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