@framers/agentos 0.1.125 → 0.1.127

package/dist/hearing/providers/BuiltInAdaptiveVadProvider.d.ts

@@ -0,0 +1,179 @@
+import type { AdaptiveVADConfig } from '../AdaptiveVAD.js';
+import type { CalibrationConfig } from '../EnvironmentalCalibrator.js';
+import type { SpeechVadDecision, SpeechVadProvider } from '../../speech/types.js';
+/**
+ * Configuration for the {@link BuiltInAdaptiveVadProvider}.
+ *
+ * All fields are optional — sensible defaults are applied for 16kHz mono
+ * audio with 20ms frames (the standard voice pipeline configuration).
+ *
+ * @see {@link BuiltInAdaptiveVadProvider} for usage examples
+ * See `AdaptiveVADConfig` for detailed VAD tuning parameters.
+ * See `CalibrationConfig` for environmental calibration settings.
+ */
+export interface BuiltInAdaptiveVadProviderConfig {
+    /**
+     * Audio sample rate in Hz. Must match the actual audio stream.
+     * @default 16_000 (16 kHz, standard for voice processing)
+     */
+    sampleRate?: number;
+    /**
+     * Duration of each audio frame in milliseconds. Determines the number
+     * of samples expected in each `processFrame()` call.
+     * At 16kHz with 20ms frames, each frame is 320 samples.
+     * @default 20
+     */
+    frameDurationMs?: number;
+    /**
+     * Environmental calibration settings. Controls how the noise floor and
+     * spectral profile are estimated from ambient audio.
+     *
+     * See `CalibrationConfig` for available options.
+     */
+    calibration?: CalibrationConfig;
+    /**
+     * VAD algorithm configuration. Controls speech detection thresholds,
+     * minimum speech duration, hangover period, etc.
+     *
+     * See `AdaptiveVADConfig` for available options.
+     */
+    vad?: AdaptiveVADConfig;
+}
+/**
+ * Built-in voice activity detection (VAD) provider backed by the
+ * `AdaptiveVAD` engine and `EnvironmentalCalibrator`.
+ *
+ * This is the default VAD provider in AgentOS and requires no external
+ * dependencies or API keys. It operates entirely locally on raw audio frames.
+ *
+ * ## How It Works
+ *
+ * 1. The `EnvironmentalCalibrator` continuously estimates the ambient
+ *    noise floor and spectral profile from incoming audio frames.
+ * 2. The `AdaptiveVAD` uses the calibrator's noise profile to set
+ *    dynamic thresholds for speech detection — louder environments get
+ *    higher thresholds to avoid false positives.
+ * 3. Each `processFrame()` call returns a {@link SpeechVadDecision} with
+ *    `isSpeech`, `confidence`, the raw VAD result, and the current noise profile.
+ *
+ * ## Configuration Defaults
+ *
+ * - Sample rate: 16 kHz (standard for voice pipelines)
+ * - Frame duration: 20ms (320 samples per frame)
+ * - VAD and calibration: Use sensible defaults from the underlying engines
+ *
+ * @see {@link BuiltInAdaptiveVadProviderConfig} for configuration options
+ * See `AdaptiveVAD` for the underlying VAD algorithm.
+ * See `EnvironmentalCalibrator` for the noise profiling engine.
+ *
+ * @example
+ * ```ts
+ * const vad = new BuiltInAdaptiveVadProvider({
+ *   sampleRate: 16_000,
+ *   frameDurationMs: 20,
+ *   vad: { minSpeechDurationMs: 100 },
+ * });
+ *
+ * const frame = new Float32Array(320); // 20ms at 16kHz
+ * // ... fill frame with audio samples ...
+ * const decision = vad.processFrame(frame);
+ * if (decision.isSpeech) {
+ *   console.log(`Speech detected (confidence: ${decision.confidence})`);
+ * }
+ * ```
+ */
+export declare class BuiltInAdaptiveVadProvider implements SpeechVadProvider {
+    /** Unique provider identifier used for registration and resolution. */
+    readonly id = "agentos-adaptive-vad";
+    /** Human-readable display name for UI and logging. */
+    readonly displayName = "AgentOS Adaptive VAD";
+    /**
+     * Environmental calibrator that estimates the ambient noise profile.
+     * Shared with the VAD engine for dynamic threshold adaptation.
+     */
+    private readonly calibrator;
+    /**
+     * Adaptive VAD engine that performs frame-by-frame speech detection
+     * using the calibrator's noise profile for dynamic thresholding.
+     */
+    private readonly vad;
+    /**
+     * Creates a new BuiltInAdaptiveVadProvider.
+     *
+     * Initializes both the environmental calibrator and the adaptive VAD
+     * engine with the provided or default configuration.
+     *
+     * @param config - Optional VAD configuration. All fields default to
+     *   standard values suitable for 16kHz mono voice audio.
+     *
+     * @example
+     * ```ts
+     * // Default configuration (16kHz, 20ms frames)
+     * const vad = new BuiltInAdaptiveVadProvider();
+     *
+     * // Custom configuration
+     * const vad = new BuiltInAdaptiveVadProvider({
+     *   sampleRate: 48_000,
+     *   frameDurationMs: 10,
+     *   vad: { minSpeechDurationMs: 200 },
+     * });
+     * ```
+     */
+    constructor(config?: BuiltInAdaptiveVadProviderConfig);
+    /**
+     * Process a single audio frame and return a speech/non-speech decision.
+     *
+     * This method must be called sequentially with consecutive audio frames.
+     * The VAD maintains internal state (speech onset tracking, hangover counters)
+     * that depends on temporal continuity between frames.
+     *
+     * @param frame - A Float32Array of audio samples for one frame. The expected
+     *   length is `sampleRate * frameDurationMs / 1000` (e.g. 320 for 16kHz/20ms).
+     *   Samples should be normalized to the range [-1.0, 1.0].
+     * @returns A decision object with `isSpeech`, `confidence`, the raw VAD result,
+     *   and the current environmental noise profile.
+     *
+     * @example
+     * ```ts
+     * const frame = new Float32Array(320);
+     * // ... fill with audio samples ...
+     * const decision = vad.processFrame(frame);
+     * console.log(decision.isSpeech, decision.confidence);
+     * ```
+     */
+    processFrame(frame: Float32Array): SpeechVadDecision;
+    /**
+     * Reset the VAD state for a new audio session.
+     *
+     * Clears internal counters (speech onset tracking, hangover timers) so the
+     * VAD starts fresh. Should be called when starting a new conversation turn
+     * or after a significant audio gap. Does NOT reset the environmental
+     * calibrator — the noise profile persists across resets.
+     *
+     * @example
+     * ```ts
+     * // Start a new conversation turn
+     * vad.reset();
+     * ```
+     */
+    reset(): void;
+    /**
+     * Returns the current environmental noise profile estimated by the calibrator.
+     *
+     * The noise profile includes the estimated noise floor RMS, spectral shape,
+     * and confidence metrics. Returns `null` if insufficient audio has been
+     * processed for a reliable estimate.
+     *
+     * @returns The current noise profile, or `null` if not yet calibrated.
+     *
+     * @example
+     * ```ts
+     * const profile = vad.getNoiseProfile();
+     * if (profile) {
+     *   console.log(`Noise floor: ${profile.noiseFloorRms}`);
+     * }
+     * ```
+     */
+    getNoiseProfile(): import("../EnvironmentalCalibrator.js").NoiseProfile | null;
+}
+//# sourceMappingURL=BuiltInAdaptiveVadProvider.d.ts.map

package/dist/hearing/providers/BuiltInAdaptiveVadProvider.d.ts.map

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package/dist/hearing/providers/BuiltInAdaptiveVadProvider.js

@@ -0,0 +1,148 @@
+import { AdaptiveVAD } from '../AdaptiveVAD.js';
+import { EnvironmentalCalibrator } from '../EnvironmentalCalibrator.js';
+/**
+ * Built-in voice activity detection (VAD) provider backed by the
+ * `AdaptiveVAD` engine and `EnvironmentalCalibrator`.
+ *
+ * This is the default VAD provider in AgentOS and requires no external
+ * dependencies or API keys. It operates entirely locally on raw audio frames.
+ *
+ * ## How It Works
+ *
+ * 1. The `EnvironmentalCalibrator` continuously estimates the ambient
+ *    noise floor and spectral profile from incoming audio frames.
+ * 2. The `AdaptiveVAD` uses the calibrator's noise profile to set
+ *    dynamic thresholds for speech detection — louder environments get
+ *    higher thresholds to avoid false positives.
+ * 3. Each `processFrame()` call returns a {@link SpeechVadDecision} with
+ *    `isSpeech`, `confidence`, the raw VAD result, and the current noise profile.
+ *
+ * ## Configuration Defaults
+ *
+ * - Sample rate: 16 kHz (standard for voice pipelines)
+ * - Frame duration: 20ms (320 samples per frame)
+ * - VAD and calibration: Use sensible defaults from the underlying engines
+ *
+ * @see {@link BuiltInAdaptiveVadProviderConfig} for configuration options
+ * See `AdaptiveVAD` for the underlying VAD algorithm.
+ * See `EnvironmentalCalibrator` for the noise profiling engine.
+ *
+ * @example
+ * ```ts
+ * const vad = new BuiltInAdaptiveVadProvider({
+ *   sampleRate: 16_000,
+ *   frameDurationMs: 20,
+ *   vad: { minSpeechDurationMs: 100 },
+ * });
+ *
+ * const frame = new Float32Array(320); // 20ms at 16kHz
+ * // ... fill frame with audio samples ...
+ * const decision = vad.processFrame(frame);
+ * if (decision.isSpeech) {
+ *   console.log(`Speech detected (confidence: ${decision.confidence})`);
+ * }
+ * ```
+ */
+export class BuiltInAdaptiveVadProvider {
+    /**
+     * Creates a new BuiltInAdaptiveVadProvider.
+     *
+     * Initializes both the environmental calibrator and the adaptive VAD
+     * engine with the provided or default configuration.
+     *
+     * @param config - Optional VAD configuration. All fields default to
+     *   standard values suitable for 16kHz mono voice audio.
+     *
+     * @example
+     * ```ts
+     * // Default configuration (16kHz, 20ms frames)
+     * const vad = new BuiltInAdaptiveVadProvider();
+     *
+     * // Custom configuration
+     * const vad = new BuiltInAdaptiveVadProvider({
+     *   sampleRate: 48_000,
+     *   frameDurationMs: 10,
+     *   vad: { minSpeechDurationMs: 200 },
+     * });
+     * ```
+     */
+    constructor(config = {}) {
+        /** Unique provider identifier used for registration and resolution. */
+        this.id = 'agentos-adaptive-vad';
+        /** Human-readable display name for UI and logging. */
+        this.displayName = 'AgentOS Adaptive VAD';
+        this.calibrator = new EnvironmentalCalibrator({
+            sampleRate: config.sampleRate ?? 16000,
+            ...(config.calibration ?? {}),
+        });
+        this.vad = new AdaptiveVAD(config.vad ?? {}, this.calibrator, config.frameDurationMs ?? 20);
+    }
+    /**
+     * Process a single audio frame and return a speech/non-speech decision.
+     *
+     * This method must be called sequentially with consecutive audio frames.
+     * The VAD maintains internal state (speech onset tracking, hangover counters)
+     * that depends on temporal continuity between frames.
+     *
+     * @param frame - A Float32Array of audio samples for one frame. The expected
+     *   length is `sampleRate * frameDurationMs / 1000` (e.g. 320 for 16kHz/20ms).
+     *   Samples should be normalized to the range [-1.0, 1.0].
+     * @returns A decision object with `isSpeech`, `confidence`, the raw VAD result,
+     *   and the current environmental noise profile.
+     *
+     * @example
+     * ```ts
+     * const frame = new Float32Array(320);
+     * // ... fill with audio samples ...
+     * const decision = vad.processFrame(frame);
+     * console.log(decision.isSpeech, decision.confidence);
+     * ```
+     */
+    processFrame(frame) {
+        const result = this.vad.processFrame(frame);
+        return {
+            isSpeech: result.isSpeech,
+            confidence: result.confidence,
+            result,
+            profile: this.calibrator.getCurrentProfile(),
+        };
+    }
+    /**
+     * Reset the VAD state for a new audio session.
+     *
+     * Clears internal counters (speech onset tracking, hangover timers) so the
+     * VAD starts fresh. Should be called when starting a new conversation turn
+     * or after a significant audio gap. Does NOT reset the environmental
+     * calibrator — the noise profile persists across resets.
+     *
+     * @example
+     * ```ts
+     * // Start a new conversation turn
+     * vad.reset();
+     * ```
+     */
+    reset() {
+        this.vad.resetState();
+    }
+    /**
+     * Returns the current environmental noise profile estimated by the calibrator.
+     *
+     * The noise profile includes the estimated noise floor RMS, spectral shape,
+     * and confidence metrics. Returns `null` if insufficient audio has been
+     * processed for a reliable estimate.
+     *
+     * @returns The current noise profile, or `null` if not yet calibrated.
+     *
+     * @example
+     * ```ts
+     * const profile = vad.getNoiseProfile();
+     * if (profile) {
+     *   console.log(`Noise floor: ${profile.noiseFloorRms}`);
+     * }
+     * ```
+     */
+    getNoiseProfile() {
+        return this.calibrator.getCurrentProfile();
+    }
+}
+//# sourceMappingURL=BuiltInAdaptiveVadProvider.js.map

package/dist/hearing/providers/BuiltInAdaptiveVadProvider.js.map

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package/dist/hearing/providers/DeepgramBatchSTTProvider.d.ts

@@ -0,0 +1,141 @@
+import type { SpeechAudioInput, SpeechToTextProvider, SpeechTranscriptionOptions, SpeechTranscriptionResult } from '../../speech/types.js';
+/**
+ * Configuration for the {@link DeepgramBatchSTTProvider}.
+ *
+ * @see {@link DeepgramBatchSTTProvider} for usage examples
+ */
+export interface DeepgramBatchSTTProviderConfig {
+    /**
+     * Deepgram API key used for authentication.
+     * Sent as `Authorization: Token <apiKey>` in the request header.
+     */
+    apiKey: string;
+    /**
+     * Deepgram model to use for transcription.
+     * See https://developers.deepgram.com/docs/models for available models.
+     * @default 'nova-2'
+     */
+    model?: string;
+    /**
+     * BCP-47 language code, e.g. `'en-US'`, `'fr-FR'`, `'de-DE'`.
+     * When omitted, Deepgram applies automatic language detection.
+     * @default 'en-US' (set at transcribe-time if not configured here)
+     */
+    language?: string;
+    /**
+     * Custom fetch implementation for dependency injection in tests.
+     * When omitted, the global `fetch` is used. This allows tests to
+     * intercept HTTP calls without mocking globals.
+     * @default globalThis.fetch
+     */
+    fetchImpl?: typeof fetch;
+}
+/**
+ * Speech-to-text provider that uses the Deepgram batch (pre-recorded) REST API.
+ *
+ * ## REST API Contract
+ *
+ * - **Endpoint:** `POST https://api.deepgram.com/v1/listen`
+ * - **Authentication:** `Authorization: Token <apiKey>` header
+ * - **Content-Type:** Set to the audio's MIME type (e.g. `audio/wav`)
+ * - **Body:** Raw audio bytes sent directly (no multipart form)
+ * - **Query parameters:** `model`, `punctuate`, `diarize`, `language`
+ * - **Response:** JSON containing `results.channels[].alternatives[]` with
+ *   transcript text, confidence scores, and optional word-level timing
+ *
+ * ## Word-Level Diarization Mapping
+ *
+ * When `enableSpeakerDiarization` is `true`, the `diarize=true` query parameter
+ * is set. Deepgram then includes a `speaker` field (zero-based integer index) on
+ * each word in the response. These speaker indices are preserved through the
+ * `wordsToSegments()` mapping into the normalized result.
+ *
+ * ## Error Handling
+ *
+ * Non-2xx responses from Deepgram trigger an `Error` with the HTTP status code
+ * and response body text included in the message for debugging. Network-level
+ * errors (DNS failures, timeouts) propagate as-is from the fetch implementation.
+ *
+ * Streaming is NOT supported by this provider — use a Deepgram WebSocket adapter
+ * for real-time transcription.
+ *
+ * @see {@link DeepgramBatchSTTProviderConfig} for configuration options
+ * See `wordsToSegments()` for the word-to-segment mapping logic.
+ *
+ * @example
+ * ```ts
+ * const provider = new DeepgramBatchSTTProvider({
+ *   apiKey: process.env.DEEPGRAM_API_KEY!,
+ *   model: 'nova-2',
+ * });
+ * const result = await provider.transcribe(
+ *   { data: audioBuffer, mimeType: 'audio/wav' },
+ *   { enableSpeakerDiarization: true },
+ * );
+ * console.log(result.text);
+ * console.log(result.segments?.map(s => `[Speaker ${s.speaker}] ${s.text}`));
+ * ```
+ */
+export declare class DeepgramBatchSTTProvider implements SpeechToTextProvider {
+    private readonly config;
+    /** Unique provider identifier used for registration and resolution. */
+    readonly id = "deepgram-batch";
+    /** Human-readable display name for UI and logging. */
+    readonly displayName = "Deepgram (Batch)";
+    /** This provider uses synchronous HTTP requests, not WebSocket streaming. */
+    readonly supportsStreaming = false;
+    /** Fetch implementation — injected for testability, defaults to global fetch. */
+    private readonly fetchImpl;
+    /**
+     * Creates a new DeepgramBatchSTTProvider.
+     *
+     * @param config - Provider configuration including API key and optional defaults.
+     *
+     * @example
+     * ```ts
+     * const provider = new DeepgramBatchSTTProvider({
+     *   apiKey: 'dg-xxxx',
+     *   model: 'nova-2',
+     *   language: 'en-US',
+     * });
+     * ```
+     */
+    constructor(config: DeepgramBatchSTTProviderConfig);
+    /**
+     * Returns the human-readable provider name.
+     *
+     * @returns The display name string `'Deepgram (Batch)'`.
+     *
+     * @example
+     * ```ts
+     * provider.getProviderName(); // 'Deepgram (Batch)'
+     * ```
+     */
+    getProviderName(): string;
+    /**
+     * Transcribes an audio buffer using the Deepgram pre-recorded API.
+     *
+     * Sends the raw audio bytes as the request body (not multipart form) with
+     * the appropriate Content-Type header. The response is parsed and normalized
+     * into a {@link SpeechTranscriptionResult}.
+     *
+     * @param audio - Raw audio data and associated metadata (buffer, MIME type,
+     *   duration). The `data` buffer is sent directly as the request body.
+     * @param options - Optional transcription settings. Supports `model`,
+     *   `language`, and `enableSpeakerDiarization` overrides.
+     * @returns A promise resolving to the normalized transcription result with
+     *   text, confidence, timing, and optional speaker-attributed segments.
+     * @throws {Error} When the Deepgram API returns a non-2xx status code.
+     *   The error message includes the HTTP status and response body for debugging.
+     *
+     * @example
+     * ```ts
+     * const result = await provider.transcribe(
+     *   { data: wavBuffer, mimeType: 'audio/wav', durationSeconds: 5.2 },
+     *   { language: 'fr-FR', enableSpeakerDiarization: true },
+     * );
+     * ```
+     */
+    transcribe(audio: SpeechAudioInput, options?: SpeechTranscriptionOptions): Promise<SpeechTranscriptionResult>;
+}
+//# sourceMappingURL=DeepgramBatchSTTProvider.d.ts.map

package/dist/hearing/providers/DeepgramBatchSTTProvider.d.ts.map

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package/dist/hearing/providers/DeepgramBatchSTTProvider.js.map

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package/dist/hearing/providers/OpenAIWhisperSpeechToTextProvider.d.ts

@@ -0,0 +1,128 @@
+import type { SpeechAudioInput, SpeechToTextProvider, SpeechTranscriptionOptions, SpeechTranscriptionResult } from '../../speech/types.js';
+/**
+ * Configuration for the {@link OpenAIWhisperSpeechToTextProvider}.
+ *
+ * @see {@link OpenAIWhisperSpeechToTextProvider} for usage examples
+ * @see https://platform.openai.com/docs/api-reference/audio/createTranscription
+ */
+export interface OpenAIWhisperSpeechToTextProviderConfig {
+    /**
+     * OpenAI API key used for authentication.
+     * Sent as `Authorization: Bearer <apiKey>` in the request header.
+     */
+    apiKey: string;
+    /**
+     * Base URL for the OpenAI API. Override for proxies, Azure OpenAI, or
+     * compatible third-party endpoints.
+     * @default 'https://api.openai.com/v1'
+     */
+    baseUrl?: string;
+    /**
+     * Default Whisper model to use for transcription.
+     * @default 'whisper-1'
+     */
+    model?: string;
+    /**
+     * Custom fetch implementation for dependency injection in tests.
+     * @default globalThis.fetch
+     */
+    fetchImpl?: typeof fetch;
+}
+/**
+ * Speech-to-text provider that uses the OpenAI Whisper transcription API.
+ *
+ * ## API Contract
+ *
+ * - **Endpoint:** `POST {baseUrl}/audio/transcriptions`
+ * - **Authentication:** `Authorization: Bearer <apiKey>`
+ * - **Content-Type:** `multipart/form-data` (FormData with file blob)
+ * - **Response format:** Controlled by the `response_format` field; defaults
+ *   to `verbose_json` which includes segments, language detection, and duration.
+ *
+ * ## Supported Response Formats
+ *
+ * - `verbose_json` — Full JSON with segments, duration, and language (default)
+ * - `json` — Minimal JSON with just the text
+ * - `text` — Plain text response (no JSON)
+ * - `srt` — SubRip subtitle format
+ * - `vtt` — WebVTT subtitle format
+ *
+ * When `text`, `srt`, or `vtt` format is used, the response is returned as
+ * plain text and segments are not available.
+ *
+ * @see {@link OpenAIWhisperSpeechToTextProviderConfig} for configuration options
+ * See `normalizeSegments()` for the segment normalization logic.
+ *
+ * @example
+ * ```ts
+ * const provider = new OpenAIWhisperSpeechToTextProvider({
+ *   apiKey: process.env.OPENAI_API_KEY!,
+ *   model: 'whisper-1',
+ * });
+ * const result = await provider.transcribe(
+ *   { data: audioBuffer, mimeType: 'audio/wav', fileName: 'recording.wav' },
+ *   { language: 'en', responseFormat: 'verbose_json' },
+ * );
+ * ```
+ */
+export declare class OpenAIWhisperSpeechToTextProvider implements SpeechToTextProvider {
+    private readonly config;
+    /** Unique provider identifier used for registration and resolution. */
+    readonly id = "openai-whisper";
+    /** Human-readable display name for UI and logging. */
+    readonly displayName = "OpenAI Whisper";
+    /** Whisper API is batch-only; streaming requires a WebSocket adapter. */
+    readonly supportsStreaming = false;
+    /** Fetch implementation — injected for testability, defaults to global fetch. */
+    private readonly fetchImpl;
+    /**
+     * Creates a new OpenAIWhisperSpeechToTextProvider.
+     *
+     * @param config - Provider configuration including API key and optional defaults.
+     *
+     * @example
+     * ```ts
+     * const provider = new OpenAIWhisperSpeechToTextProvider({
+     *   apiKey: 'sk-xxxx',
+     *   baseUrl: 'https://api.openai.com/v1', // default
+     *   model: 'whisper-1', // default
+     * });
+     * ```
+     */
+    constructor(config: OpenAIWhisperSpeechToTextProviderConfig);
+    /**
+     * Returns the human-readable provider name.
+     *
+     * @returns The display name string `'OpenAI Whisper'`.
+     *
+     * @example
+     * ```ts
+     * provider.getProviderName(); // 'OpenAI Whisper'
+     * ```
+     */
+    getProviderName(): string;
+    /**
+     * Transcribes an audio buffer using the OpenAI Whisper API.
+     *
+     * The audio is sent as a multipart form upload with the file, model, and
+     * optional parameters (language, prompt, temperature, response_format).
+     *
+     * @param audio - Raw audio data and metadata. The `data` buffer is wrapped
+     *   in a Blob and sent as a form file field. If `fileName` is not provided,
+     *   a default name is generated from the `format` field.
+     * @param options - Optional transcription settings including language hint,
+     *   context prompt, temperature for sampling, and response format.
+     * @returns A promise resolving to the normalized transcription result.
+     * @throws {Error} When the OpenAI API returns a non-2xx status code.
+     *
+     * @example
+     * ```ts
+     * const result = await provider.transcribe(
+     *   { data: mp3Buffer, mimeType: 'audio/mpeg', fileName: 'voice.mp3' },
+     *   { language: 'fr', prompt: 'Discussion about AI' },
+     * );
+     * ```
+     */
+    transcribe(audio: SpeechAudioInput, options?: SpeechTranscriptionOptions): Promise<SpeechTranscriptionResult>;
+}
+//# sourceMappingURL=OpenAIWhisperSpeechToTextProvider.d.ts.map

package/dist/hearing/providers/OpenAIWhisperSpeechToTextProvider.d.ts.map

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package/dist/hearing/providers/OpenAIWhisperSpeechToTextProvider.js.map

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