@nfreeness/clawdbot 0.124.22101 → 0.124.22103

package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/sessions.d.ts.map

@@ -1 +0,0 @@
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package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/sessions.js

@@ -1,32 +0,0 @@
-"use strict";
-// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
-Object.defineProperty(exports, "__esModule", { value: true });
-exports.Sessions = void 0;
-const resource_1 = require("../../../core/resource.js");
-const headers_1 = require("../../../internal/headers.js");
-class Sessions extends resource_1.APIResource {
-    /**
-     * Create an ephemeral API token for use in client-side applications with the
-     * Realtime API. Can be configured with the same session parameters as the
-     * `session.update` client event.
-     *
-     * It responds with a session object, plus a `client_secret` key which contains a
-     * usable ephemeral API token that can be used to authenticate browser clients for
-     * the Realtime API.
-     *
-     * @example
-     * ```ts
-     * const session =
-     *   await client.beta.realtime.sessions.create();
-     * ```
-     */
-    create(body, options) {
-        return this._client.post('/realtime/sessions', {
-            body,
-            ...options,
-            headers: (0, headers_1.buildHeaders)([{ 'OpenAI-Beta': 'assistants=v2' }, options?.headers]),
-        });
-    }
-}
-exports.Sessions = Sessions;
-//# sourceMappingURL=sessions.js.map

package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/sessions.js.map

@@ -1 +0,0 @@
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package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/sessions.mjs

@@ -1,28 +0,0 @@
-// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
-import { APIResource } from "../../../core/resource.mjs";
-import { buildHeaders } from "../../../internal/headers.mjs";
-export class Sessions extends APIResource {
-    /**
-     * Create an ephemeral API token for use in client-side applications with the
-     * Realtime API. Can be configured with the same session parameters as the
-     * `session.update` client event.
-     *
-     * It responds with a session object, plus a `client_secret` key which contains a
-     * usable ephemeral API token that can be used to authenticate browser clients for
-     * the Realtime API.
-     *
-     * @example
-     * ```ts
-     * const session =
-     *   await client.beta.realtime.sessions.create();
-     * ```
-     */
-    create(body, options) {
-        return this._client.post('/realtime/sessions', {
-            body,
-            ...options,
-            headers: buildHeaders([{ 'OpenAI-Beta': 'assistants=v2' }, options?.headers]),
-        });
-    }
-}
-//# sourceMappingURL=sessions.mjs.map

package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/sessions.mjs.map

@@ -1 +0,0 @@
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package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/transcription-sessions.d.mts

@@ -1,299 +0,0 @@
-import { APIResource } from "../../../core/resource.mjs";
-import { APIPromise } from "../../../core/api-promise.mjs";
-import { RequestOptions } from "../../../internal/request-options.mjs";
-export declare class TranscriptionSessions extends APIResource {
-    /**
-     * Create an ephemeral API token for use in client-side applications with the
-     * Realtime API specifically for realtime transcriptions. Can be configured with
-     * the same session parameters as the `transcription_session.update` client event.
-     *
-     * It responds with a session object, plus a `client_secret` key which contains a
-     * usable ephemeral API token that can be used to authenticate browser clients for
-     * the Realtime API.
-     *
-     * @example
-     * ```ts
-     * const transcriptionSession =
-     *   await client.beta.realtime.transcriptionSessions.create();
-     * ```
-     */
-    create(body: TranscriptionSessionCreateParams, options?: RequestOptions): APIPromise<TranscriptionSession>;
-}
-/**
- * A new Realtime transcription session configuration.
- *
- * When a session is created on the server via REST API, the session object also
- * contains an ephemeral key. Default TTL for keys is 10 minutes. This property is
- * not present when a session is updated via the WebSocket API.
- */
-export interface TranscriptionSession {
-    /**
-     * Ephemeral key returned by the API. Only present when the session is created on
-     * the server via REST API.
-     */
-    client_secret: TranscriptionSession.ClientSecret;
-    /**
-     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
-     */
-    input_audio_format?: string;
-    /**
-     * Configuration of the transcription model.
-     */
-    input_audio_transcription?: TranscriptionSession.InputAudioTranscription;
-    /**
-     * The set of modalities the model can respond with. To disable audio, set this to
-     * ["text"].
-     */
-    modalities?: Array<'text' | 'audio'>;
-    /**
-     * Configuration for turn detection. Can be set to `null` to turn off. Server VAD
-     * means that the model will detect the start and end of speech based on audio
-     * volume and respond at the end of user speech.
-     */
-    turn_detection?: TranscriptionSession.TurnDetection;
-}
-export declare namespace TranscriptionSession {
-    /**
-     * Ephemeral key returned by the API. Only present when the session is created on
-     * the server via REST API.
-     */
-    interface ClientSecret {
-        /**
-         * Timestamp for when the token expires. Currently, all tokens expire after one
-         * minute.
-         */
-        expires_at: number;
-        /**
-         * Ephemeral key usable in client environments to authenticate connections to the
-         * Realtime API. Use this in client-side environments rather than a standard API
-         * token, which should only be used server-side.
-         */
-        value: string;
-    }
-    /**
-     * Configuration of the transcription model.
-     */
-    interface InputAudioTranscription {
-        /**
-         * The language of the input audio. Supplying the input language in
-         * [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`)
-         * format will improve accuracy and latency.
-         */
-        language?: string;
-        /**
-         * The model to use for transcription. Can be `gpt-4o-transcribe`,
-         * `gpt-4o-mini-transcribe`, or `whisper-1`.
-         */
-        model?: 'gpt-4o-transcribe' | 'gpt-4o-mini-transcribe' | 'whisper-1';
-        /**
-         * An optional text to guide the model's style or continue a previous audio
-         * segment. The
-         * [prompt](https://platform.openai.com/docs/guides/speech-to-text#prompting)
-         * should match the audio language.
-         */
-        prompt?: string;
-    }
-    /**
-     * Configuration for turn detection. Can be set to `null` to turn off. Server VAD
-     * means that the model will detect the start and end of speech based on audio
-     * volume and respond at the end of user speech.
-     */
-    interface TurnDetection {
-        /**
-         * Amount of audio to include before the VAD detected speech (in milliseconds).
-         * Defaults to 300ms.
-         */
-        prefix_padding_ms?: number;
-        /**
-         * Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms.
-         * With shorter values the model will respond more quickly, but may jump in on
-         * short pauses from the user.
-         */
-        silence_duration_ms?: number;
-        /**
-         * Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher
-         * threshold will require louder audio to activate the model, and thus might
-         * perform better in noisy environments.
-         */
-        threshold?: number;
-        /**
-         * Type of turn detection, only `server_vad` is currently supported.
-         */
-        type?: string;
-    }
-}
-export interface TranscriptionSessionCreateParams {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    client_secret?: TranscriptionSessionCreateParams.ClientSecret;
-    /**
-     * The set of items to include in the transcription. Current available items are:
-     *
-     * - `item.input_audio_transcription.logprobs`
-     */
-    include?: Array<string>;
-    /**
-     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`. For
-     * `pcm16`, input audio must be 16-bit PCM at a 24kHz sample rate, single channel
-     * (mono), and little-endian byte order.
-     */
-    input_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
-    /**
-     * Configuration for input audio noise reduction. This can be set to `null` to turn
-     * off. Noise reduction filters audio added to the input audio buffer before it is
-     * sent to VAD and the model. Filtering the audio can improve VAD and turn
-     * detection accuracy (reducing false positives) and model performance by improving
-     * perception of the input audio.
-     */
-    input_audio_noise_reduction?: TranscriptionSessionCreateParams.InputAudioNoiseReduction;
-    /**
-     * Configuration for input audio transcription. The client can optionally set the
-     * language and prompt for transcription, these offer additional guidance to the
-     * transcription service.
-     */
-    input_audio_transcription?: TranscriptionSessionCreateParams.InputAudioTranscription;
-    /**
-     * The set of modalities the model can respond with. To disable audio, set this to
-     * ["text"].
-     */
-    modalities?: Array<'text' | 'audio'>;
-    /**
-     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
-     * set to `null` to turn off, in which case the client must manually trigger model
-     * response. Server VAD means that the model will detect the start and end of
-     * speech based on audio volume and respond at the end of user speech. Semantic VAD
-     * is more advanced and uses a turn detection model (in conjunction with VAD) to
-     * semantically estimate whether the user has finished speaking, then dynamically
-     * sets a timeout based on this probability. For example, if user audio trails off
-     * with "uhhm", the model will score a low probability of turn end and wait longer
-     * for the user to continue speaking. This can be useful for more natural
-     * conversations, but may have a higher latency.
-     */
-    turn_detection?: TranscriptionSessionCreateParams.TurnDetection;
-}
-export declare namespace TranscriptionSessionCreateParams {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    interface ClientSecret {
-        /**
-         * Configuration for the ephemeral token expiration.
-         */
-        expires_at?: ClientSecret.ExpiresAt;
-    }
-    namespace ClientSecret {
-        /**
-         * Configuration for the ephemeral token expiration.
-         */
-        interface ExpiresAt {
-            /**
-             * The anchor point for the ephemeral token expiration. Only `created_at` is
-             * currently supported.
-             */
-            anchor?: 'created_at';
-            /**
-             * The number of seconds from the anchor point to the expiration. Select a value
-             * between `10` and `7200`.
-             */
-            seconds?: number;
-        }
-    }
-    /**
-     * Configuration for input audio noise reduction. This can be set to `null` to turn
-     * off. Noise reduction filters audio added to the input audio buffer before it is
-     * sent to VAD and the model. Filtering the audio can improve VAD and turn
-     * detection accuracy (reducing false positives) and model performance by improving
-     * perception of the input audio.
-     */
-    interface InputAudioNoiseReduction {
-        /**
-         * Type of noise reduction. `near_field` is for close-talking microphones such as
-         * headphones, `far_field` is for far-field microphones such as laptop or
-         * conference room microphones.
-         */
-        type?: 'near_field' | 'far_field';
-    }
-    /**
-     * Configuration for input audio transcription. The client can optionally set the
-     * language and prompt for transcription, these offer additional guidance to the
-     * transcription service.
-     */
-    interface InputAudioTranscription {
-        /**
-         * The language of the input audio. Supplying the input language in
-         * [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`)
-         * format will improve accuracy and latency.
-         */
-        language?: string;
-        /**
-         * The model to use for transcription, current options are `gpt-4o-transcribe`,
-         * `gpt-4o-mini-transcribe`, and `whisper-1`.
-         */
-        model?: 'gpt-4o-transcribe' | 'gpt-4o-mini-transcribe' | 'whisper-1';
-        /**
-         * An optional text to guide the model's style or continue a previous audio
-         * segment. For `whisper-1`, the
-         * [prompt is a list of keywords](https://platform.openai.com/docs/guides/speech-to-text#prompting).
-         * For `gpt-4o-transcribe` models, the prompt is a free text string, for example
-         * "expect words related to technology".
-         */
-        prompt?: string;
-    }
-    /**
-     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
-     * set to `null` to turn off, in which case the client must manually trigger model
-     * response. Server VAD means that the model will detect the start and end of
-     * speech based on audio volume and respond at the end of user speech. Semantic VAD
-     * is more advanced and uses a turn detection model (in conjunction with VAD) to
-     * semantically estimate whether the user has finished speaking, then dynamically
-     * sets a timeout based on this probability. For example, if user audio trails off
-     * with "uhhm", the model will score a low probability of turn end and wait longer
-     * for the user to continue speaking. This can be useful for more natural
-     * conversations, but may have a higher latency.
-     */
-    interface TurnDetection {
-        /**
-         * Whether or not to automatically generate a response when a VAD stop event
-         * occurs. Not available for transcription sessions.
-         */
-        create_response?: boolean;
-        /**
-         * Used only for `semantic_vad` mode. The eagerness of the model to respond. `low`
-         * will wait longer for the user to continue speaking, `high` will respond more
-         * quickly. `auto` is the default and is equivalent to `medium`.
-         */
-        eagerness?: 'low' | 'medium' | 'high' | 'auto';
-        /**
-         * Whether or not to automatically interrupt any ongoing response with output to
-         * the default conversation (i.e. `conversation` of `auto`) when a VAD start event
-         * occurs. Not available for transcription sessions.
-         */
-        interrupt_response?: boolean;
-        /**
-         * Used only for `server_vad` mode. Amount of audio to include before the VAD
-         * detected speech (in milliseconds). Defaults to 300ms.
-         */
-        prefix_padding_ms?: number;
-        /**
-         * Used only for `server_vad` mode. Duration of silence to detect speech stop (in
-         * milliseconds). Defaults to 500ms. With shorter values the model will respond
-         * more quickly, but may jump in on short pauses from the user.
-         */
-        silence_duration_ms?: number;
-        /**
-         * Used only for `server_vad` mode. Activation threshold for VAD (0.0 to 1.0), this
-         * defaults to 0.5. A higher threshold will require louder audio to activate the
-         * model, and thus might perform better in noisy environments.
-         */
-        threshold?: number;
-        /**
-         * Type of turn detection.
-         */
-        type?: 'server_vad' | 'semantic_vad';
-    }
-}
-export declare namespace TranscriptionSessions {
-    export { type TranscriptionSession as TranscriptionSession, type TranscriptionSessionCreateParams as TranscriptionSessionCreateParams, };
-}
-//# sourceMappingURL=transcription-sessions.d.mts.map

package/extensions/memory-lancedb/node_modules/.ignored_openai/resources/beta/realtime/transcription-sessions.d.mts.map

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