@livekit/agents 1.0.41 → 1.0.43

package/dist/ipc/supervised_proc.d.ts

@@ -3,12 +3,19 @@ import type { ChildProcess } from 'node:child_process';
 import type { RunningJobInfo } from '../job.js';
 import { Future } from '../utils.js';
 export interface ProcOpts {
+    /** Timeout for process initialization in milliseconds. */
     initializeTimeout: number;
+    /** Timeout for process shutdown in milliseconds. */
     closeTimeout: number;
+    /** Memory usage warning threshold in megabytes. */
     memoryWarnMB: number;
+    /** Memory usage limit in megabytes. */
     memoryLimitMB: number;
+    /** Interval for health check pings in milliseconds. */
     pingInterval: number;
+    /** Timeout waiting for pong response in milliseconds. */
     pingTimeout: number;
+    /** Threshold for warning about unresponsive processes in milliseconds. */
     highPingThreshold: number;
 }
 export declare abstract class SupervisedProc {

package/dist/ipc/supervised_proc.d.ts.map

@@ -1 +1 @@
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package/dist/ipc/supervised_proc.js.map

@@ -1 +1 @@
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Error('runner already started');\n    } else if (this.#closing) {\n      throw new Error('runner is closed');\n    }\n\n    this.proc = this.createProcess();\n\n    this.#started = true;\n    this.run();\n  }\n\n  async run() {\n    await this.init.await;\n\n    this.#pingInterval = setInterval(() => {\n      if (this.proc?.connected) {\n        this.proc.send({ case: 'pingRequest', value: { timestamp: Date.now() } });\n      }\n    }, this.#opts.pingInterval);\n\n    this.#pongTimeout = setTimeout(() => {\n      this.#logger.warn('job is unresponsive');\n      clearTimeout(this.#pongTimeout);\n      clearInterval(this.#pingInterval);\n      this.proc!.kill();\n      this.#join.resolve();\n    }, this.#opts.pingTimeout);\n\n    this.#memoryMonitorInterval = setInterval(async () => {\n      const memoryMB = await this.getChildMemoryUsageMB();\n      if (this.#opts.memoryLimitMB > 0 && memoryMB > this.#opts.memoryLimitMB) {\n        this.#logger\n          .child({ memoryUsageMB: memoryMB, memoryLimitMB: this.#opts.memoryLimitMB })\n          .error('process exceeded memory limit, killing process');\n        this.close();\n      } else if (this.#opts.memoryWarnMB > 0 && memoryMB > this.#opts.memoryWarnMB) {\n        this.#logger\n          .child({\n            memoryUsageMB: memoryMB,\n            memoryWarnMB: this.#opts.memoryWarnMB,\n            memoryLimitMB: this.#opts.memoryLimitMB,\n          })\n          .warn('process memory usage is high');\n      }\n    }, 5000);\n\n    const listener = (msg: IPCMessage) => {\n      switch (msg.case) {\n        case 'pongResponse': {\n          const delay = Date.now() - msg.value.timestamp;\n          if (delay > this.#opts.highPingThreshold) {\n            this.#logger.child({ delay }).warn('job executor is unresponsive');\n          }\n          this.#pongTimeout?.refresh();\n          break;\n        }\n        case 'exiting': {\n          this.#logger.child({ reason: msg.value.reason }).debug('job exiting');\n 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clearTimeout(this.#pongTimeout);\n      clearInterval(this.#pingInterval);\n      this.proc!.kill();\n      this.#join.resolve();\n    }, this.#opts.pingTimeout);\n\n    this.#memoryMonitorInterval = setInterval(async () => {\n      const memoryMB = await this.getChildMemoryUsageMB();\n      if (this.#opts.memoryLimitMB > 0 && memoryMB > this.#opts.memoryLimitMB) {\n        this.#logger\n          .child({ memoryUsageMB: memoryMB, memoryLimitMB: this.#opts.memoryLimitMB })\n          .error('process exceeded memory limit, killing process');\n        this.close();\n      } else if (this.#opts.memoryWarnMB > 0 && memoryMB > this.#opts.memoryWarnMB) {\n        this.#logger\n          .child({\n            memoryUsageMB: memoryMB,\n            memoryWarnMB: this.#opts.memoryWarnMB,\n            memoryLimitMB: this.#opts.memoryLimitMB,\n          })\n          .warn('process memory usage is high');\n      }\n    }, 5000);\n\n    const listener = (msg: IPCMessage) => {\n      switch 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Error('first message must be InitializeResponse');\n      }\n    });\n    this.init.resolve();\n  }\n\n  async close() {\n    if (!this.#started) {\n      return;\n    }\n    this.#closing = true;\n\n    if (this.proc?.connected) {\n      this.proc.send({ case: 'shutdownRequest' });\n    }\n\n    const timer = setTimeout(() => {\n      this.#logger.error('job shutdown is taking too much time');\n      this.proc!.kill();\n    }, this.#opts.closeTimeout);\n    await this.#join.await.then(() => {\n      clearTimeout(timer);\n      this.clearTimers();\n    });\n  }\n\n  async launchJob(info: RunningJobInfo) {\n    if (this.#runningJob) {\n      throw new Error('executor already has a running job');\n    }\n    if (!this.proc?.connected) {\n      throw new Error('process not connected');\n    }\n    this.#runningJob = info;\n    this.proc.send({ case: 'startJobRequest', value: { runningJob: info } });\n  }\n\n  private async getChildMemoryUsageMB(): Promise<number> {\n    const pid = 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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, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\nimport type { TimedString } from '../voice/index.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  words?: TimedString[];\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   * Whether this STT supports aligned transcripts with word/chunk timestamps.\n   * - 'word': Provider returns word-level timestamps\n   * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)\n   * - false: Provider does not support aligned transcripts\n   */\n  alignedTranscript?: 'word' | 'chunk' | false;\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, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\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\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): 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. 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Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().finally(() => 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 = intervalForRetry(this._connOptions, 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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+{"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, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\nimport type { TimedString } from '../voice/index.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 code of the speech. */\n  language: string;\n  /** Transcribed text. */\n  text: string;\n  /** Start time of the speech segment in seconds. */\n  startTime: number;\n  /** End time of the speech segment in seconds. */\n  endTime: number;\n  /** Confidence score of the transcription (0-1). */\n  confidence: number;\n  /** Word-level timing information. */\n  words?: TimedString[];\n}\n\nexport interface RecognitionUsage {\n  /** Duration of the audio that was recognized in seconds. */\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   * Whether this STT supports aligned transcripts with word/chunk timestamps.\n   * - 'word': Provider returns word-level timestamps\n   * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)\n   * - false: Provider does not support aligned transcripts\n   */\n  alignedTranscript?: 'word' | 'chunk' | false;\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, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\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\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): 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. 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Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().finally(() => 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 = intervalForRetry(this._connOptions, 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: 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    if (!this.output.closed) {\n      this.output.close();\n    }\n  }\n\n  protected abstract run(): Promise<void>;\n\n  protected get abortSignal(): AbortSignal {\n    return this.abortController.signal;\n  }\n\n  get startTimeOffset(): number {\n    return this._startTimeOffset;\n  }\n\n  set startTimeOffset(value: number) {\n    if (value < 0) {\n      throw new Error('startTimeOffset must be non-negative');\n    }\n    this._startTimeOffset = value;\n  }\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    if (!this.input.closed) this.input.close();\n    if (!this.queue.closed) this.queue.close();\n    if (!this.output.closed) this.output.close();\n    if (!this.abortController.signal.aborted) this.abortController.abort();\n    this.closed = true;\n  }\n\n  [Symbol.asyncIterator](): SpeechStream {\n    return this;\n  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package/dist/stt/stt.d.cts

@@ -40,14 +40,21 @@ export declare enum SpeechEventType {
 }
 /** SpeechData contains metadata about this {@link SpeechEvent}. */
 export interface SpeechData {
+    /** Language code of the speech. */
     language: string;
+    /** Transcribed text. */
     text: string;
+    /** Start time of the speech segment in seconds. */
     startTime: number;
+    /** End time of the speech segment in seconds. */
     endTime: number;
+    /** Confidence score of the transcription (0-1). */
     confidence: number;
+    /** Word-level timing information. */
     words?: TimedString[];
 }
 export interface RecognitionUsage {
+    /** Duration of the audio that was recognized in seconds. */
     audioDuration: number;
 }
 /** SpeechEvent is a packet of speech-to-text data. */

package/dist/stt/stt.d.ts

@@ -40,14 +40,21 @@ export declare enum SpeechEventType {
 }
 /** SpeechData contains metadata about this {@link SpeechEvent}. */
 export interface SpeechData {
+    /** Language code of the speech. */
     language: string;
+    /** Transcribed text. */
     text: string;
+    /** Start time of the speech segment in seconds. */
     startTime: number;
+    /** End time of the speech segment in seconds. */
     endTime: number;
+    /** Confidence score of the transcription (0-1). */
     confidence: number;
+    /** Word-level timing information. */
     words?: TimedString[];
 }
 export interface RecognitionUsage {
+    /** Duration of the audio that was recognized in seconds. */
     audioDuration: number;
 }
 /** SpeechEvent is a packet of speech-to-text data. */

package/dist/stt/stt.d.ts.map

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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, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\nimport type { TimedString } from '../voice/index.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  words?: TimedString[];\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   * Whether this STT supports aligned transcripts with word/chunk timestamps.\n   * - 'word': Provider returns word-level timestamps\n   * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)\n   * - false: Provider does not support aligned transcripts\n   */\n  alignedTranscript?: 'word' | 'chunk' | false;\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, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\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\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): 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  private _startTimeOffset: number = 0;\n\n  protected abortController = new AbortController();\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().finally(() => 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 = intervalForRetry(this._connOptions, 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      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: 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    if (!this.output.closed) {\n      this.output.close();\n    }\n  }\n\n  protected abstract run(): Promise<void>;\n\n  protected get abortSignal(): AbortSignal {\n    return this.abortController.signal;\n  }\n\n  get startTimeOffset(): number {\n    return this._startTimeOffset;\n  }\n\n  set startTimeOffset(value: number) {\n    if (value < 0) {\n      throw new Error('startTimeOffset must be non-negative');\n    }\n    this._startTimeOffset = value;\n  }\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 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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    if (!this.abortController.signal.aborted) this.abortController.abort();\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, intervalForRetry } from '../types.js';\nimport type { AudioBuffer } from '../utils.js';\nimport { AsyncIterableQueue, delay, startSoon, toError } from '../utils.js';\nimport type { TimedString } from '../voice/index.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 code of the speech. */\n  language: string;\n  /** Transcribed text. */\n  text: string;\n  /** Start time of the speech segment in seconds. */\n  startTime: number;\n  /** End time of the speech segment in seconds. */\n  endTime: number;\n  /** Confidence score of the transcription (0-1). */\n  confidence: number;\n  /** Word-level timing information. */\n  words?: TimedString[];\n}\n\nexport interface RecognitionUsage {\n  /** Duration of the audio that was recognized in seconds. */\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   * Whether this STT supports aligned transcripts with word/chunk timestamps.\n   * - 'word': Provider returns word-level timestamps\n   * - 'chunk': Provider returns chunk-level timestamps (e.g., sentence/phrase boundaries)\n   * - false: Provider does not support aligned transcripts\n   */\n  alignedTranscript?: 'word' | 'chunk' | false;\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, abortSignal?: AbortSignal): Promise<SpeechEvent> {\n    const startTime = process.hrtime.bigint();\n    const event = await this._recognize(frame, abortSignal);\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\n  protected abstract _recognize(\n    frame: AudioBuffer,\n    abortSignal?: AbortSignal,\n  ): Promise<SpeechEvent>;\n\n  /**\n   * Returns a {@link SpeechStream} that can be used to push audio frames and receive\n   * transcriptions\n   *\n   * @param options - Optional configuration including connection options\n   */\n  abstract stream(options?: { connOptions?: APIConnectOptions }): 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. 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Otherwise we get\n    // runtime error when trying to access class variables in the\n    // `run` method.\n    startSoon(() => this.mainTask().finally(() => 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 = intervalForRetry(this._connOptions, 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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package/dist/transcription.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/transcription.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { TranscriptionSegment } from '@livekit/protocol';\nimport { AudioFrame } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport { basic } from './tokenize/index.js';\nimport type { SentenceStream, SentenceTokenizer } from './tokenize/tokenizer.js';\nimport { AsyncIterableQueue, Future, shortuuid } from './utils.js';\n\n// standard speech rate in hyphens/ms\nconst STANDARD_SPEECH_RATE = 3830;\n\nexport interface TextSyncOptions {\n  language: string;\n  speed: number;\n  newSentenceDelay: number;\n  sentenceTokenizer: SentenceTokenizer;\n  hyphenateWord: (word: string) => string[];\n  splitWords: (words: string) => [string, number, number][];\n}\n\nexport const defaultTextSyncOptions: TextSyncOptions = {\n  language: '',\n  speed: 1,\n  newSentenceDelay: 400,\n  sentenceTokenizer: new basic.SentenceTokenizer(),\n  hyphenateWord: basic.hyphenateWord,\n  splitWords: basic.splitWords,\n};\n\ninterface AudioData {\n  pushedDuration: number;\n  done: boolean;\n}\n\ninterface TextData {\n  sentenceStream: SentenceStream;\n  pushedText: string;\n  done: boolean;\n  forwardedHyphens: number;\n  forwardedSentences: number;\n}\n\ntype SyncCallbacks = {\n  textUpdated: (text: TranscriptionSegment) => void;\n};\n\nexport class TextAudioSynchronizer extends (EventEmitter as new () => TypedEmitter<SyncCallbacks>) {\n  #opts: TextSyncOptions;\n  #speed: number;\n\n  #closed = false;\n  #interrupted = false;\n  #closeFut = new Future();\n\n  #playingSegIndex = -1;\n  #finishedSegIndex = -1;\n\n  #textQChanged = new AsyncIterableQueue<number>();\n  #textQ: (TextData | undefined)[] = [];\n  #audioQChanged = new AsyncIterableQueue<number>();\n  #audioQ: (AudioData | undefined)[] = [];\n\n  #playedText = '';\n  #task?: Promise<void>;\n\n  #audioData?: AudioData;\n  #textData?: TextData;\n\n  constructor(opts: TextSyncOptions) {\n    super();\n\n    this.#opts = opts;\n    this.#speed = opts.speed * STANDARD_SPEECH_RATE;\n  }\n\n  pushAudio(frame: AudioFrame) {\n    this.#checkNotClosed();\n    if (!this.#audioData) {\n      this.#audioData = { pushedDuration: 0, done: false };\n      this.#audioQ.push(this.#audioData);\n      this.#audioQChanged.put(1);\n    }\n    this.#audioData.pushedDuration += frame.samplesPerChannel / frame.sampleRate;\n  }\n\n  pushText(text: string) {\n    this.#checkNotClosed();\n    if (!this.#textData) {\n      this.#textData = {\n        sentenceStream: this.#opts.sentenceTokenizer.stream(),\n        pushedText: '',\n        done: false,\n        forwardedHyphens: 0,\n        forwardedSentences: 0,\n      };\n      this.#textQ.push(this.#textData);\n      this.#textQChanged.put(1);\n    }\n\n    this.#textData.pushedText += text;\n    this.#textData.sentenceStream.pushText(text);\n  }\n\n  markAudioSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#audioData) {\n      // create empty audio data if none exists\n      this.pushAudio(new AudioFrame(new Int16Array(), 24000, 1, 0));\n    }\n\n    this.#audioData!.done = true;\n    this.#audioData = undefined;\n  }\n\n  markTextSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#textData) {\n      this.pushText('');\n    }\n\n    this.#textData!.done = true;\n    this.#textData?.sentenceStream.flush();\n    this.#textData?.sentenceStream.close();\n    this.#textData = undefined;\n  }\n\n  segmentPlayoutStarted() {\n    this.#checkNotClosed();\n    this.#playingSegIndex++;\n\n    if (!this.#task) {\n      this.#task = this.#mainLoop();\n    }\n  }\n\n  segmentPlayoutFinished() {\n    this.#checkNotClosed();\n    this.#finishedSegIndex++;\n  }\n\n  get playedText(): string {\n    return this.#playedText;\n  }\n\n  async close(interrupt: boolean) {\n    if (this.#closed) {\n      return;\n    }\n    this.#closed = true;\n    this.#interrupted = interrupt;\n    this.#closeFut.resolve();\n\n    for (const textData of this.#textQ) {\n      textData?.sentenceStream.close();\n    }\n\n    this.#textQ.push(undefined);\n    this.#audioQ.push(undefined);\n    this.#textQChanged.put(1);\n    this.#audioQChanged.put(1);\n\n    await this.#task;\n  }\n\n  async #mainLoop() {\n    let segIndex = 0;\n    let qDone = false;\n\n    while (!qDone) {\n      await this.#textQChanged.next();\n      await this.#audioQChanged.next();\n\n      while (this.#textQ.length && this.#audioQ.length) {\n        const textData = this.#textQ.pop();\n        const audioData = this.#audioQ.pop();\n\n        if (!(textData && audioData)) {\n          qDone = true;\n          break;\n        }\n\n        // wait for segment to start playing\n        while (!this.#closed) {\n          if (this.#playingSegIndex >= segIndex) break;\n          await 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+{"version":3,"sources":["../src/transcription.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { TranscriptionSegment } from '@livekit/protocol';\nimport { AudioFrame } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport { basic } from './tokenize/index.js';\nimport type { SentenceStream, SentenceTokenizer } from './tokenize/tokenizer.js';\nimport { AsyncIterableQueue, Future, shortuuid } from './utils.js';\n\n// standard speech rate in hyphens/ms\nconst STANDARD_SPEECH_RATE = 3830;\n\nexport interface TextSyncOptions {\n  /** Language code for transcription. */\n  language: string;\n  /** Speech speed multiplier. */\n  speed: number;\n  /** Delay between sentences in milliseconds. */\n  newSentenceDelay: number;\n  /** Tokenizer for splitting text into sentences. */\n  sentenceTokenizer: SentenceTokenizer;\n  /** Function to hyphenate words. */\n  hyphenateWord: (word: string) => string[];\n  /** Function to split text into words with positions. */\n  splitWords: (words: string) => [string, number, number][];\n}\n\nexport const defaultTextSyncOptions: TextSyncOptions = {\n  language: '',\n  speed: 1,\n  newSentenceDelay: 400,\n  sentenceTokenizer: new basic.SentenceTokenizer(),\n  hyphenateWord: basic.hyphenateWord,\n  splitWords: basic.splitWords,\n};\n\ninterface AudioData {\n  pushedDuration: number;\n  done: boolean;\n}\n\ninterface TextData {\n  sentenceStream: SentenceStream;\n  pushedText: string;\n  done: boolean;\n  forwardedHyphens: number;\n  forwardedSentences: number;\n}\n\ntype SyncCallbacks = {\n  textUpdated: (text: TranscriptionSegment) => void;\n};\n\nexport class TextAudioSynchronizer extends (EventEmitter as new () => TypedEmitter<SyncCallbacks>) {\n  #opts: TextSyncOptions;\n  #speed: number;\n\n  #closed = false;\n  #interrupted = false;\n  #closeFut = new Future();\n\n  #playingSegIndex = -1;\n  #finishedSegIndex = -1;\n\n  #textQChanged = new AsyncIterableQueue<number>();\n  #textQ: (TextData | undefined)[] = [];\n  #audioQChanged = new AsyncIterableQueue<number>();\n  #audioQ: (AudioData | undefined)[] = [];\n\n  #playedText = '';\n  #task?: Promise<void>;\n\n  #audioData?: AudioData;\n  #textData?: TextData;\n\n  constructor(opts: TextSyncOptions) {\n    super();\n\n    this.#opts = opts;\n    this.#speed = opts.speed * STANDARD_SPEECH_RATE;\n  }\n\n  pushAudio(frame: AudioFrame) {\n    this.#checkNotClosed();\n    if (!this.#audioData) {\n      this.#audioData = { pushedDuration: 0, done: false };\n      this.#audioQ.push(this.#audioData);\n      this.#audioQChanged.put(1);\n    }\n    this.#audioData.pushedDuration += frame.samplesPerChannel / frame.sampleRate;\n  }\n\n  pushText(text: string) {\n    this.#checkNotClosed();\n    if (!this.#textData) {\n      this.#textData = {\n        sentenceStream: this.#opts.sentenceTokenizer.stream(),\n        pushedText: '',\n        done: false,\n        forwardedHyphens: 0,\n        forwardedSentences: 0,\n      };\n      this.#textQ.push(this.#textData);\n      this.#textQChanged.put(1);\n    }\n\n    this.#textData.pushedText += text;\n    this.#textData.sentenceStream.pushText(text);\n  }\n\n  markAudioSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#audioData) {\n      // create empty audio data if none exists\n      this.pushAudio(new AudioFrame(new Int16Array(), 24000, 1, 0));\n    }\n\n    this.#audioData!.done = true;\n    this.#audioData = undefined;\n  }\n\n  markTextSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#textData) {\n      this.pushText('');\n    }\n\n    this.#textData!.done = true;\n    this.#textData?.sentenceStream.flush();\n    this.#textData?.sentenceStream.close();\n    this.#textData = undefined;\n  }\n\n  segmentPlayoutStarted() {\n    this.#checkNotClosed();\n    this.#playingSegIndex++;\n\n    if (!this.#task) {\n      this.#task = this.#mainLoop();\n    }\n  }\n\n  segmentPlayoutFinished() {\n    this.#checkNotClosed();\n    this.#finishedSegIndex++;\n  }\n\n  get playedText(): string {\n    return this.#playedText;\n  }\n\n  async close(interrupt: boolean) {\n    if (this.#closed) {\n      return;\n    }\n    this.#closed = true;\n    this.#interrupted = interrupt;\n    this.#closeFut.resolve();\n\n    for (const textData of this.#textQ) {\n      textData?.sentenceStream.close();\n    }\n\n    this.#textQ.push(undefined);\n    this.#audioQ.push(undefined);\n    this.#textQChanged.put(1);\n    this.#audioQChanged.put(1);\n\n    await this.#task;\n  }\n\n  async #mainLoop() {\n    let segIndex = 0;\n    let qDone = false;\n\n    while (!qDone) {\n      await this.#textQChanged.next();\n      await this.#audioQChanged.next();\n\n      while (this.#textQ.length && this.#audioQ.length) {\n        const textData = this.#textQ.pop();\n        const audioData = this.#audioQ.pop();\n\n        if (!(textData && audioData)) {\n          qDone = true;\n          break;\n        }\n\n        // wait for segment to start playing\n        while (!this.#closed) {\n          if (this.#playingSegIndex >= segIndex) break;\n          await this.#sleepIfNotClosed(125);\n        }\n\n        const sentenceStream = textData.sentenceStream;\n        const forwardStartTime = Date.now();\n\n        for await (const ev of sentenceStream) {\n          await this.#syncSentence(segIndex, forwardStartTime, textData, audioData, ev.token);\n        }\n\n        segIndex++;\n      }\n    }\n  }\n\n  async #syncSentence(\n    segIndex: number,\n    segStartTime: number,\n    textData: TextData,\n    audioData: AudioData,\n    sentence: string,\n  ) {\n    let realSpeed: number | undefined;\n    if (audioData.pushedDuration > 0 && audioData.done) {\n      realSpeed = this.#calcHyphens(textData.pushedText).length / audioData.pushedDuration;\n    }\n\n    const segId = shortuuid('SG_');\n    const words = this.#opts.splitWords(sentence);\n    const processedWords: string[] = [];\n\n    const ogText = this.#playedText;\n    // eslint-disable-next-line @typescript-eslint/no-unused-vars\n    for (const [word, _, end] of words) {\n      if (segIndex <= this.#finishedSegIndex) break;\n      if (this.#interrupted) return;\n\n      const wordHyphens = this.#opts.hyphenateWord(word).length;\n      processedWords.push(word);\n\n      const elapsed = Date.now() - segStartTime;\n      const text = sentence.slice(0, end); // TODO: rstrip punctuations\n\n      let speed = this.#speed;\n      let delay: number;\n      if (realSpeed) {\n        speed = realSpeed;\n        const estimatedPausesMs = textData.forwardedSentences * this.#opts.newSentenceDelay;\n        const hyphPauses = estimatedPausesMs * speed;\n        const targetHyphens = Math.round(speed * elapsed);\n        const dt = targetHyphens - textData.forwardedHyphens - hyphPauses;\n        const toWaitHyphens = Math.max(0, wordHyphens - dt);\n        delay = toWaitHyphens / speed;\n      } else {\n        delay = wordHyphens / speed;\n      }\n\n      const firstDelay = Math.min(delay / 2, 2 / speed);\n      await this.#sleepIfNotClosed(firstDelay * 1000000);\n\n      this.emit(\n        'textUpdated',\n        new TranscriptionSegment({\n          id: segId,\n          text: text,\n          startTime: BigInt(0),\n          endTime: BigInt(0),\n          final: false,\n          language: this.#opts.language,\n        }),\n      );\n\n      this.#playedText = `${ogText} ${text}`;\n      await this.#sleepIfNotClosed((delay - firstDelay) * 1000000);\n      textData.forwardedHyphens += wordHyphens;\n    }\n\n    this.emit(\n      'textUpdated',\n      new TranscriptionSegment({\n        id: segId,\n        text: sentence,\n        startTime: BigInt(0),\n        endTime: BigInt(0),\n        final: true,\n        language: this.#opts.language,\n      }),\n    );\n\n    this.#playedText = `${ogText} ${sentence}`;\n\n    await this.#sleepIfNotClosed(this.#opts.newSentenceDelay);\n    textData.forwardedSentences++;\n  }\n\n  async #sleepIfNotClosed(delay: number) {\n    await Promise.race([\n      this.#closeFut.await,\n      new Promise((resolve) => setTimeout(resolve, delay)),\n    ]);\n  }\n\n  #calcHyphens(text: string): string[] {\n    const hyphens: string[] = [];\n    const words = this.#opts.splitWords(text);\n    for (const word of words) {\n      const n = this.#opts.hyphenateWord(word[0]);\n      hyphens.push(...n);\n    }\n    return hyphens;\n  }\n\n  #checkNotClosed() {\n    if (this.#closed) {\n      throw new Error('TextAudioSynchronizer is closed');\n    }\n  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package/dist/transcription.d.cts

@@ -3,11 +3,17 @@ import { AudioFrame } from '@livekit/rtc-node';
 import type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';
 import type { SentenceTokenizer } from './tokenize/tokenizer.js';
 export interface TextSyncOptions {
+    /** Language code for transcription. */
     language: string;
+    /** Speech speed multiplier. */
     speed: number;
+    /** Delay between sentences in milliseconds. */
     newSentenceDelay: number;
+    /** Tokenizer for splitting text into sentences. */
     sentenceTokenizer: SentenceTokenizer;
+    /** Function to hyphenate words. */
     hyphenateWord: (word: string) => string[];
+    /** Function to split text into words with positions. */
     splitWords: (words: string) => [string, number, number][];
 }
 export declare const defaultTextSyncOptions: TextSyncOptions;

package/dist/transcription.d.ts

@@ -3,11 +3,17 @@ import { AudioFrame } from '@livekit/rtc-node';
 import type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';
 import type { SentenceTokenizer } from './tokenize/tokenizer.js';
 export interface TextSyncOptions {
+    /** Language code for transcription. */
     language: string;
+    /** Speech speed multiplier. */
     speed: number;
+    /** Delay between sentences in milliseconds. */
     newSentenceDelay: number;
+    /** Tokenizer for splitting text into sentences. */
     sentenceTokenizer: SentenceTokenizer;
+    /** Function to hyphenate words. */
     hyphenateWord: (word: string) => string[];
+    /** Function to split text into words with positions. */
     splitWords: (words: string) => [string, number, number][];
 }
 export declare const defaultTextSyncOptions: TextSyncOptions;

package/dist/transcription.d.ts.map

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package/dist/transcription.js.map

@@ -1 +1 @@
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this.#textData.sentenceStream.pushText(text);\n  }\n\n  markAudioSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#audioData) {\n      // create empty audio data if none exists\n      this.pushAudio(new AudioFrame(new Int16Array(), 24000, 1, 0));\n    }\n\n    this.#audioData!.done = true;\n    this.#audioData = undefined;\n  }\n\n  markTextSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#textData) {\n      this.pushText('');\n    }\n\n    this.#textData!.done = true;\n    this.#textData?.sentenceStream.flush();\n    this.#textData?.sentenceStream.close();\n    this.#textData = undefined;\n  }\n\n  segmentPlayoutStarted() {\n    this.#checkNotClosed();\n    this.#playingSegIndex++;\n\n    if (!this.#task) {\n      this.#task = this.#mainLoop();\n    }\n  }\n\n  segmentPlayoutFinished() {\n    this.#checkNotClosed();\n    this.#finishedSegIndex++;\n  }\n\n  get playedText(): string {\n    return this.#playedText;\n  }\n\n  async close(interrupt: boolean) {\n    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this.emit(\n        'textUpdated',\n        new TranscriptionSegment({\n          id: segId,\n          text: text,\n          startTime: BigInt(0),\n          endTime: BigInt(0),\n          final: false,\n          language: this.#opts.language,\n        }),\n      );\n\n      this.#playedText = `${ogText} ${text}`;\n      await this.#sleepIfNotClosed((delay - firstDelay) * 1000000);\n      textData.forwardedHyphens += wordHyphens;\n    }\n\n    this.emit(\n      'textUpdated',\n      new TranscriptionSegment({\n        id: segId,\n        text: sentence,\n        startTime: BigInt(0),\n        endTime: BigInt(0),\n        final: true,\n        language: this.#opts.language,\n      }),\n    );\n\n    this.#playedText = `${ogText} ${sentence}`;\n\n    await this.#sleepIfNotClosed(this.#opts.newSentenceDelay);\n    textData.forwardedSentences++;\n  }\n\n  async #sleepIfNotClosed(delay: number) {\n    await Promise.race([\n      this.#closeFut.await,\n      new Promise((resolve) => 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+{"version":3,"sources":["../src/transcription.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { TranscriptionSegment } from '@livekit/protocol';\nimport { AudioFrame } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport { basic } from './tokenize/index.js';\nimport type { SentenceStream, SentenceTokenizer } from './tokenize/tokenizer.js';\nimport { AsyncIterableQueue, Future, shortuuid } from './utils.js';\n\n// standard speech rate in hyphens/ms\nconst STANDARD_SPEECH_RATE = 3830;\n\nexport interface TextSyncOptions {\n  /** Language code for transcription. */\n  language: string;\n  /** Speech speed multiplier. */\n  speed: number;\n  /** Delay between sentences in milliseconds. */\n  newSentenceDelay: number;\n  /** Tokenizer for splitting text into sentences. */\n  sentenceTokenizer: 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#closeFut = new Future();\n\n  #playingSegIndex = -1;\n  #finishedSegIndex = -1;\n\n  #textQChanged = new AsyncIterableQueue<number>();\n  #textQ: (TextData | undefined)[] = [];\n  #audioQChanged = new AsyncIterableQueue<number>();\n  #audioQ: (AudioData | undefined)[] = [];\n\n  #playedText = '';\n  #task?: Promise<void>;\n\n  #audioData?: AudioData;\n  #textData?: TextData;\n\n  constructor(opts: TextSyncOptions) {\n    super();\n\n    this.#opts = opts;\n    this.#speed = opts.speed * STANDARD_SPEECH_RATE;\n  }\n\n  pushAudio(frame: AudioFrame) {\n    this.#checkNotClosed();\n    if (!this.#audioData) {\n      this.#audioData = { pushedDuration: 0, done: false };\n      this.#audioQ.push(this.#audioData);\n      this.#audioQChanged.put(1);\n    }\n    this.#audioData.pushedDuration += frame.samplesPerChannel / frame.sampleRate;\n  }\n\n  pushText(text: string) {\n    this.#checkNotClosed();\n    if (!this.#textData) {\n      this.#textData = {\n        sentenceStream: this.#opts.sentenceTokenizer.stream(),\n        pushedText: '',\n        done: false,\n        forwardedHyphens: 0,\n        forwardedSentences: 0,\n      };\n      this.#textQ.push(this.#textData);\n      this.#textQChanged.put(1);\n    }\n\n    this.#textData.pushedText += text;\n    this.#textData.sentenceStream.pushText(text);\n  }\n\n  markAudioSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#audioData) {\n      // create empty audio data if none exists\n      this.pushAudio(new AudioFrame(new Int16Array(), 24000, 1, 0));\n    }\n\n    this.#audioData!.done = true;\n    this.#audioData = undefined;\n  }\n\n  markTextSegmentEnd() {\n    this.#checkNotClosed();\n\n    if (!this.#textData) {\n      this.pushText('');\n    }\n\n    this.#textData!.done = true;\n    this.#textData?.sentenceStream.flush();\n    this.#textData?.sentenceStream.close();\n    this.#textData = undefined;\n  }\n\n  segmentPlayoutStarted() {\n    this.#checkNotClosed();\n    this.#playingSegIndex++;\n\n    if (!this.#task) {\n      this.#task = this.#mainLoop();\n    }\n  }\n\n  segmentPlayoutFinished() {\n    this.#checkNotClosed();\n    this.#finishedSegIndex++;\n  }\n\n  get playedText(): string {\n    return this.#playedText;\n  }\n\n  async close(interrupt: boolean) {\n    if (this.#closed) {\n      return;\n    }\n    this.#closed = true;\n    this.#interrupted = interrupt;\n    this.#closeFut.resolve();\n\n    for (const textData of this.#textQ) {\n      textData?.sentenceStream.close();\n    }\n\n    this.#textQ.push(undefined);\n    this.#audioQ.push(undefined);\n    this.#textQChanged.put(1);\n    this.#audioQChanged.put(1);\n\n    await this.#task;\n  }\n\n  async #mainLoop() {\n    let segIndex = 0;\n    let qDone = false;\n\n    while (!qDone) {\n      await this.#textQChanged.next();\n      await this.#audioQChanged.next();\n\n      while (this.#textQ.length && this.#audioQ.length) {\n        const textData = this.#textQ.pop();\n        const audioData = this.#audioQ.pop();\n\n        if (!(textData && audioData)) {\n          qDone = true;\n          break;\n        }\n\n        // wait for segment to start playing\n        while (!this.#closed) {\n          if (this.#playingSegIndex >= segIndex) break;\n          await this.#sleepIfNotClosed(125);\n        }\n\n        const sentenceStream = textData.sentenceStream;\n        const forwardStartTime = Date.now();\n\n        for await (const ev of sentenceStream) {\n          await this.#syncSentence(segIndex, forwardStartTime, textData, audioData, ev.token);\n        }\n\n        segIndex++;\n      }\n    }\n  }\n\n  async #syncSentence(\n    segIndex: number,\n    segStartTime: number,\n    textData: TextData,\n    audioData: AudioData,\n    sentence: string,\n  ) {\n    let realSpeed: number | undefined;\n    if (audioData.pushedDuration > 0 && audioData.done) {\n      realSpeed = this.#calcHyphens(textData.pushedText).length / audioData.pushedDuration;\n    }\n\n    const segId = shortuuid('SG_');\n    const words = this.#opts.splitWords(sentence);\n    const processedWords: string[] = [];\n\n    const ogText = this.#playedText;\n    // eslint-disable-next-line @typescript-eslint/no-unused-vars\n    for (const [word, _, end] of words) {\n      if (segIndex <= this.#finishedSegIndex) break;\n      if (this.#interrupted) return;\n\n      const wordHyphens = this.#opts.hyphenateWord(word).length;\n      processedWords.push(word);\n\n      const elapsed = Date.now() - segStartTime;\n      const text = sentence.slice(0, end); // TODO: rstrip punctuations\n\n      let speed = this.#speed;\n      let delay: number;\n      if (realSpeed) {\n        speed = realSpeed;\n        const estimatedPausesMs = textData.forwardedSentences * this.#opts.newSentenceDelay;\n        const hyphPauses = estimatedPausesMs * speed;\n        const targetHyphens = Math.round(speed * elapsed);\n        const dt = targetHyphens - textData.forwardedHyphens - hyphPauses;\n        const toWaitHyphens = Math.max(0, wordHyphens - dt);\n        delay = toWaitHyphens / speed;\n      } else {\n        delay = wordHyphens / speed;\n      }\n\n      const firstDelay = Math.min(delay / 2, 2 / speed);\n      await this.#sleepIfNotClosed(firstDelay * 1000000);\n\n      this.emit(\n        'textUpdated',\n        new TranscriptionSegment({\n          id: segId,\n          text: text,\n          startTime: BigInt(0),\n          endTime: BigInt(0),\n          final: false,\n          language: this.#opts.language,\n        }),\n      );\n\n      this.#playedText = `${ogText} ${text}`;\n      await this.#sleepIfNotClosed((delay - firstDelay) * 1000000);\n      textData.forwardedHyphens += wordHyphens;\n    }\n\n    this.emit(\n      'textUpdated',\n      new TranscriptionSegment({\n        id: segId,\n        text: sentence,\n        startTime: BigInt(0),\n        endTime: BigInt(0),\n        final: true,\n        language: this.#opts.language,\n      }),\n    );\n\n    this.#playedText = `${ogText} ${sentence}`;\n\n    await this.#sleepIfNotClosed(this.#opts.newSentenceDelay);\n    textData.forwardedSentences++;\n  }\n\n  async #sleepIfNotClosed(delay: number) {\n    await Promise.race([\n      this.#closeFut.await,\n      new Promise((resolve) => setTimeout(resolve, delay)),\n    ]);\n  }\n\n  #calcHyphens(text: string): string[] {\n    const hyphens: string[] = [];\n    const words = this.#opts.splitWords(text);\n    for (const word of words) {\n      const n = this.#opts.hyphenateWord(word[0]);\n      hyphens.push(...n);\n    }\n    return hyphens;\n  }\n\n  #checkNotClosed() {\n    if (this.#closed) {\n      throw new Error('TextAudioSynchronizer is closed');\n    }\n  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package/dist/vad.cjs

@@ -108,7 +108,7 @@ class VADStream {
       switch (value.type) {
         case 0 /* START_OF_SPEECH */:
           inferenceCount++;
-          if (inferenceCount >= 1 / this.#vad.capabilities.updateInterval) {
+          if (inferenceCount >= 1e3 / this.#vad.capabilities.updateInterval) {
             this.#vad.emit("metrics_collected", {
               type: "vad_metrics",
               timestamp: Date.now(),

package/dist/vad.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../src/vad.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type {\n  ReadableStream,\n  ReadableStreamDefaultReader,\n  WritableStreamDefaultWriter,\n} from 'node:stream/web';\nimport { log } from './log.js';\nimport type { VADMetrics } from './metrics/base.js';\nimport { DeferredReadableStream } from './stream/deferred_stream.js';\nimport { IdentityTransform } from './stream/identity_transform.js';\n\nexport enum VADEventType {\n  START_OF_SPEECH,\n  INFERENCE_DONE,\n  END_OF_SPEECH,\n  METRICS_COLLECTED,\n}\n\nexport interface VADEvent {\n  /** Type of the VAD event (e.g., start of speech, end of speech, inference done). */\n  type: VADEventType;\n  /**\n   * Index of the audio sample where the event occurred, relative to the inference sample rate.\n   */\n  samplesIndex: number;\n  /** Timestamp when the event was fired. */\n  timestamp: number;\n  /** Duration of the speech segment. */\n  speechDuration: number;\n  /** Duration of the silence segment. */\n  silenceDuration: number;\n  /**\n   * List of audio frames associated with the speech.\n   *\n   * @remarks\n   * - For `start_of_speech` events, this contains the audio chunks that triggered the detection.\n   * - For `inference_done` events, this contains the audio chunks that were processed.\n   * - For `end_of_speech` events, this contains the complete user speech.\n   */\n  frames: AudioFrame[];\n  /** Probability that speech is present (only for `INFERENCE_DONE` events). */\n  probability: number;\n  /** Time taken to perform the inference, in seconds (only for `INFERENCE_DONE` events). */\n  inferenceDuration: number;\n  /** Indicates whether speech was detected in the frames. */\n  speaking: boolean;\n  /** Threshold used to detect silence. */\n  rawAccumulatedSilence: number;\n  /** Threshold used to detect speech. */\n  rawAccumulatedSpeech: number;\n}\n\nexport interface VADCapabilities {\n  updateInterval: number;\n}\n\nexport type VADCallbacks = {\n  ['metrics_collected']: (metrics: VADMetrics) => void;\n};\n\nexport abstract class VAD extends (EventEmitter as new () => TypedEmitter<VADCallbacks>) {\n  #capabilities: VADCapabilities;\n  abstract label: string;\n\n  constructor(capabilities: VADCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  get capabilities(): VADCapabilities {\n    return this.#capabilities;\n  }\n\n  /**\n   * Returns a {@link VADStream} that can be used to push audio frames and receive VAD events.\n   */\n  abstract stream(): VADStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\nexport abstract class VADStream implements AsyncIterableIterator<VADEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new IdentityTransform<AudioFrame | typeof VADStream.FLUSH_SENTINEL>();\n  protected output = new IdentityTransform<VADEvent>();\n  protected inputWriter: WritableStreamDefaultWriter<AudioFrame | typeof VADStream.FLUSH_SENTINEL>;\n  protected inputReader: ReadableStreamDefaultReader<AudioFrame | typeof VADStream.FLUSH_SENTINEL>;\n  protected outputWriter: WritableStreamDefaultWriter<VADEvent>;\n  protected outputReader: ReadableStreamDefaultReader<VADEvent>;\n  protected closed = false;\n  protected inputClosed = false;\n\n  #vad: VAD;\n  #lastActivityTime = BigInt(0);\n  private logger = log();\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n\n  private metricsStream: ReadableStream<VADEvent>;\n  constructor(vad: VAD) {\n    this.#vad = vad;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n\n    this.inputWriter = this.input.writable.getWriter();\n    this.inputReader = this.input.readable.getReader();\n    this.outputWriter = this.output.writable.getWriter();\n\n    const [outputStream, metricsStream] = this.output.readable.tee();\n    this.metricsStream = metricsStream;\n    this.outputReader = outputStream.getReader();\n\n    this.pumpDeferredStream();\n    this.monitorMetrics();\n  }\n\n  /**\n   * Reads from the deferred input stream and forwards chunks to the input writer.\n   *\n   * Note: we can't just do this.deferredInputStream.stream.pipeTo(this.input.writable)\n   * because the inputWriter locks the this.input.writable stream. All writes must go through\n   * the inputWriter.\n   */\n  private async pumpDeferredStream() {\n    const reader = this.deferredInputStream.stream.getReader();\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        await this.inputWriter.write(value);\n      }\n    } catch (e) {\n      this.logger.error(`Error pumping deferred stream: ${e}`);\n      throw e;\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    let inferenceDurationTotalMs = 0;\n    let inferenceCount = 0;\n    const metricsReader = this.metricsStream.getReader();\n    while (true) {\n      const { done, value } = await metricsReader.read();\n      if (done) {\n        break;\n      }\n      switch (value.type) {\n        case VADEventType.START_OF_SPEECH:\n          inferenceCount++;\n          if (inferenceCount >= 1 / this.#vad.capabilities.updateInterval) {\n            this.#vad.emit('metrics_collected', {\n              type: 'vad_metrics',\n              timestamp: Date.now(),\n              idleTimeMs: Math.trunc(\n                Number((process.hrtime.bigint() - this.#lastActivityTime) / BigInt(1000000)),\n              ),\n              inferenceDurationTotalMs,\n              inferenceCount,\n              label: this.#vad.label,\n            });\n\n            inferenceCount = 0;\n            inferenceDurationTotalMs = 0;\n          }\n          break;\n        case VADEventType.INFERENCE_DONE:\n          inferenceDurationTotalMs += Math.round(value.inferenceDuration);\n          this.#lastActivityTime = process.hrtime.bigint();\n          break;\n        case VADEventType.END_OF_SPEECH:\n          this.#lastActivityTime = process.hrtime.bigint();\n          break;\n      }\n    }\n  }\n\n  /**\n   * Safely send a VAD event to the output stream, handling writer release errors during shutdown.\n   * @returns true if the event was sent, false 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+{"version":3,"sources":["../src/vad.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2024 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport type { TypedEventEmitter as TypedEmitter } from '@livekit/typed-emitter';\nimport { EventEmitter } from 'node:events';\nimport type {\n  ReadableStream,\n  ReadableStreamDefaultReader,\n  WritableStreamDefaultWriter,\n} from 'node:stream/web';\nimport { log } from './log.js';\nimport type { VADMetrics } from './metrics/base.js';\nimport { DeferredReadableStream } from './stream/deferred_stream.js';\nimport { IdentityTransform } from './stream/identity_transform.js';\n\nexport enum VADEventType {\n  START_OF_SPEECH,\n  INFERENCE_DONE,\n  END_OF_SPEECH,\n  METRICS_COLLECTED,\n}\n\nexport interface VADEvent {\n  /** Type of the VAD event (e.g., start of speech, end of speech, inference done). */\n  type: VADEventType;\n  /**\n   * Index of the audio sample where the event occurred, relative to the inference sample rate.\n   */\n  samplesIndex: number;\n  /** Timestamp when the event was fired. */\n  timestamp: number;\n  /** Duration of the speech segment in seconds. */\n  speechDuration: number;\n  /** Duration of the silence segment in seconds. */\n  silenceDuration: number;\n  /**\n   * List of audio frames associated with the speech.\n   *\n   * @remarks\n   * - For `start_of_speech` events, this contains the audio chunks that triggered the detection.\n   * - For `inference_done` events, this contains the audio chunks that were processed.\n   * - For `end_of_speech` events, this contains the complete user speech.\n   */\n  frames: AudioFrame[];\n  /** Probability that speech is present (only for `INFERENCE_DONE` events). */\n  probability: number;\n  /** Time taken to perform the inference, in seconds (only for `INFERENCE_DONE` events). */\n  inferenceDuration: number;\n  /** Indicates whether speech was detected in the frames. */\n  speaking: boolean;\n  /** Threshold used to detect silence. */\n  rawAccumulatedSilence: number;\n  /** Threshold used to detect speech. */\n  rawAccumulatedSpeech: number;\n}\n\nexport interface VADCapabilities {\n  /** Duration of each VAD inference window in milliseconds. Used to batch metrics emissions to roughly once per second. */\n  updateInterval: number;\n}\n\nexport type VADCallbacks = {\n  ['metrics_collected']: (metrics: VADMetrics) => void;\n};\n\nexport abstract class VAD extends (EventEmitter as new () => TypedEmitter<VADCallbacks>) {\n  #capabilities: VADCapabilities;\n  abstract label: string;\n\n  constructor(capabilities: VADCapabilities) {\n    super();\n    this.#capabilities = capabilities;\n  }\n\n  get capabilities(): VADCapabilities {\n    return this.#capabilities;\n  }\n\n  /**\n   * Returns a {@link VADStream} that can be used to push audio frames and receive VAD events.\n   */\n  abstract stream(): VADStream;\n\n  async close(): Promise<void> {\n    return;\n  }\n}\n\nexport abstract class VADStream implements AsyncIterableIterator<VADEvent> {\n  protected static readonly FLUSH_SENTINEL = Symbol('FLUSH_SENTINEL');\n  protected input = new IdentityTransform<AudioFrame | typeof VADStream.FLUSH_SENTINEL>();\n  protected output = new IdentityTransform<VADEvent>();\n  protected inputWriter: WritableStreamDefaultWriter<AudioFrame | typeof VADStream.FLUSH_SENTINEL>;\n  protected inputReader: ReadableStreamDefaultReader<AudioFrame | typeof VADStream.FLUSH_SENTINEL>;\n  protected outputWriter: WritableStreamDefaultWriter<VADEvent>;\n  protected outputReader: ReadableStreamDefaultReader<VADEvent>;\n  protected closed = false;\n  protected inputClosed = false;\n\n  #vad: VAD;\n  #lastActivityTime = BigInt(0);\n  private logger = log();\n  private deferredInputStream: DeferredReadableStream<AudioFrame>;\n\n  private metricsStream: ReadableStream<VADEvent>;\n  constructor(vad: VAD) {\n    this.#vad = vad;\n    this.deferredInputStream = new DeferredReadableStream<AudioFrame>();\n\n    this.inputWriter = this.input.writable.getWriter();\n    this.inputReader = this.input.readable.getReader();\n    this.outputWriter = this.output.writable.getWriter();\n\n    const [outputStream, metricsStream] = this.output.readable.tee();\n    this.metricsStream = metricsStream;\n    this.outputReader = outputStream.getReader();\n\n    this.pumpDeferredStream();\n    this.monitorMetrics();\n  }\n\n  /**\n   * Reads from the deferred input stream and forwards chunks to the input writer.\n   *\n   * Note: we can't just do this.deferredInputStream.stream.pipeTo(this.input.writable)\n   * because the inputWriter locks the this.input.writable stream. All writes must go through\n   * the inputWriter.\n   */\n  private async pumpDeferredStream() {\n    const reader = this.deferredInputStream.stream.getReader();\n    try {\n      while (true) {\n        const { done, value } = await reader.read();\n        if (done) break;\n        await this.inputWriter.write(value);\n      }\n    } catch (e) {\n      this.logger.error(`Error pumping deferred stream: ${e}`);\n      throw e;\n    } finally {\n      reader.releaseLock();\n    }\n  }\n\n  protected async monitorMetrics() {\n    let inferenceDurationTotalMs = 0;\n    let inferenceCount = 0;\n    const metricsReader = this.metricsStream.getReader();\n    while (true) {\n      const { done, value } = await metricsReader.read();\n      if (done) {\n        break;\n      }\n      switch (value.type) {\n        case VADEventType.START_OF_SPEECH:\n          inferenceCount++;\n          if (inferenceCount >= 1000 / this.#vad.capabilities.updateInterval) {\n            this.#vad.emit('metrics_collected', {\n              type: 'vad_metrics',\n              timestamp: Date.now(),\n              idleTimeMs: Math.trunc(\n                Number((process.hrtime.bigint() - this.#lastActivityTime) / BigInt(1000000)),\n              ),\n              inferenceDurationTotalMs,\n              inferenceCount,\n              label: this.#vad.label,\n            });\n\n            inferenceCount = 0;\n            inferenceDurationTotalMs = 0;\n          }\n          break;\n        case VADEventType.INFERENCE_DONE:\n          inferenceDurationTotalMs += Math.round(value.inferenceDuration);\n          this.#lastActivityTime = process.hrtime.bigint();\n          break;\n        case VADEventType.END_OF_SPEECH:\n          this.#lastActivityTime = process.hrtime.bigint();\n          break;\n      }\n    }\n  }\n\n  /**\n   * Safely send a VAD event to the output stream, handling writer release errors during shutdown.\n   * @returns true if the event was sent, false 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package/dist/vad.d.cts

@@ -19,9 +19,9 @@ export interface VADEvent {
     samplesIndex: number;
     /** Timestamp when the event was fired. */
     timestamp: number;
-    /** Duration of the speech segment. */
+    /** Duration of the speech segment in seconds. */
     speechDuration: number;
-    /** Duration of the silence segment. */
+    /** Duration of the silence segment in seconds. */
     silenceDuration: number;
     /**
      * List of audio frames associated with the speech.
@@ -44,6 +44,7 @@ export interface VADEvent {
     rawAccumulatedSpeech: number;
 }
 export interface VADCapabilities {
+    /** Duration of each VAD inference window in milliseconds. Used to batch metrics emissions to roughly once per second. */
     updateInterval: number;
 }
 export type VADCallbacks = {