAAG,KAAK,KAAK,UAAU,cAAc,CAAC;AAAA,IACxC;AAEA,UAAM,gBAAgB,CAAC,WAAmB,UAAmB;AAC3D,UAAI,WAAW;AACb,aAAK,MAAM,IAAI,EAAE,WAAW,WAAW,OAAO,WAAW,MAAM,CAAC;AAChE,oBAAY;AAAA,MACd;AAAA,IACF;AAEA,UAAM,kBAAkB,YAAY;AAClC,uBAAiB,QAAQ,KAAK,OAAO;AACnC,YAAI,KAAK,gBAAgB,OAAO,QAAS;AACzC,YAAI,SAAS,iBAAiB,gBAAgB;AAC5C,8BAAoB,MAAM;AAC1B;AAAA,QACF;AACA,4BAAoB,SAAS,IAAI;AAAA,MACnC;AACA,0BAAoB,SAAS;AAAA,IAC/B;AAEA,UAAM,2BAA2B,YAAY;AAC3C,uBAAiB,MAAM,qBAAqB;AAC1C,YAAI,KAAK,gBAAgB,OAAO,QAAS;AAEzC,wBAAgB;AAAA,UACd,MAAM;AAAA,UACN,YAAY,GAAG,QAAQ;AAAA,QACzB,CAAC;AAAA,MACH;AAEA,sBAAgB,EAAE,MAAM,gBAAgB,CAAC;AAAA,IAC3C;AAEA,UAAM,uBAAuB,OAAOA,QAAkB;AACpD,aAAO,IAAI,QAAc,CAAC,SAAS,WAAW;AAC5C,aAAK,gBAAgB,OAAO,iBAAiB,SAAS,MAAM;AAC1D,0BAAgB;AAChB,kBAAQ;AAAA,QACV,CAAC;AAED,QAAAA,IAAG,GAAG,WAAW,OAAO,SAAS;AAC/B,gBAAM,YAAY,KAAK,MAAM,KAAK,SAAS,CAAC;AAC5C,gBAAM,iBAAiB,qBAAqB,MAAM,SAAS;AAC3D,uBAAa,MAAM,cAAc;AAAA,QACnC,CAAC;AAED,QAAAA,IAAG,GAAG,SAAS,CAAC,MAAM;AACpB,eAAK,QAAQ,MAAM,EAAE,OAAO,EAAE,GAAG,iBAAiB;AAClD,0BAAgB;AAChB,iBAAO,CAAC;AAAA,QACV,CAAC;AAED,QAAAA,IAAG,GAAG,SAAS,MAAM;AACnB,0BAAgB;AAEhB,cAAI,CAAC,QAAS,QAAO,KAAK,QAAQ,MAAM,+BAA+B;AACvE,cAAI,cAAe,QAAO,QAAQ;AAElC;AAAA,YACE,IAAI,eAAe;AAAA,cACjB,SAAS;AAAA,cACT,SAAS,EAAE,UAAU;AAAA,YACvB,CAAC;AAAA,UACH;AAAA,QACF,CAAC;AAAA,MACH,CAAC;AAAA,IACH;AAEA,UAAM,iBAAiB,YAAY;AACjC,UAAI,mBAAkC;AAEtC,YAAM,UAAU,IAAI,gBAAgB,KAAK,KAAK,YAAY,YAAY;AACtE,YAAM,oBAAoB,aAAa,OAAO;AAC9C,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package/dist/stt/stt.cjs

@@ -149,7 +149,18 @@ class SpeechStream {
   }
   async monitorMetrics() {
     for await (const event of this.queue) {
-      this.output.put(event);
+      if (!this.output.closed) {
+        try {
+          this.output.put(event);
+        } catch (e) {
+          if (e instanceof Error && e.message.includes("Queue is closed")) {
+            this.logger.warn(
+              { err: e },
+              "Queue closed during transcript processing (expected during disconnect)"
+            );
+          }
+        }
+      }
       if (event.type !== 4 /* RECOGNITION_USAGE */) continue;
       const metrics = {
         type: "stt_metrics",
@@ -162,7 +173,9 @@ class SpeechStream {
       };
       this.#stt.emit("metrics_collected", metrics);
     }
-    this.output.close();
+    if (!this.output.closed) {
+      this.output.close();
+    }
   }
   updateInputStream(audioStream) {
     this.deferredInputStream.setSource(audioStream);
@@ -217,9 +230,9 @@ class SpeechStream {
   }
   /** Close both the input and output of the STT stream */
   close() {
-    this.input.close();
-    this.queue.close();
-    this.output.close();
+    if (!this.input.closed) this.input.close();
+    if (!this.queue.closed) this.queue.close();
+    if (!this.output.closed) this.output.close();
     this.closed = true;
   }
   [Symbol.asyncIterator]() {

package/dist/stt/stt.cjs.map

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

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+{"version":3,"file":"stt.d.ts","sourceRoot":"","sources":["../../src/stt/stt.ts"],"names":[],"mappings":";AAGA,OAAO,EAAE,KAAK,UAAU,EAAE,cAAc,EAAE,MAAM,mBAAmB,CAAC;AACpE,OAAO,KAAK,EAAE,iBAAiB,IAAI,YAAY,EAAE,MAAM,wBAAwB,CAAC;AAEhF,OAAO,KAAK,EAAE,cAAc,EAAE,MAAM,iBAAiB,CAAC;AAItD,OAAO,KAAK,EAAE,UAAU,EAAE,MAAM,oBAAoB,CAAC;AAErD,OAAO,EAAE,KAAK,iBAAiB,EAA+B,MAAM,aAAa,CAAC;AAClF,OAAO,KAAK,EAAE,WAAW,EAAE,MAAM,aAAa,CAAC;AAC/C,OAAO,EAAE,kBAAkB,EAA6B,MAAM,aAAa,CAAC;AAE5E,2CAA2C;AAC3C,oBAAY,eAAe;IACzB;;;;OAIG;IACH,eAAe,IAAI;IACnB;;OAEG;IACH,kBAAkB,IAAI;IACtB;;;OAGG;IACH,gBAAgB,IAAI;IACpB;;;OAGG;IACH,aAAa,IAAI;IACjB,mEAAmE;IACnE,iBAAiB,IAAI;IACrB;;;;OAIG;IACH,oBAAoB,IAAI;CACzB;AAED,mEAAmE;AACnE,MAAM,WAAW,UAAU;IACzB,QAAQ,EAAE,MAAM,CAAC;IACjB,IAAI,EAAE,MAAM,CAAC;IACb,SAAS,EAAE,MAAM,CAAC;IAClB,OAAO,EAAE,MAAM,CAAC;IAChB,UAAU,EAAE,MAAM,CAAC;CACpB;AAED,MAAM,WAAW,gBAAgB;IAC/B,aAAa,EAAE,MAAM,CAAC;CACvB;AAED,sDAAsD;AACtD,MAAM,WAAW,WAAW;IAC1B,IAAI,EAAE,eAAe,CAAC;IACtB,YAAY,CAAC,EAAE,CAAC,UAAU,EAAE,GAAG,UAAU,EAAE,CAAC,CAAC;IAC7C,SAAS,CAAC,EAAE,MAAM,CAAC;IACnB,gBAAgB,CAAC,EAAE,gBAAgB,CAAC;CACrC;AAED;;;;;GAKG;AACH,MAAM,WAAW,eAAe;IAC9B,SAAS,EAAE,OAAO,CAAC;IACnB,cAAc,EAAE,OAAO,CAAC;CACzB;AAED,MAAM,WAAW,QAAQ;IACvB,IAAI,EAAE,WAAW,CAAC;IAClB,SAAS,EAAE,MAAM,CAAC;IAClB,KAAK,EAAE,MAAM,CAAC;IACd,KAAK,EAAE,KAAK,CAAC;IACb,WAAW,EAAE,OAAO,CAAC;CACtB;AAED,MAAM,MAAM,YAAY,GAAG;IACzB,CAAC,mBAAmB,CAAC,EAAE,CAAC,OAAO,EAAE,UAAU,KAAK,IAAI,CAAC;IACrD,CAAC,OAAO,CAAC,EAAE,CAAC,KAAK,EAAE,QAAQ,KAAK,IAAI,CAAC;CACtC,CAAC;kCAS2D,aAAa,YAAY,CAAC;AAPvF;;;;;;GAMG;AACH,8BAAsB,GAAI,SAAQ,QAAsD;;IACtF,QAAQ,CAAC,KAAK,EAAE,MAAM,CAAC;gBAGX,YAAY,EAAE,eAAe;IAKzC,sCAAsC;IACtC,IAAI,YAAY,IAAI,eAAe,CAElC;IAED,8FAA8F;IACxF,SAAS,CAAC,KAAK,EAAE,WAAW,GAAG,OAAO,CAAC,WAAW,CAAC;IAezD,SAAS,CAAC,QAAQ,CAAC,UAAU,CAAC,KAAK,EAAE,WAAW,GAAG,OAAO,CAAC,WAAW,CAAC;IAEvE;;;OAGG;IACH,QAAQ,CAAC,MAAM,IAAI,YAAY;IAEzB,KAAK,IAAI,OAAO,CAAC,IAAI,CAAC;CAG7B;AAED;;;;;;;;;;;;;;;GAeG;AACH,8BAAsB,YAAa,YAAW,qBAAqB,CAAC,WAAW,CAAC;;IAC9E,SAAS,CAAC,MAAM,CAAC,QAAQ,CAAC,cAAc,gBAA4B;IACpE,SAAS,CAAC,KAAK,sEAA6E;IAC5F,SAAS,CAAC,MAAM,kCAAyC;IACzD,SAAS,CAAC,KAAK,kCAAyC;IACxD,SAAS,CAAC,gBAAgB,CAAC,EAAE,MAAM,CAAC;IACpC,SAAS,CAAC,SAAS,CAAC,EAAE,cAAc,CAAC;IACrC,QAAQ,CAAC,KAAK,EAAE,MAAM,CAAC;IACvB,SAAS,CAAC,MAAM,UAAS;IAEzB,OAAO,CAAC,mBAAmB,CAAqC;IAChE,OAAO,CAAC,MAAM,CAAS;IACvB,OAAO,CAAC,YAAY,CAAoB;gBAGtC,GAAG,EAAE,GAAG,EACR,UAAU,CAAC,EAAE,MAAM,EACnB,iBAAiB,GAAE,iBAA+C;YAgBtD,QAAQ;IAqCtB,OAAO,CAAC,SAAS;cAUD,SAAS;cAkBT,cAAc;IA+B9B,SAAS,CAAC,QAAQ,CAAC,GAAG,IAAI,OAAO,CAAC,IAAI,CAAC;IAEvC,iBAAiB,CAAC,WAAW,EAAE,cAAc,CAAC,UAAU,CAAC;IAIzD,iBAAiB;IAIjB,qCAAqC;IACrC,SAAS,CAAC,KAAK,EAAE,UAAU;IAwB3B,4DAA4D;IAC5D,KAAK;IAUL,2DAA2D;IAC3D,QAAQ;IAUR,IAAI,IAAI,OAAO,CAAC,cAAc,CAAC,WAAW,CAAC,CAAC;IAI5C,wDAAwD;IACxD,KAAK;IAOL,CAAC,MAAM,CAAC,aAAa,CAAC,IAAI,YAAY;CAGvC"}

package/dist/stt/stt.js

@@ -124,7 +124,18 @@ class SpeechStream {
   }
   async monitorMetrics() {
     for await (const event of this.queue) {
-      this.output.put(event);
+      if (!this.output.closed) {
+        try {
+          this.output.put(event);
+        } catch (e) {
+          if (e instanceof Error && e.message.includes("Queue is closed")) {
+            this.logger.warn(
+              { err: e },
+              "Queue closed during transcript processing (expected during disconnect)"
+            );
+          }
+        }
+      }
       if (event.type !== 4 /* RECOGNITION_USAGE */) continue;
       const metrics = {
         type: "stt_metrics",
@@ -137,7 +148,9 @@ class SpeechStream {
       };
       this.#stt.emit("metrics_collected", metrics);
     }
-    this.output.close();
+    if (!this.output.closed) {
+      this.output.close();
+    }
   }
   updateInputStream(audioStream) {
     this.deferredInputStream.setSource(audioStream);
@@ -192,9 +205,9 @@ class SpeechStream {
   }
   /** Close both the input and output of the STT stream */
   close() {
-    this.input.close();
-    this.queue.close();
-    this.output.close();
+    if (!this.input.closed) this.input.close();
+    if (!this.queue.closed) this.queue.close();
+    if (!this.output.closed) this.output.close();
     this.closed = true;
   }
   [Symbol.asyncIterator]() {

package/dist/stt/stt.js.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n  protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   */\n  abstract stream(): SpeechStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().then(() => this.queue.close()));\n  }\n\n  private async mainTask() {\n    for (let i = 0; i < this._connOptions.maxRetry + 1; i++) {\n      try {\n        return await this.run();\n      } catch (error) {\n        if (error instanceof APIError) {\n          const retryInterval = this._connOptions._intervalForRetry(i);\n\n          if (this._connOptions.maxRetry === 0 || !error.retryable) {\n            this.emitError({ error, recoverable: false });\n            throw error;\n          } else if (i === this._connOptions.maxRetry) {\n            this.emitError({ error, recoverable: false });\n            throw new APIConnectionError({\n              message: `failed to recognize speech after ${this._connOptions.maxRetry + 1} attempts`,\n              options: { retryable: false },\n            });\n          } else {\n            // Don't emit error event for recoverable errors during retry loop\n            // to avoid ERR_UNHANDLED_ERROR or premature session termination\n            this.logger.warn(\n              { tts: this.#stt.label, attempt: i + 1, error },\n              `failed to recognize speech, retrying in ${retryInterval}s`,\n            );\n          }\n\n          if (retryInterval > 0) {\n            await delay(retryInterval);\n          }\n        } else {\n          this.emitError({ error: toError(error), recoverable: false });\n          throw error;\n        }\n      }\n    }\n  }\n\n  private emitError({ error, recoverable }: { error: Error; recoverable: boolean }) {\n    this.#stt.emit('error', {\n      type: 'stt_error',\n      timestamp: Date.now(),\n      label: this.#stt.label,\n      error,\n      recoverable,\n    });\n  }\n\n  protected async pumpInput() {\n    // TODO(AJS-35): Implement STT with webstreams API\n    const inputStream = this.deferredInputStream.stream;\n    const reader = inputStream.getReader();\n\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        this.pushFrame(value);\n      }\n    } catch (error) {\n      this.logger.error('Error in STTStream mainTask:', error);\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    for await (const event of this.queue) {\n      this.output.put(event);\n      if (event.type !== SpeechEventType.RECOGNITION_USAGE) continue;\n      const metrics: STTMetrics = {\n        type: 'stt_metrics',\n        timestamp: Date.now(),\n        requestId: event.requestId!,\n        durationMs: 0,\n        label: this.#stt.label,\n        audioDurationMs: Math.round(event.recognitionUsage!.audioDuration * 1000),\n        streamed: true,\n      };\n      this.#stt.emit('metrics_collected', metrics);\n    }\n    this.output.close();\n  }\n\n  protected abstract run(): Promise<void>;\n\n  updateInputStream(audioStream: ReadableStream<AudioFrame>) {\n    this.deferredInputStream.setSource(audioStream);\n  }\n\n  detachInputStream() {\n    this.deferredInputStream.detachSource();\n  }\n\n  /** Push an audio frame to the STT */\n  pushFrame(frame: AudioFrame) {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n\n    if (this.neededSampleRate && frame.sampleRate !== this.neededSampleRate) {\n      if (!this.resampler) {\n        this.resampler = new AudioResampler(frame.sampleRate, this.neededSampleRate);\n      }\n    }\n\n    if (this.resampler) {\n      const frames = this.resampler.push(frame);\n      for (const frame of frames) {\n        this.input.put(frame);\n      }\n    } else {\n      this.input.put(frame);\n    }\n  }\n\n  /** Flush the STT, causing it to process all pending text */\n  flush() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.put(SpeechStream.FLUSH_SENTINEL);\n  }\n\n  /** Mark the input as ended and forbid additional pushes */\n  endInput() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.close();\n  }\n\n  next(): Promise<IteratorResult<SpeechEvent>> {\n    return this.output.next();\n  }\n\n  /** Close both the input and output of the STT stream */\n  close() {\n    this.input.close();\n    this.queue.close();\n    this.output.close();\n    this.closed = true;\n  }\n\n  [Symbol.asyncIterator](): SpeechStream {\n    return this;\n  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+{"version":3,"sources":["../../src/stt/stt.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { type AudioFrame, AudioResampler } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type { ReadableStream } from 'node:stream/web';\nimport { APIConnectionError, APIError } from '../_exceptions.js';\nimport { calculateAudioDurationSeconds } from '../audio.js';\nimport { log } from '../log.js';\nimport type { STTMetrics } from '../metrics/base.js';\nimport { DeferredReadableStream } from '../stream/deferred_stream.js';\nimport { type APIConnectOptions, DEFAULT_API_CONNECT_OPTIONS } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\n\n/** Indicates start/middle/end of speech */\nexport enum SpeechEventType {\n  /**\n   * Indicate the start of speech.\n   * If the STT doesn't support this event, this will be emitted at the same time\n   * as the first INTERIM_TRANSCRIPT.\n   */\n  START_OF_SPEECH = 0,\n  /**\n   * Interim transcript, useful for real-time transcription.\n   */\n  INTERIM_TRANSCRIPT = 1,\n  /**\n   * Final transcript, emitted when the STT is confident enough that a certain\n   * portion of the speech will not change.\n   */\n  FINAL_TRANSCRIPT = 2,\n  /**\n   * Indicate the end of speech, emitted when the user stops speaking.\n   * The first alternative is a combination of all the previous FINAL_TRANSCRIPT events.\n   */\n  END_OF_SPEECH = 3,\n  /** Usage event, emitted periodically to indicate usage metrics. */\n  RECOGNITION_USAGE = 4,\n  /**\n   * Preflight transcript, emitted before final transcript when STT has high confidence\n   * but hasn't fully committed yet. Includes all pre-committed transcripts including\n   * final transcript from the previous STT run.\n   */\n  PREFLIGHT_TRANSCRIPT = 5,\n}\n\n/** SpeechData contains metadata about this {@link SpeechEvent}. */\nexport interface SpeechData {\n  language: string;\n  text: string;\n  startTime: number;\n  endTime: number;\n  confidence: number;\n}\n\nexport interface RecognitionUsage {\n  audioDuration: number;\n}\n\n/** SpeechEvent is a packet of speech-to-text data. */\nexport interface SpeechEvent {\n  type: SpeechEventType;\n  alternatives?: [SpeechData, ...SpeechData[]];\n  requestId?: string;\n  recognitionUsage?: RecognitionUsage;\n}\n\n/**\n * Describes the capabilities of the STT provider.\n *\n * @remarks\n * At present, the framework only supports providers that have a streaming endpoint.\n */\nexport interface STTCapabilities {\n  streaming: boolean;\n  interimResults: boolean;\n}\n\nexport interface STTError {\n  type: 'stt_error';\n  timestamp: number;\n  label: string;\n  error: Error;\n  recoverable: boolean;\n}\n\nexport type STTCallbacks = {\n  ['metrics_collected']: (metrics: STTMetrics) => void;\n  ['error']: (error: STTError) => void;\n};\n\n/**\n * An instance of a speech-to-text adapter.\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child STT class, which inherits this class's methods.\n */\nexport abstract class STT extends (EventEmitter as new () => TypedEmitter<STTCallbacks>) {\n  abstract label: string;\n  #capabilities: STTCapabilities;\n\n  constructor(capabilities: STTCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  /** Returns this STT's capabilities */\n  get capabilities(): STTCapabilities {\n    return this.#capabilities;\n  }\n\n  /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */\n  async recognize(frame: AudioBuffer): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame);\n    const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1000000));\n    this.emit('metrics_collected', {\n      type: 'stt_metrics',\n      requestId: event.requestId ?? '',\n      timestamp: Date.now(),\n      durationMs,\n      label: this.label,\n      audioDurationMs: Math.round(calculateAudioDurationSeconds(frame) * 1000),\n      streamed: false,\n    });\n    return event;\n  }\n  protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   */\n  abstract stream(): SpeechStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\n/**\n * An instance of a speech-to-text stream, as an asynchronous iterable iterator.\n *\n * @example Looping through frames\n * ```ts\n * for await (const event of stream) {\n *   if (event.type === SpeechEventType.FINAL_TRANSCRIPT) {\n *     console.log(event.alternatives[0].text)\n *   }\n * }\n * ```\n *\n * @remarks\n * This class is abstract, and as such cannot be used directly. Instead, use a provider plugin that\n * exports its own child SpeechStream class, which inherits this class's methods.\n */\nexport abstract class SpeechStream implements AsyncIterableIterator<SpeechEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new AsyncIterableQueue<AudioFrame | typeof SpeechStream.FLUSH_SENTINEL>();\n  protected output = new AsyncIterableQueue<SpeechEvent>();\n  protected queue = new AsyncIterableQueue<SpeechEvent>();\n  protected neededSampleRate?: number;\n  protected resampler?: AudioResampler;\n  abstract label: string;\n  protected closed = false;\n  #stt: STT;\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n  private logger = log();\n  private _connOptions: APIConnectOptions;\n\n  constructor(\n    stt: STT,\n    sampleRate?: number,\n    connectionOptions: APIConnectOptions = DEFAULT_API_CONNECT_OPTIONS,\n  ) {\n    this.#stt = stt;\n    this._connOptions = connectionOptions;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n    this.neededSampleRate = sampleRate;\n    this.monitorMetrics();\n    this.pumpInput();\n\n    // this is a hack to immitate asyncio.create_task so that mainTask\n    // is run **after** the constructor has finished. Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().then(() => this.queue.close()));\n  }\n\n  private async mainTask() {\n    for (let i = 0; i < this._connOptions.maxRetry + 1; i++) {\n      try {\n        return await this.run();\n      } catch (error) {\n        if (error instanceof APIError) {\n          const retryInterval = this._connOptions._intervalForRetry(i);\n\n          if (this._connOptions.maxRetry === 0 || !error.retryable) {\n            this.emitError({ error, recoverable: false });\n            throw error;\n          } else if (i === this._connOptions.maxRetry) {\n            this.emitError({ error, recoverable: false });\n            throw new APIConnectionError({\n              message: `failed to recognize speech after ${this._connOptions.maxRetry + 1} attempts`,\n              options: { retryable: false },\n            });\n          } else {\n            // Don't emit 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inputStream.getReader();\n\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        this.pushFrame(value);\n      }\n    } catch (error) {\n      this.logger.error('Error in STTStream mainTask:', error);\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    for await (const event of this.queue) {\n      if (!this.output.closed) {\n        try {\n          this.output.put(event);\n        } catch (e) {\n          if (e instanceof Error && e.message.includes('Queue is closed')) {\n            this.logger.warn(\n              { err: e },\n              'Queue closed during transcript processing (expected during disconnect)',\n            );\n          }\n        }\n      }\n      if (event.type !== SpeechEventType.RECOGNITION_USAGE) continue;\n      const metrics: STTMetrics = {\n        type: 'stt_metrics',\n        timestamp: Date.now(),\n        requestId: 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this.neededSampleRate);\n      }\n    }\n\n    if (this.resampler) {\n      const frames = this.resampler.push(frame);\n      for (const frame of frames) {\n        this.input.put(frame);\n      }\n    } else {\n      this.input.put(frame);\n    }\n  }\n\n  /** Flush the STT, causing it to process all pending text */\n  flush() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.put(SpeechStream.FLUSH_SENTINEL);\n  }\n\n  /** Mark the input as ended and forbid additional pushes */\n  endInput() {\n    if (this.input.closed) {\n      throw new Error('Input is closed');\n    }\n    if (this.closed) {\n      throw new Error('Stream is closed');\n    }\n    this.input.close();\n  }\n\n  next(): Promise<IteratorResult<SpeechEvent>> {\n    return this.output.next();\n  }\n\n  /** Close both the input and output of the STT stream */\n  close() {\n    if (!this.input.closed) this.input.close();\n    if (!this.queue.closed) this.queue.close();\n    if (!this.output.closed) this.output.close();\n    this.closed = true;\n  }\n\n  [Symbol.asyncIterator](): SpeechStream {\n    return this;\n  }\n}\n"],"mappings":"AAGA,SAA0B,sBAAsB;AAEhD,SAAS,oBAAoB;AAE7B,SAAS,oBAAoB,gBAAgB;AAC7C,SAAS,qCAAqC;AAC9C,SAAS,WAAW;AAEpB,SAAS,8BAA8B;AACvC,SAAiC,mCAAmC;AAEpE,SAAS,oBAAoB,OAAO,WAAW,eAAe;AAGvD,IAAK,kBAAL,kBAAKA,qBAAL;AAML,EAAAA,kCAAA,qBAAkB,KAAlB;AAIA,EAAAA,kCAAA,wBAAqB,KAArB;AAKA,EAAAA,kCAAA,sBAAmB,KAAnB;AAKA,EAAAA,kCAAA,mBAAgB,KAAhB;AAEA,EAAAA,kCAAA,uBAAoB,KAApB;AAMA,EAAAA,kCAAA,0BAAuB,KAAvB;AA5BU,SAAAA;AAAA,GAAA;AAmFL,MAAe,YAAa,aAAsD;AAAA,EAEvF;AAAA,EAEA,YAAY,cAA+B;AACzC,UAAM;AACN,SAAK,gBAAgB;AAAA,EACvB;AAAA;AAAA,EAGA,IAAI,eAAgC;AAClC,WAAO,KAAK;AAAA,EACd;AAAA;AAAA,EAGA,MAAM,UAAU,OAA0C;AACxD,UAAM,YAAY,QAAQ,OAAO,OAAO;AACxC,UAAM,QAAQ,MAAM,KAAK,WAAW,KAAK;AACzC,UAAM,aAAa,QAAQ,QAAQ,OAAO,OAAO,IAAI,aAAa,OAAO,GAAO,CAAC;AACjF,SAAK,KAAK,qBAAqB;AAAA,MAC7B,MAAM;AAAA,MACN,WAAW,MAAM,aAAa;AAAA,MAC9B,WAAW,KAAK,IAAI;AAAA,MACpB;AAAA,MACA,OAAO,KAAK;AAAA,MACZ,iBAAiB,KAAK,MAAM,8BAA8B,KAAK,IAAI,GAAI;AAAA,MACvE,UAAU;AAAA,IACZ,CAAC;AACD,WAAO;AAAA,EACT;AAAA,EASA,MAAM,QAAuB;AAC3B;AAAA,EACF;AACF;AAkBO,MAAe,aAA2D;AAAA,EAC/E,OAA0B,iBAAiB,OAAO,gBAAgB;AAAA,EACxD,QAAQ,IAAI,mBAAoE;AAAA,EAChF,SAAS,IAAI,mBAAgC;AAAA,EAC7C,QAAQ,IAAI,mBAAgC;AAAA,EAC5C;AAAA,EACA;AAAA,EAEA,SAAS;AAAA,EACnB;AAAA,EACQ;AAAA,EACA,SAAS,IAAI;AAAA,EACb;AAAA,EAER,YACE,KACA,YACA,oBAAuC,6BACvC;AACA,SAAK,OAAO;AACZ,SAAK,eAAe;AACpB,SAAK,sBAAsB,IAAI,uBAAmC;AAClE,SAAK,mBAAmB;AACxB,SAAK,eAAe;AACpB,SAAK,UAAU;AAMf,cAAU,MAAM,KAAK,SAAS,EAAE,KAAK,MAAM,KAAK,MAAM,MAAM,CAAC,CAAC;AAAA,EAChE;AAAA,EAEA,MAAc,WAAW;AACvB,aAAS,IAAI,GAAG,IAAI,KAAK,aAAa,WAAW,GAAG,KAAK;AACvD,UAAI;AACF,eAAO,MAAM,KAAK,IAAI;AAAA,MACxB,SAAS,OAAO;AACd,YAAI,iBAAiB,UAAU;AAC7B,gBAAM,gBAAgB,KAAK,aAAa,kBAAkB,CAAC;AAE3D,cAAI,KAAK,aAAa,aAAa,KAAK,CAAC,MAAM,WAAW;AACxD,iBAAK,UAAU,EAAE,OAAO,aAAa,MAAM,CAAC;AAC5C,kBAAM;AAAA,UACR,WAAW,MAAM,KAAK,aAAa,UAAU;AAC3C,iBAAK,UAAU,EAAE,OAAO,aAAa,MAAM,CAAC;AAC5C,kBAAM,IAAI,mBAAmB;AAAA,cAC3B,SAAS,oCAAoC,KAAK,aAAa,WAAW,CAAC;AAAA,cAC3E,SAAS,EAAE,WAAW,MAAM;AAAA,YAC9B,CAAC;AAAA,UACH,OAAO;AAGL,iBAAK,OAAO;AAAA,cACV,EAAE,KAAK,KAAK,KAAK,OAAO,SAAS,IAAI,GAAG,MAAM;AAAA,cAC9C,2CAA2C,aAAa;AAAA,YAC1D;AAAA,UACF;AAEA,cAAI,gBAAgB,GAAG;AACrB,kBAAM,MAAM,aAAa;AAAA,UAC3B;AAAA,QACF,OAAO;AACL,eAAK,UAAU,EAAE,OAAO,QAAQ,KAAK,GAAG,aAAa,MAAM,CAAC;AAC5D,gBAAM;AAAA,QACR;AAAA,MACF;AAAA,IACF;AAAA,EACF;AAAA,EAEQ,UAAU,EAAE,OAAO,YAAY,GAA2C;AAChF,SAAK,KAAK,KAAK,SAAS;AAAA,MACtB,MAAM;AAAA,MACN,WAAW,KAAK,IAAI;AAAA,MACpB,OAAO,KAAK,KAAK;AAAA,MACjB;AAAA,MACA;AAAA,IACF,CAAC;AAAA,EACH;AAAA,EAEA,MAAgB,YAAY;AAE1B,UAAM,cAAc,KAAK,oBAAoB;AAC7C,UAAM,SAAS,YAAY,UAAU;AAErC,QAAI;AACF,aAAO,MAAM;AACX,cAAM,EAAE,MAAM,MAAM,IAAI,MAAM,OAAO,KAAK;AAC1C,YAAI,KAAM;AACV,aAAK,UAAU,KAAK;AAAA,MACtB;AAAA,IACF,SAAS,OAAO;AACd,WAAK,OAAO,MAAM,gCAAgC,KAAK;AAAA,IACzD,UAAE;AACA,aAAO,YAAY;AAAA,IACrB;AAAA,EACF;AAAA,EAEA,MAAgB,iBAAiB;AAC/B,qBAAiB,SAAS,KAAK,OAAO;AACpC,UAAI,CAAC,KAAK,OAAO,QAAQ;AACvB,YAAI;AACF,eAAK,OAAO,IAAI,KAAK;AAAA,QACvB,SAAS,GAAG;AACV,cAAI,aAAa,SAAS,EAAE,QAAQ,SAAS,iBAAiB,GAAG;AAC/D,iBAAK,OAAO;AAAA,cACV,EAAE,KAAK,EAAE;AAAA,cACT;AAAA,YACF;AAAA,UACF;AAAA,QACF;AAAA,MACF;AACA,UAAI,MAAM,SAAS,0BAAmC;AACtD,YAAM,UAAsB;AAAA,QAC1B,MAAM;AAAA,QACN,WAAW,KAAK,IAAI;AAAA,QACpB,WAAW,MAAM;AAAA,QACjB,YAAY;AAAA,QACZ,OAAO,KAAK,KAAK;AAAA,QACjB,iBAAiB,KAAK,MAAM,MAAM,iBAAkB,gBAAgB,GAAI;AAAA,QACxE,UAAU;AAAA,MACZ;AACA,WAAK,KAAK,KAAK,qBAAqB,OAAO;AAAA,IAC7C;AACA,QAAI,CAAC,KAAK,OAAO,QAAQ;AACvB,WAAK,OAAO,MAAM;AAAA,IACpB;AAAA,EACF;AAAA,EAIA,kBAAkB,aAAyC;AACzD,SAAK,oBAAoB,UAAU,WAAW;AAAA,EAChD;AAAA,EAEA,oBAAoB;AAClB,SAAK,oBAAoB,aAAa;AAAA,EACxC;AAAA;AAAA,EAGA,UAAU,OAAmB;AAC3B,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AAEA,QAAI,KAAK,oBAAoB,MAAM,eAAe,KAAK,kBAAkB;AACvE,UAAI,CAAC,KAAK,WAAW;AACnB,aAAK,YAAY,IAAI,eAAe,MAAM,YAAY,KAAK,gBAAgB;AAAA,MAC7E;AAAA,IACF;AAEA,QAAI,KAAK,WAAW;AAClB,YAAM,SAAS,KAAK,UAAU,KAAK,KAAK;AACxC,iBAAWC,UAAS,QAAQ;AAC1B,aAAK,MAAM,IAAIA,MAAK;AAAA,MACtB;AAAA,IACF,OAAO;AACL,WAAK,MAAM,IAAI,KAAK;AAAA,IACtB;AAAA,EACF;AAAA;AAAA,EAGA,QAAQ;AACN,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AACA,SAAK,MAAM,IAAI,aAAa,cAAc;AAAA,EAC5C;AAAA;AAAA,EAGA,WAAW;AACT,QAAI,KAAK,MAAM,QAAQ;AACrB,YAAM,IAAI,MAAM,iBAAiB;AAAA,IACnC;AACA,QAAI,KAAK,QAAQ;AACf,YAAM,IAAI,MAAM,kBAAkB;AAAA,IACpC;AACA,SAAK,MAAM,MAAM;AAAA,EACnB;AAAA,EAEA,OAA6C;AAC3C,WAAO,KAAK,OAAO,KAAK;AAAA,EAC1B;AAAA;AAAA,EAGA,QAAQ;AACN,QAAI,CAAC,KAAK,MAAM,OAAQ,MAAK,MAAM,MAAM;AACzC,QAAI,CAAC,KAAK,MAAM,OAAQ,MAAK,MAAM,MAAM;AACzC,QAAI,CAAC,KAAK,OAAO,OAAQ,MAAK,OAAO,MAAM;AAC3C,SAAK,SAAS;AAAA,EAChB;AAAA,EAEA,CAAC,OAAO,aAAa,IAAkB;AACrC,WAAO;AAAA,EACT;AACF;","names":["SpeechEventType","frame"]}

package/dist/tts/tts.cjs

@@ -124,7 +124,16 @@ class SynthesizeStream {
       recoverable
     });
   }
-  // TODO(AJS-37) Remove when refactoring TTS to use streams
+  // NOTE(AJS-37): The implementation below uses an AsyncIterableQueue (`this.input`)
+  // bridged from a DeferredReadableStream (`this.deferredInputStream`) rather than
+  // consuming the stream directly.
+  //
+  // A full refactor to native Web Streams was considered but is currently deferred.
+  // The primary reason is to maintain architectural parity with the Python SDK,
+  // which is a key design goal for the project. This ensures a consistent developer
+  // experience across both platforms.
+  //
+  // For more context, see the discussion in GitHub issue # 844.
   async pumpInput() {
     const reader = this.deferredInputStream.stream.getReader();
     try {
@@ -200,11 +209,8 @@ class SynthesizeStream {
       this.#monitorMetricsTask.finally(() => this.output.close());
     }
     this.#metricsText += text;
-    if (this.input.closed) {
-      throw new Error("Input is closed");
-    }
-    if (this.closed) {
-      throw new Error("Stream is closed");
+    if (this.input.closed || this.closed) {
+      return;
     }
     this.input.put(text);
   }
@@ -214,22 +220,16 @@ class SynthesizeStream {
       this.#metricsPendingTexts.push(this.#metricsText);
       this.#metricsText = "";
     }
-    if (this.input.closed) {
-      throw new Error("Input is closed");
-    }
-    if (this.closed) {
-      throw new Error("Stream is closed");
+    if (this.input.closed || this.closed) {
+      return;
     }
     this.input.put(SynthesizeStream.FLUSH_SENTINEL);
   }
   /** Mark the input as ended and forbid additional pushes */
   endInput() {
     this.flush();
-    if (this.input.closed) {
-      throw new Error("Input is closed");
-    }
-    if (this.closed) {
-      throw new Error("Stream is closed");
+    if (this.input.closed || this.closed) {
+      return;
     }
     this.input.close();
   }