@livekit/agents 1.0.24 → 1.0.27

package/dist/stt/stt.cjs

@@ -51,9 +51,9 @@ class STT extends import_node_events.EventEmitter {
     return this.#capabilities;
   }
   /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */
-  async recognize(frame) {
+  async recognize(frame, abortSignal) {
     const startTime = process.hrtime.bigint();
-    const event = await this._recognize(frame);
+    const event = await this._recognize(frame, abortSignal);
     const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1e6));
     this.emit("metrics_collected", {
       type: "stt_metrics",
@@ -82,6 +82,7 @@ class SpeechStream {
   deferredInputStream;
   logger = (0, import_log.log)();
   _connOptions;
+  abortController = new AbortController();
   constructor(stt, sampleRate, connectionOptions = import_types.DEFAULT_API_CONNECT_OPTIONS) {
     this.#stt = stt;
     this._connOptions = connectionOptions;
@@ -97,7 +98,7 @@ class SpeechStream {
         return await this.run();
       } catch (error) {
         if (error instanceof import_exceptions.APIError) {
-          const retryInterval = this._connOptions._intervalForRetry(i);
+          const retryInterval = (0, import_types.intervalForRetry)(this._connOptions, i);
           if (this._connOptions.maxRetry === 0 || !error.retryable) {
             this.emitError({ error, recoverable: false });
             throw error;
@@ -177,6 +178,9 @@ class SpeechStream {
       this.output.close();
     }
   }
+  get abortSignal() {
+    return this.abortController.signal;
+  }
   updateInputStream(audioStream) {
     this.deferredInputStream.setSource(audioStream);
   }
@@ -233,6 +237,7 @@ class SpeechStream {
     if (!this.input.closed) this.input.close();
     if (!this.queue.closed) this.queue.close();
     if (!this.output.closed) this.output.close();
+    if (!this.abortController.signal.aborted) this.abortController.abort();
     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      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  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) 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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';\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, 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\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().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 = 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: 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(!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 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package/dist/stt/stt.d.cts

@@ -91,13 +91,17 @@ export declare abstract class STT extends STT_base {
     /** Returns this STT's capabilities */
     get capabilities(): STTCapabilities;
     /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */
-    recognize(frame: AudioBuffer): Promise<SpeechEvent>;
-    protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;
+    recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent>;
+    protected abstract _recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent>;
     /**
      * Returns a {@link SpeechStream} that can be used to push audio frames and receive
      * transcriptions
+     *
+     * @param options - Optional configuration including connection options
      */
-    abstract stream(): SpeechStream;
+    abstract stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
     close(): Promise<void>;
 }
 /**
@@ -129,12 +133,14 @@ export declare abstract class SpeechStream implements AsyncIterableIterator<Spee
     private deferredInputStream;
     private logger;
     private _connOptions;
+    protected abortController: AbortController;
     constructor(stt: STT, sampleRate?: number, connectionOptions?: APIConnectOptions);
     private mainTask;
     private emitError;
     protected pumpInput(): Promise<void>;
     protected monitorMetrics(): Promise<void>;
     protected abstract run(): Promise<void>;
+    protected get abortSignal(): AbortSignal;
     updateInputStream(audioStream: ReadableStream<AudioFrame>): void;
     detachInputStream(): void;
     /** Push an audio frame to the STT */

package/dist/stt/stt.d.ts

@@ -91,13 +91,17 @@ export declare abstract class STT extends STT_base {
     /** Returns this STT's capabilities */
     get capabilities(): STTCapabilities;
     /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */
-    recognize(frame: AudioBuffer): Promise<SpeechEvent>;
-    protected abstract _recognize(frame: AudioBuffer): Promise<SpeechEvent>;
+    recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent>;
+    protected abstract _recognize(frame: AudioBuffer, abortSignal?: AbortSignal): Promise<SpeechEvent>;
     /**
      * Returns a {@link SpeechStream} that can be used to push audio frames and receive
      * transcriptions
+     *
+     * @param options - Optional configuration including connection options
      */
-    abstract stream(): SpeechStream;
+    abstract stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): SpeechStream;
     close(): Promise<void>;
 }
 /**
@@ -129,12 +133,14 @@ export declare abstract class SpeechStream implements AsyncIterableIterator<Spee
     private deferredInputStream;
     private logger;
     private _connOptions;
+    protected abortController: AbortController;
     constructor(stt: STT, sampleRate?: number, connectionOptions?: APIConnectOptions);
     private mainTask;
     private emitError;
     protected pumpInput(): Promise<void>;
     protected monitorMetrics(): Promise<void>;
     protected abstract run(): Promise<void>;
+    protected get abortSignal(): AbortSignal;
     updateInputStream(audioStream: ReadableStream<AudioFrame>): void;
     detachInputStream(): void;
     /** Push an audio frame to the STT */

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

@@ -1 +1 @@
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package/dist/stt/stt.js

@@ -4,7 +4,7 @@ import { APIConnectionError, APIError } from "../_exceptions.js";
 import { calculateAudioDurationSeconds } from "../audio.js";
 import { log } from "../log.js";
 import { DeferredReadableStream } from "../stream/deferred_stream.js";
-import { DEFAULT_API_CONNECT_OPTIONS } from "../types.js";
+import { DEFAULT_API_CONNECT_OPTIONS, intervalForRetry } from "../types.js";
 import { AsyncIterableQueue, delay, startSoon, toError } from "../utils.js";
 var SpeechEventType = /* @__PURE__ */ ((SpeechEventType2) => {
   SpeechEventType2[SpeechEventType2["START_OF_SPEECH"] = 0] = "START_OF_SPEECH";
@@ -26,9 +26,9 @@ class STT extends EventEmitter {
     return this.#capabilities;
   }
   /** Receives an audio buffer and returns transcription in the form of a {@link SpeechEvent} */
-  async recognize(frame) {
+  async recognize(frame, abortSignal) {
     const startTime = process.hrtime.bigint();
-    const event = await this._recognize(frame);
+    const event = await this._recognize(frame, abortSignal);
     const durationMs = Number((process.hrtime.bigint() - startTime) / BigInt(1e6));
     this.emit("metrics_collected", {
       type: "stt_metrics",
@@ -57,6 +57,7 @@ class SpeechStream {
   deferredInputStream;
   logger = log();
   _connOptions;
+  abortController = new AbortController();
   constructor(stt, sampleRate, connectionOptions = DEFAULT_API_CONNECT_OPTIONS) {
     this.#stt = stt;
     this._connOptions = connectionOptions;
@@ -72,7 +73,7 @@ class SpeechStream {
         return await this.run();
       } catch (error) {
         if (error instanceof APIError) {
-          const retryInterval = this._connOptions._intervalForRetry(i);
+          const retryInterval = intervalForRetry(this._connOptions, i);
           if (this._connOptions.maxRetry === 0 || !error.retryable) {
             this.emitError({ error, recoverable: false });
             throw error;
@@ -152,6 +153,9 @@ class SpeechStream {
       this.output.close();
     }
   }
+  get abortSignal() {
+    return this.abortController.signal;
+  }
   updateInputStream(audioStream) {
     this.deferredInputStream.setSource(audioStream);
   }
@@ -208,6 +212,7 @@ class SpeechStream {
     if (!this.input.closed) this.input.close();
     if (!this.queue.closed) this.queue.close();
     if (!this.output.closed) this.output.close();
+    if (!this.abortController.signal.aborted) this.abortController.abort();
     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      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  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, 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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';\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, 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\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().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 = 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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(!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 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package/dist/telemetry/traces.cjs

@@ -45,6 +45,7 @@ var import_sdk_trace_node = require("@opentelemetry/sdk-trace-node");
 var import_semantic_conventions = require("@opentelemetry/semantic-conventions");
 var import_form_data = __toESM(require("form-data"), 1);
 var import_livekit_server_sdk = require("livekit-server-sdk");
+var import_promises = __toESM(require("node:fs/promises"), 1);
 var import_log = require("../log.cjs");
 var import_otel_http_exporter = require("./otel_http_exporter.cjs");
 var import_pino_otel_transport = require("./pino_otel_transport.cjs");
@@ -336,13 +337,14 @@ async function uploadSessionReport(options) {
   token.addObservabilityGrant({ write: true });
   const jwt = await token.toJwt();
   const formData = new import_form_data.default();
+  const audioStartTime = report.audioRecordingStartedAt ?? 0;
   const headerMsg = new import_protocol.MetricsRecordingHeader({
     roomId: report.roomId,
     duration: BigInt(0),
     // TODO: Calculate actual duration from report
     startTime: {
-      seconds: BigInt(Math.floor(report.timestamp / 1e3)),
-      nanos: Math.floor(report.timestamp % 1e3 * 1e6)
+      seconds: BigInt(Math.floor(audioStartTime / 1e3)),
+      nanos: Math.floor(audioStartTime % 1e3 * 1e6)
     }
   });
   const headerBytes = Buffer.from(headerMsg.toBinary());
@@ -366,6 +368,25 @@ async function uploadSessionReport(options) {
       "Content-Length": chatHistoryBuffer.length.toString()
     }
   });
+  if (report.audioRecordingPath && report.audioRecordingStartedAt) {
+    let audioBytes;
+    try {
+      audioBytes = await import_promises.default.readFile(report.audioRecordingPath);
+    } catch {
+      audioBytes = Buffer.alloc(0);
+    }
+    if (audioBytes.length > 0) {
+      formData.append("audio", audioBytes, {
+        filename: "recording.ogg",
+        contentType: "audio/ogg",
+        knownLength: audioBytes.length,
+        header: {
+          "Content-Type": "audio/ogg",
+          "Content-Length": audioBytes.length.toString()
+        }
+      });
+    }
+  }
   return new Promise((resolve, reject) => {
     formData.submit(
       {

package/dist/telemetry/traces.cjs.map

@@ -1 +1 @@
-{"version":3,"sources":["../../src/telemetry/traces.ts"],"sourcesContent":["// SPDX-FileCopyrightText: 2025 LiveKit, Inc.\n//\n// SPDX-License-Identifier: Apache-2.0\nimport { MetricsRecordingHeader } from '@livekit/protocol';\nimport {\n  type Attributes,\n  type Context,\n  type Span,\n  type SpanOptions,\n  type Tracer,\n  type TracerProvider,\n  context as otelContext,\n  trace,\n} from '@opentelemetry/api';\nimport { SeverityNumber } from '@opentelemetry/api-logs';\nimport { OTLPTraceExporter } from '@opentelemetry/exporter-trace-otlp-proto';\nimport { CompressionAlgorithm } from '@opentelemetry/otlp-exporter-base';\nimport { Resource } from '@opentelemetry/resources';\nimport type { ReadableSpan, SpanProcessor } from '@opentelemetry/sdk-trace-base';\nimport { BatchSpanProcessor, NodeTracerProvider } from '@opentelemetry/sdk-trace-node';\nimport { ATTR_SERVICE_NAME } from '@opentelemetry/semantic-conventions';\nimport FormData from 'form-data';\nimport { AccessToken } from 'livekit-server-sdk';\nimport type { ChatContent, ChatItem } from '../llm/index.js';\nimport { enableOtelLogging } from '../log.js';\nimport type { SessionReport } from '../voice/report.js';\nimport { type SimpleLogRecord, SimpleOTLPHttpLogExporter } from './otel_http_exporter.js';\nimport { flushPinoLogs, initPinoCloudExporter } from './pino_otel_transport.js';\n\nexport interface StartSpanOptions {\n  /** Name of the span */\n  name: string;\n  /** Optional parent context to use for this span */\n  context?: Context;\n  /** Attributes to set on the span when it starts */\n  attributes?: Attributes;\n  /** Whether to end the span when the function exits (default: true) */\n  endOnExit?: boolean;\n}\n\n/**\n * A dynamic tracer that allows the tracer provider to be changed at runtime.\n */\nclass DynamicTracer {\n  private tracerProvider: TracerProvider;\n  private tracer: Tracer;\n  private readonly instrumentingModuleName: string;\n\n  constructor(instrumentingModuleName: string) {\n    this.instrumentingModuleName = instrumentingModuleName;\n    this.tracerProvider = trace.getTracerProvider();\n    this.tracer = trace.getTracer(instrumentingModuleName);\n  }\n\n  /**\n   * Set a new tracer provider. This updates the underlying tracer instance.\n   * @param provider - The new tracer provider to use\n   */\n  setProvider(provider: TracerProvider): void {\n    this.tracerProvider = provider;\n    this.tracer = this.tracerProvider.getTracer(this.instrumentingModuleName);\n  }\n\n  /**\n   * Get the underlying OpenTelemetry tracer.\n   * Use this to access the full Tracer API when needed.\n   */\n  getTracer(): Tracer {\n    return this.tracer;\n  }\n\n  /**\n   * Start a span manually (without making it active).\n   * You must call span.end() when done.\n   *\n   * @param options - Span configuration including name\n   * @returns The created span\n   */\n  startSpan(options: StartSpanOptions): Span {\n    const ctx = options.context || otelContext.active();\n    const span = this.tracer.startSpan(\n      options.name,\n      {\n        attributes: options.attributes,\n      },\n      ctx,\n    );\n\n    return span;\n  }\n\n  /**\n   * Start a new span and make it active in the current context.\n   * The span will automatically be ended when the provided function completes (unless endOnExit=false).\n   *\n   * @param fn - The function to execute within the span context\n   * @param options - Span configuration including name\n   * @returns The result of the provided function\n   */\n  async startActiveSpan<T>(fn: (span: Span) => Promise<T>, options: StartSpanOptions): Promise<T> {\n    const ctx = options.context || otelContext.active();\n    const endOnExit = options.endOnExit === undefined ? true : options.endOnExit; // default true\n    const opts: SpanOptions = { attributes: options.attributes };\n\n    // Directly return the tracer's startActiveSpan result - it handles async correctly\n    return await this.tracer.startActiveSpan(options.name, opts, ctx, async (span) => {\n      try {\n        return await fn(span);\n      } finally {\n        if (endOnExit) {\n          span.end();\n        }\n      }\n    });\n  }\n\n  /**\n   * Synchronous version of startActiveSpan for non-async operations.\n   *\n   * @param fn - The function to execute within the span context\n   * @param options - Span configuration including name\n   * @returns The result of the provided function\n   */\n  startActiveSpanSync<T>(fn: (span: Span) => T, options: StartSpanOptions): T {\n    const ctx = options.context || otelContext.active();\n    const endOnExit = options.endOnExit === undefined ? true : options.endOnExit; // default true\n    const opts: SpanOptions = { attributes: options.attributes };\n\n    return this.tracer.startActiveSpan(options.name, opts, ctx, (span) => {\n      try {\n        return fn(span);\n      } finally {\n        if (endOnExit) {\n          span.end();\n        }\n      }\n    });\n  }\n}\n\n/**\n * The global tracer instance used throughout the agents framework.\n * This tracer can have its provider updated at runtime via setTracerProvider().\n */\nexport const tracer = new DynamicTracer('livekit-agents');\n\nclass MetadataSpanProcessor implements SpanProcessor {\n  private metadata: Attributes;\n\n  constructor(metadata: Attributes) {\n    this.metadata = metadata;\n  }\n\n  onStart(span: Span, _parentContext: Context): void {\n    span.setAttributes(this.metadata);\n  }\n\n  onEnd(_span: ReadableSpan): void {}\n\n  shutdown(): Promise<void> {\n    return Promise.resolve();\n  }\n\n  forceFlush(): Promise<void> {\n    return Promise.resolve();\n  }\n}\n\n/**\n * Set the tracer provider for the livekit-agents framework.\n * This should be called before agent session start if using custom tracer providers.\n *\n * @param provider - The tracer provider to use (must be a NodeTracerProvider)\n * @param options - Optional configuration with metadata property to inject into all spans\n *\n * @example\n * ```typescript\n * import { NodeTracerProvider } from '@opentelemetry/sdk-trace-node';\n * import { setTracerProvider } from '@livekit/agents/telemetry';\n *\n * const provider = new NodeTracerProvider();\n * setTracerProvider(provider, {\n *   metadata: { room_id: 'room123', job_id: 'job456' }\n * });\n * ```\n */\nexport function setTracerProvider(\n  provider: NodeTracerProvider,\n  options?: { metadata?: Attributes },\n): void {\n  if (options?.metadata) {\n    provider.addSpanProcessor(new MetadataSpanProcessor(options.metadata));\n  }\n\n  tracer.setProvider(provider);\n}\n\n/**\n * Setup OpenTelemetry tracer for LiveKit Cloud observability.\n * This configures OTLP exporters to send traces to LiveKit Cloud.\n *\n * @param options - Configuration for cloud tracer with roomId, jobId, and cloudHostname properties\n *\n * @internal\n */\nexport async function setupCloudTracer(options: {\n  roomId: string;\n  jobId: string;\n  cloudHostname: string;\n}): Promise<void> {\n  const { roomId, jobId, cloudHostname } = options;\n\n  const apiKey = process.env.LIVEKIT_API_KEY;\n  const apiSecret = process.env.LIVEKIT_API_SECRET;\n\n  if (!apiKey || !apiSecret) {\n    throw new Error('LIVEKIT_API_KEY and LIVEKIT_API_SECRET must be set for cloud tracing');\n  }\n\n  const token = new AccessToken(apiKey, apiSecret, {\n    identity: 'livekit-agents-telemetry',\n    ttl: '6h',\n  });\n  token.addObservabilityGrant({ write: true });\n\n  try {\n    const jwt = await token.toJwt();\n\n    const headers = {\n      Authorization: `Bearer ${jwt}`,\n    };\n\n    const metadata: Attributes = {\n      room_id: roomId,\n      job_id: jobId,\n    };\n\n    const resource = new Resource({\n      [ATTR_SERVICE_NAME]: 'livekit-agents',\n      room_id: roomId,\n      job_id: jobId,\n    });\n\n    // Configure OTLP exporter to send traces to LiveKit Cloud\n    const spanExporter = new OTLPTraceExporter({\n      url: `https://${cloudHostname}/observability/traces/otlp/v0`,\n      headers,\n      compression: CompressionAlgorithm.GZIP,\n    });\n\n    const tracerProvider = new NodeTracerProvider({\n      resource,\n      spanProcessors: [new MetadataSpanProcessor(metadata), new BatchSpanProcessor(spanExporter)],\n    });\n    tracerProvider.register();\n\n    setTracerProvider(tracerProvider);\n\n    // Initialize standalone Pino cloud exporter (no OTEL SDK dependency)\n    initPinoCloudExporter({\n      cloudHostname,\n      roomId,\n      jobId,\n    });\n\n    enableOtelLogging();\n  } catch (error) {\n    console.error('Failed to setup cloud tracer:', error);\n    throw error;\n  }\n}\n\n/**\n * Flush all pending Pino logs to ensure they are exported.\n * Call this before session/job ends to ensure all logs are sent.\n *\n * @internal\n */\nexport async function flushOtelLogs(): Promise<void> {\n  await flushPinoLogs();\n}\n\n/**\n * Convert ChatItem to proto-compatible dictionary format.\n * TODO: Use actual agent_session proto types once @livekit/protocol v1.43.1+ is published\n */\nfunction chatItemToProto(item: ChatItem): Record<string, any> {\n  const itemDict: Record<string, any> = {};\n\n  if (item.type === 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 msg.metrics = {};\n    //   if (metrics.started_speaking_at) {\n    //     msg.metrics.startedSpeakingAt = toRFC3339(metrics.started_speaking_at);\n    //   }\n    //   if (metrics.stopped_speaking_at) {\n    //     msg.metrics.stoppedSpeakingAt = toRFC3339(metrics.stopped_speaking_at);\n    //   }\n    //   if (metrics.transcription_delay !== undefined) {\n    //     msg.metrics.transcriptionDelay = metrics.transcription_delay;\n    //   }\n    //   if (metrics.end_of_turn_delay !== undefined) {\n    //     msg.metrics.endOfTurnDelay = metrics.end_of_turn_delay;\n    //   }\n    //   if (metrics.on_user_turn_completed_delay !== undefined) {\n    //     msg.metrics.onUserTurnCompletedDelay = metrics.on_user_turn_completed_delay;\n    //   }\n    //   if (metrics.llm_node_ttft !== undefined) {\n    //     msg.metrics.llmNodeTtft = metrics.llm_node_ttft;\n    //   }\n    //   if (metrics.tts_node_ttfb !== undefined) {\n    //     msg.metrics.ttsNodeTtfb = metrics.tts_node_ttfb;\n    //  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This updates the underlying tracer instance.\n   * @param provider - The new tracer provider to use\n   */\n  setProvider(provider: TracerProvider): void {\n    this.tracerProvider = provider;\n    this.tracer = this.tracerProvider.getTracer(this.instrumentingModuleName);\n  }\n\n  /**\n   * Get the underlying OpenTelemetry tracer.\n   * Use this to access the full Tracer API when needed.\n   */\n  getTracer(): Tracer {\n    return this.tracer;\n  }\n\n  /**\n   * Start a span manually (without making it active).\n   * You must call span.end() when done.\n   *\n   * @param options - Span configuration including name\n   * @returns The created span\n   */\n  startSpan(options: StartSpanOptions): Span {\n    const ctx = options.context || otelContext.active();\n    const span = this.tracer.startSpan(\n      options.name,\n      {\n        attributes: options.attributes,\n      },\n      ctx,\n    );\n\n    return span;\n  }\n\n  /**\n   * Start a new span and make it active in the 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NodeTracerProvider();\n * setTracerProvider(provider, {\n *   metadata: { room_id: 'room123', job_id: 'job456' }\n * });\n * ```\n */\nexport function setTracerProvider(\n  provider: NodeTracerProvider,\n  options?: { metadata?: Attributes },\n): void {\n  if (options?.metadata) {\n    provider.addSpanProcessor(new MetadataSpanProcessor(options.metadata));\n  }\n\n  tracer.setProvider(provider);\n}\n\n/**\n * Setup OpenTelemetry tracer for LiveKit Cloud observability.\n * This configures OTLP exporters to send traces to LiveKit Cloud.\n *\n * @param options - Configuration for cloud tracer with roomId, jobId, and cloudHostname properties\n *\n * @internal\n */\nexport async function setupCloudTracer(options: {\n  roomId: string;\n  jobId: string;\n  cloudHostname: string;\n}): Promise<void> {\n  const { roomId, jobId, cloudHostname } = options;\n\n  const apiKey = process.env.LIVEKIT_API_KEY;\n  const apiSecret = process.env.LIVEKIT_API_SECRET;\n\n  if (!apiKey || !apiSecret) {\n    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package/dist/telemetry/traces.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"traces.d.ts","sourceRoot":"","sources":["../../src/telemetry/traces.ts"],"names":[],"mappings":"AAIA,OAAO,EACL,KAAK,UAAU,EACf,KAAK,OAAO,EACZ,KAAK,IAAI,EAET,KAAK,MAAM,EACX,KAAK,cAAc,EAGpB,MAAM,oBAAoB,CAAC;AAM5B,OAAO,EAAsB,kBAAkB,EAAE,MAAM,+BAA+B,CAAC;AAOvF,OAAO,KAAK,EAAE,aAAa,EAAE,MAAM,oBAAoB,CAAC;AAIxD,MAAM,WAAW,gBAAgB;IAC/B,uBAAuB;IACvB,IAAI,EAAE,MAAM,CAAC;IACb,mDAAmD;IACnD,OAAO,CAAC,EAAE,OAAO,CAAC;IAClB,mDAAmD;IACnD,UAAU,CAAC,EAAE,UAAU,CAAC;IACxB,sEAAsE;IACtE,SAAS,CAAC,EAAE,OAAO,CAAC;CACrB;AAED;;GAEG;AACH,cAAM,aAAa;IACjB,OAAO,CAAC,cAAc,CAAiB;IACvC,OAAO,CAAC,MAAM,CAAS;IACvB,OAAO,CAAC,QAAQ,CAAC,uBAAuB,CAAS;gBAErC,uBAAuB,EAAE,MAAM;IAM3C;;;OAGG;IACH,WAAW,CAAC,QAAQ,EAAE,cAAc,GAAG,IAAI;IAK3C;;;OAGG;IACH,SAAS,IAAI,MAAM;IAInB;;;;;;OAMG;IACH,SAAS,CAAC,OAAO,EAAE,gBAAgB,GAAG,IAAI;IAa1C;;;;;;;OAOG;IACG,eAAe,CAAC,CAAC,EAAE,EAAE,EAAE,CAAC,IAAI,EAAE,IAAI,KAAK,OAAO,CAAC,CAAC,CAAC,EAAE,OAAO,EAAE,gBAAgB,GAAG,OAAO,CAAC,CAAC,CAAC;IAiB/F;;;;;;OAMG;IACH,mBAAmB,CAAC,CAAC,EAAE,EAAE,EAAE,CAAC,IAAI,EAAE,IAAI,KAAK,CAAC,EAAE,OAAO,EAAE,gBAAgB,GAAG,CAAC;CAe5E;AAED;;;GAGG;AACH,eAAO,MAAM,MAAM,eAAsC,CAAC;AAwB1D;;;;;;;;;;;;;;;;;GAiBG;AACH,wBAAgB,iBAAiB,CAC/B,QAAQ,EAAE,kBAAkB,EAC5B,OAAO,CAAC,EAAE;IAAE,QAAQ,CAAC,EAAE,UAAU,CAAA;CAAE,GAClC,IAAI,CAMN;AAED;;;;;;;GAOG;AACH,wBAAsB,gBAAgB,CAAC,OAAO,EAAE;IAC9C,MAAM,EAAE,MAAM,CAAC;IACf,KAAK,EAAE,MAAM,CAAC;IACd,aAAa,EAAE,MAAM,CAAC;CACvB,GAAG,OAAO,CAAC,IAAI,CAAC,CA6DhB;AAED;;;;;GAKG;AACH,wBAAsB,aAAa,IAAI,OAAO,CAAC,IAAI,CAAC,CAEnD;AA8HD;;;GAGG;AACH,wBAAsB,mBAAmB,CAAC,OAAO,EAAE;IACjD,SAAS,EAAE,MAAM,CAAC;IAClB,aAAa,EAAE,MAAM,CAAC;IACtB,MAAM,EAAE,aAAa,CAAC;CACvB,GAAG,OAAO,CAAC,IAAI,CAAC,CA4KhB"}

package/dist/telemetry/traces.js

@@ -11,6 +11,7 @@ import { BatchSpanProcessor, NodeTracerProvider } from "@opentelemetry/sdk-trace
 import { ATTR_SERVICE_NAME } from "@opentelemetry/semantic-conventions";
 import FormData from "form-data";
 import { AccessToken } from "livekit-server-sdk";
+import fs from "node:fs/promises";
 import { enableOtelLogging } from "../log.js";
 import { SimpleOTLPHttpLogExporter } from "./otel_http_exporter.js";
 import { flushPinoLogs, initPinoCloudExporter } from "./pino_otel_transport.js";
@@ -302,13 +303,14 @@ async function uploadSessionReport(options) {
   token.addObservabilityGrant({ write: true });
   const jwt = await token.toJwt();
   const formData = new FormData();
+  const audioStartTime = report.audioRecordingStartedAt ?? 0;
   const headerMsg = new MetricsRecordingHeader({
     roomId: report.roomId,
     duration: BigInt(0),
     // TODO: Calculate actual duration from report
     startTime: {
-      seconds: BigInt(Math.floor(report.timestamp / 1e3)),
-      nanos: Math.floor(report.timestamp % 1e3 * 1e6)
+      seconds: BigInt(Math.floor(audioStartTime / 1e3)),
+      nanos: Math.floor(audioStartTime % 1e3 * 1e6)
     }
   });
   const headerBytes = Buffer.from(headerMsg.toBinary());
@@ -332,6 +334,25 @@ async function uploadSessionReport(options) {
       "Content-Length": chatHistoryBuffer.length.toString()
     }
   });
+  if (report.audioRecordingPath && report.audioRecordingStartedAt) {
+    let audioBytes;
+    try {
+      audioBytes = await fs.readFile(report.audioRecordingPath);
+    } catch {
+      audioBytes = Buffer.alloc(0);
+    }
+    if (audioBytes.length > 0) {
+      formData.append("audio", audioBytes, {
+        filename: "recording.ogg",
+        contentType: "audio/ogg",
+        knownLength: audioBytes.length,
+        header: {
+          "Content-Type": "audio/ogg",
+          "Content-Length": audioBytes.length.toString()
+        }
+      });
+    }
+  }
   return new Promise((resolve, reject) => {
     formData.submit(
       {