AIServer/server/node_modules/openai/resources/beta/realtime/sessions.d.ts

import { APIResource } from "../../../resource.js";
import * as Core from "../../../core.js";
export declare 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: SessionCreateParams, options?: Core.RequestOptions): Core.APIPromise<SessionCreateResponse>;
}
/**
 * Realtime session object configuration.
 */
export interface Session {
    /**
     * Unique identifier for the session that looks like `sess_1234567890abcdef`.
     */
    id?: 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?: Session.InputAudioNoiseReduction;
    /**
     * Configuration for input audio transcription, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through
     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
     * and should be treated as guidance of input audio content rather than precisely
     * what the model heard. The client can optionally set the language and prompt for
     * transcription, these offer additional guidance to the transcription service.
     */
    input_audio_transcription?: Session.InputAudioTranscription;
    /**
     * The default system instructions (i.e. system message) prepended to model calls.
     * This field allows the client to guide the model on desired responses. The model
     * can be instructed on response content and format, (e.g. "be extremely succinct",
     * "act friendly", "here are examples of good responses") and on audio behavior
     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
     * instructions are not guaranteed to be followed by the model, but they provide
     * guidance to the model on the desired behavior.
     *
     * Note that the server sets default instructions which will be used if this field
     * is not set and are visible in the `session.created` event at the start of the
     * session.
     */
    instructions?: string;
    /**
     * Maximum number of output tokens for a single assistant response, inclusive of
     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
     */
    max_response_output_tokens?: number | 'inf';
    /**
     * The set of modalities the model can respond with. To disable audio, set this to
     * ["text"].
     */
    modalities?: Array<'text' | 'audio'>;
    /**
     * The Realtime model used for this session.
     */
    model?: 'gpt-4o-realtime-preview' | 'gpt-4o-realtime-preview-2024-10-01' | 'gpt-4o-realtime-preview-2024-12-17' | 'gpt-4o-mini-realtime-preview' | 'gpt-4o-mini-realtime-preview-2024-12-17';
    /**
     * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
     * For `pcm16`, output audio is sampled at a rate of 24kHz.
     */
    output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
    /**
     * Sampling temperature for the model, limited to [0.6, 1.2]. For audio models a
     * temperature of 0.8 is highly recommended for best performance.
     */
    temperature?: number;
    /**
     * How the model chooses tools. Options are `auto`, `none`, `required`, or specify
     * a function.
     */
    tool_choice?: string;
    /**
     * Tools (functions) available to the model.
     */
    tools?: Array<Session.Tool>;
    /**
     * 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 conjuction 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?: Session.TurnDetection;
    /**
     * The voice the model uses to respond. Voice cannot be changed during the session
     * once the model has responded with audio at least once. Current voice options are
     * `alloy`, `ash`, `ballad`, `coral`, `echo` `sage`, `shimmer` and `verse`.
     */
    voice?: (string & {}) | 'alloy' | 'ash' | 'ballad' | 'coral' | 'echo' | 'fable' | 'onyx' | 'nova' | 'sage' | 'shimmer' | 'verse';
}
export declare namespace Session {
    /**
     * 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, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through
     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
     * and should be treated as guidance of input audio content rather than precisely
     * what the model heard. 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?: string;
        /**
         * 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;
    }
    interface Tool {
        /**
         * The description of the function, including guidance on when and how to call it,
         * and guidance about what to tell the user when calling (if anything).
         */
        description?: string;
        /**
         * The name of the function.
         */
        name?: string;
        /**
         * Parameters of the function in JSON Schema.
         */
        parameters?: unknown;
        /**
         * The type of the tool, i.e. `function`.
         */
        type?: 'function';
    }
    /**
     * 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 conjuction 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.
         */
        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.
         */
        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';
    }
}
/**
 * A new Realtime session configuration, with an ephermeral key. Default TTL for
 * keys is one minute.
 */
export interface SessionCreateResponse {
    /**
     * Ephemeral key returned by the API.
     */
    client_secret: SessionCreateResponse.ClientSecret;
    /**
     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
     */
    input_audio_format?: string;
    /**
     * Configuration for input audio transcription, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through Whisper and should be treated as rough guidance rather
     * than the representation understood by the model.
     */
    input_audio_transcription?: SessionCreateResponse.InputAudioTranscription;
    /**
     * The default system instructions (i.e. system message) prepended to model calls.
     * This field allows the client to guide the model on desired responses. The model
     * can be instructed on response content and format, (e.g. "be extremely succinct",
     * "act friendly", "here are examples of good responses") and on audio behavior
     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
     * instructions are not guaranteed to be followed by the model, but they provide
     * guidance to the model on the desired behavior.
     *
     * Note that the server sets default instructions which will be used if this field
     * is not set and are visible in the `session.created` event at the start of the
     * session.
     */
    instructions?: string;
    /**
     * Maximum number of output tokens for a single assistant response, inclusive of
     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
     */
    max_response_output_tokens?: number | 'inf';
    /**
     * The set of modalities the model can respond with. To disable audio, set this to
     * ["text"].
     */
    modalities?: Array<'text' | 'audio'>;
    /**
     * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
     */
    output_audio_format?: string;
    /**
     * Sampling temperature for the model, limited to [0.6, 1.2]. Defaults to 0.8.
     */
    temperature?: number;
    /**
     * How the model chooses tools. Options are `auto`, `none`, `required`, or specify
     * a function.
     */
    tool_choice?: string;
    /**
     * Tools (functions) available to the model.
     */
    tools?: Array<SessionCreateResponse.Tool>;
    /**
     * 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?: SessionCreateResponse.TurnDetection;
    /**
     * The voice the model uses to respond. Voice cannot be changed during the session
     * once the model has responded with audio at least once. Current voice options are
     * `alloy`, `ash`, `ballad`, `coral`, `echo` `sage`, `shimmer` and `verse`.
     */
    voice?: (string & {}) | 'alloy' | 'ash' | 'ballad' | 'coral' | 'echo' | 'fable' | 'onyx' | 'nova' | 'sage' | 'shimmer' | 'verse';
}
export declare namespace SessionCreateResponse {
    /**
     * Ephemeral key returned by the 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 for input audio transcription, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through Whisper and should be treated as rough guidance rather
     * than the representation understood by the model.
     */
    interface InputAudioTranscription {
        /**
         * The model to use for transcription, `whisper-1` is the only currently supported
         * model.
         */
        model?: string;
    }
    interface Tool {
        /**
         * The description of the function, including guidance on when and how to call it,
         * and guidance about what to tell the user when calling (if anything).
         */
        description?: string;
        /**
         * The name of the function.
         */
        name?: string;
        /**
         * Parameters of the function in JSON Schema.
         */
        parameters?: unknown;
        /**
         * The type of the tool, i.e. `function`.
         */
        type?: 'function';
    }
    /**
     * 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 SessionCreateParams {
    /**
     * Configuration options for the generated client secret.
     */
    client_secret?: SessionCreateParams.ClientSecret;
    /**
     * 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?: SessionCreateParams.InputAudioNoiseReduction;
    /**
     * Configuration for input audio transcription, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through
     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
     * and should be treated as guidance of input audio content rather than precisely
     * what the model heard. The client can optionally set the language and prompt for
     * transcription, these offer additional guidance to the transcription service.
     */
    input_audio_transcription?: SessionCreateParams.InputAudioTranscription;
    /**
     * The default system instructions (i.e. system message) prepended to model calls.
     * This field allows the client to guide the model on desired responses. The model
     * can be instructed on response content and format, (e.g. "be extremely succinct",
     * "act friendly", "here are examples of good responses") and on audio behavior
     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
     * instructions are not guaranteed to be followed by the model, but they provide
     * guidance to the model on the desired behavior.
     *
     * Note that the server sets default instructions which will be used if this field
     * is not set and are visible in the `session.created` event at the start of the
     * session.
     */
    instructions?: string;
    /**
     * Maximum number of output tokens for a single assistant response, inclusive of
     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
     */
    max_response_output_tokens?: number | 'inf';
    /**
     * The set of modalities the model can respond with. To disable audio, set this to
     * ["text"].
     */
    modalities?: Array<'text' | 'audio'>;
    /**
     * The Realtime model used for this session.
     */
    model?: 'gpt-4o-realtime-preview' | 'gpt-4o-realtime-preview-2024-10-01' | 'gpt-4o-realtime-preview-2024-12-17' | 'gpt-4o-mini-realtime-preview' | 'gpt-4o-mini-realtime-preview-2024-12-17';
    /**
     * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
     * For `pcm16`, output audio is sampled at a rate of 24kHz.
     */
    output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
    /**
     * Sampling temperature for the model, limited to [0.6, 1.2]. For audio models a
     * temperature of 0.8 is highly recommended for best performance.
     */
    temperature?: number;
    /**
     * How the model chooses tools. Options are `auto`, `none`, `required`, or specify
     * a function.
     */
    tool_choice?: string;
    /**
     * Tools (functions) available to the model.
     */
    tools?: Array<SessionCreateParams.Tool>;
    /**
     * 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 conjuction 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?: SessionCreateParams.TurnDetection;
    /**
     * The voice the model uses to respond. Voice cannot be changed during the session
     * once the model has responded with audio at least once. Current voice options are
     * `alloy`, `ash`, `ballad`, `coral`, `echo`, `fable`, `onyx`, `nova`, `sage`,
     * `shimmer`, and `verse`.
     */
    voice?: (string & {}) | 'alloy' | 'ash' | 'ballad' | 'coral' | 'echo' | 'fable' | 'onyx' | 'nova' | 'sage' | 'shimmer' | 'verse';
}
export declare namespace SessionCreateParams {
    /**
     * 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, defaults to off and can be set to
     * `null` to turn off once on. Input audio transcription is not native to the
     * model, since the model consumes audio directly. Transcription runs
     * asynchronously through
     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
     * and should be treated as guidance of input audio content rather than precisely
     * what the model heard. 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?: string;
        /**
         * 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;
    }
    interface Tool {
        /**
         * The description of the function, including guidance on when and how to call it,
         * and guidance about what to tell the user when calling (if anything).
         */
        description?: string;
        /**
         * The name of the function.
         */
        name?: string;
        /**
         * Parameters of the function in JSON Schema.
         */
        parameters?: unknown;
        /**
         * The type of the tool, i.e. `function`.
         */
        type?: 'function';
    }
    /**
     * 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 conjuction 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.
         */
        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.
         */
        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 Sessions {
    export { type Session as Session, type SessionCreateResponse as SessionCreateResponse, type SessionCreateParams as SessionCreateParams, };
}
//# sourceMappingURL=sessions.d.ts.map