@nocobase/plugin-ai 2.0.10 → 2.0.11

package/dist/node_modules/openai/src/resources/realtime/client-secrets.ts

@@ -0,0 +1,747 @@
+// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+
+import { APIResource } from '../../core/resource';
+import * as ClientSecretsAPI from './client-secrets';
+import * as RealtimeAPI from './realtime';
+import * as ResponsesAPI from '../responses/responses';
+import { APIPromise } from '../../core/api-promise';
+import { RequestOptions } from '../../internal/request-options';
+
+export class ClientSecrets extends APIResource {
+  /**
+   * Create a Realtime client secret with an associated session configuration.
+   *
+   * @example
+   * ```ts
+   * const clientSecret =
+   *   await client.realtime.clientSecrets.create();
+   * ```
+   */
+  create(body: ClientSecretCreateParams, options?: RequestOptions): APIPromise<ClientSecretCreateResponse> {
+    return this._client.post('/realtime/client_secrets', { body, ...options });
+  }
+}
+
+/**
+ * Ephemeral key returned by the API.
+ */
+export interface RealtimeSessionClientSecret {
+  /**
+   * 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;
+}
+
+/**
+ * A new Realtime session configuration, with an ephemeral key. Default TTL for
+ * keys is one minute.
+ */
+export interface RealtimeSessionCreateResponse {
+  /**
+   * Ephemeral key returned by the API.
+   */
+  client_secret: RealtimeSessionClientSecret;
+
+  /**
+   * The type of session to create. Always `realtime` for the Realtime API.
+   */
+  type: 'realtime';
+
+  /**
+   * Configuration for input and output audio.
+   */
+  audio?: RealtimeSessionCreateResponse.Audio;
+
+  /**
+   * Additional fields to include in server outputs.
+   *
+   * `item.input_audio_transcription.logprobs`: Include logprobs for input audio
+   * transcription.
+   */
+  include?: Array<'item.input_audio_transcription.logprobs'>;
+
+  /**
+   * 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_output_tokens?: number | 'inf';
+
+  /**
+   * The Realtime model used for this session.
+   */
+  model?:
+    | (string & {})
+    | 'gpt-realtime'
+    | 'gpt-realtime-2025-08-28'
+    | 'gpt-4o-realtime-preview'
+    | 'gpt-4o-realtime-preview-2024-10-01'
+    | 'gpt-4o-realtime-preview-2024-12-17'
+    | 'gpt-4o-realtime-preview-2025-06-03'
+    | 'gpt-4o-mini-realtime-preview'
+    | 'gpt-4o-mini-realtime-preview-2024-12-17'
+    | 'gpt-realtime-mini'
+    | 'gpt-realtime-mini-2025-10-06'
+    | 'gpt-realtime-mini-2025-12-15'
+    | 'gpt-audio-mini'
+    | 'gpt-audio-mini-2025-10-06'
+    | 'gpt-audio-mini-2025-12-15';
+
+  /**
+   * The set of modalities the model can respond with. It defaults to `["audio"]`,
+   * indicating that the model will respond with audio plus a transcript. `["text"]`
+   * can be used to make the model respond with text only. It is not possible to
+   * request both `text` and `audio` at the same time.
+   */
+  output_modalities?: Array<'text' | 'audio'>;
+
+  /**
+   * Reference to a prompt template and its variables.
+   * [Learn more](https://platform.openai.com/docs/guides/text?api-mode=responses#reusable-prompts).
+   */
+  prompt?: ResponsesAPI.ResponsePrompt | null;
+
+  /**
+   * How the model chooses tools. Provide one of the string modes or force a specific
+   * function/MCP tool.
+   */
+  tool_choice?: ResponsesAPI.ToolChoiceOptions | ResponsesAPI.ToolChoiceFunction | ResponsesAPI.ToolChoiceMcp;
+
+  /**
+   * Tools available to the model.
+   */
+  tools?: Array<RealtimeAPI.RealtimeFunctionTool | RealtimeSessionCreateResponse.McpTool>;
+
+  /**
+   * Realtime API can write session traces to the
+   * [Traces Dashboard](/logs?api=traces). Set to null to disable tracing. Once
+   * tracing is enabled for a session, the configuration cannot be modified.
+   *
+   * `auto` will create a trace for the session with default values for the workflow
+   * name, group id, and metadata.
+   */
+  tracing?: 'auto' | RealtimeSessionCreateResponse.TracingConfiguration | null;
+
+  /**
+   * When the number of tokens in a conversation exceeds the model's input token
+   * limit, the conversation be truncated, meaning messages (starting from the
+   * oldest) will not be included in the model's context. A 32k context model with
+   * 4,096 max output tokens can only include 28,224 tokens in the context before
+   * truncation occurs.
+   *
+   * Clients can configure truncation behavior to truncate with a lower max token
+   * limit, which is an effective way to control token usage and cost.
+   *
+   * Truncation will reduce the number of cached tokens on the next turn (busting the
+   * cache), since messages are dropped from the beginning of the context. However,
+   * clients can also configure truncation to retain messages up to a fraction of the
+   * maximum context size, which will reduce the need for future truncations and thus
+   * improve the cache rate.
+   *
+   * Truncation can be disabled entirely, which means the server will never truncate
+   * but would instead return an error if the conversation exceeds the model's input
+   * token limit.
+   */
+  truncation?: RealtimeAPI.RealtimeTruncation;
+}
+
+export namespace RealtimeSessionCreateResponse {
+  /**
+   * Configuration for input and output audio.
+   */
+  export interface Audio {
+    input?: Audio.Input;
+
+    output?: Audio.Output;
+  }
+
+  export namespace Audio {
+    export interface Input {
+      /**
+       * The format of the input audio.
+       */
+      format?: RealtimeAPI.RealtimeAudioFormats;
+
+      /**
+       * 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.
+       */
+      noise_reduction?: Input.NoiseReduction;
+
+      /**
+       * 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.
+       */
+      transcription?: RealtimeAPI.AudioTranscription;
+
+      /**
+       * 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?: Input.ServerVad | Input.SemanticVad | null;
+    }
+
+    export namespace Input {
+      /**
+       * 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.
+       */
+      export interface NoiseReduction {
+        /**
+         * 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?: RealtimeAPI.NoiseReductionType;
+      }
+
+      /**
+       * Server-side voice activity detection (VAD) which flips on when user speech is
+       * detected and off after a period of silence.
+       */
+      export interface ServerVad {
+        /**
+         * Type of turn detection, `server_vad` to turn on simple Server VAD.
+         */
+        type: 'server_vad';
+
+        /**
+         * Whether or not to automatically generate a response when a VAD stop event
+         * occurs. If `interrupt_response` is set to `false` this may fail to create a
+         * response if the model is already responding.
+         *
+         * If both `create_response` and `interrupt_response` are set to `false`, the model
+         * will never respond automatically but VAD events will still be emitted.
+         */
+        create_response?: boolean;
+
+        /**
+         * Optional timeout after which a model response will be triggered automatically.
+         * This is useful for situations in which a long pause from the user is unexpected,
+         * such as a phone call. The model will effectively prompt the user to continue the
+         * conversation based on the current context.
+         *
+         * The timeout value will be applied after the last model response's audio has
+         * finished playing, i.e. it's set to the `response.done` time plus audio playback
+         * duration.
+         *
+         * An `input_audio_buffer.timeout_triggered` event (plus events associated with the
+         * Response) will be emitted when the timeout is reached. Idle timeout is currently
+         * only supported for `server_vad` mode.
+         */
+        idle_timeout_ms?: number | null;
+
+        /**
+         * Whether or not to automatically interrupt (cancel) any ongoing response with
+         * output to the default conversation (i.e. `conversation` of `auto`) when a VAD
+         * start event occurs. If `true` then the response will be cancelled, otherwise it
+         * will continue until complete.
+         *
+         * If both `create_response` and `interrupt_response` are set to `false`, the model
+         * will never respond automatically but VAD events will still be emitted.
+         */
+        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;
+      }
+
+      /**
+       * Server-side semantic turn detection which uses a model to determine when the
+       * user has finished speaking.
+       */
+      export interface SemanticVad {
+        /**
+         * Type of turn detection, `semantic_vad` to turn on Semantic VAD.
+         */
+        type: 'semantic_vad';
+
+        /**
+         * 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`. `low`, `medium`,
+         * and `high` have max timeouts of 8s, 4s, and 2s respectively.
+         */
+        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;
+      }
+    }
+
+    export interface Output {
+      /**
+       * The format of the output audio.
+       */
+      format?: RealtimeAPI.RealtimeAudioFormats;
+
+      /**
+       * The speed of the model's spoken response as a multiple of the original speed.
+       * 1.0 is the default speed. 0.25 is the minimum speed. 1.5 is the maximum speed.
+       * This value can only be changed in between model turns, not while a response is
+       * in progress.
+       *
+       * This parameter is a post-processing adjustment to the audio after it is
+       * generated, it's also possible to prompt the model to speak faster or slower.
+       */
+      speed?: number;
+
+      /**
+       * 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`, `verse`, `marin`,
+       * and `cedar`. We recommend `marin` and `cedar` for best quality.
+       */
+      voice?:
+        | (string & {})
+        | 'alloy'
+        | 'ash'
+        | 'ballad'
+        | 'coral'
+        | 'echo'
+        | 'sage'
+        | 'shimmer'
+        | 'verse'
+        | 'marin'
+        | 'cedar';
+    }
+  }
+
+  /**
+   * Give the model access to additional tools via remote Model Context Protocol
+   * (MCP) servers.
+   * [Learn more about MCP](https://platform.openai.com/docs/guides/tools-remote-mcp).
+   */
+  export interface McpTool {
+    /**
+     * A label for this MCP server, used to identify it in tool calls.
+     */
+    server_label: string;
+
+    /**
+     * The type of the MCP tool. Always `mcp`.
+     */
+    type: 'mcp';
+
+    /**
+     * List of allowed tool names or a filter object.
+     */
+    allowed_tools?: Array<string> | McpTool.McpToolFilter | null;
+
+    /**
+     * An OAuth access token that can be used with a remote MCP server, either with a
+     * custom MCP server URL or a service connector. Your application must handle the
+     * OAuth authorization flow and provide the token here.
+     */
+    authorization?: string;
+
+    /**
+     * Identifier for service connectors, like those available in ChatGPT. One of
+     * `server_url` or `connector_id` must be provided. Learn more about service
+     * connectors
+     * [here](https://platform.openai.com/docs/guides/tools-remote-mcp#connectors).
+     *
+     * Currently supported `connector_id` values are:
+     *
+     * - Dropbox: `connector_dropbox`
+     * - Gmail: `connector_gmail`
+     * - Google Calendar: `connector_googlecalendar`
+     * - Google Drive: `connector_googledrive`
+     * - Microsoft Teams: `connector_microsoftteams`
+     * - Outlook Calendar: `connector_outlookcalendar`
+     * - Outlook Email: `connector_outlookemail`
+     * - SharePoint: `connector_sharepoint`
+     */
+    connector_id?:
+      | 'connector_dropbox'
+      | 'connector_gmail'
+      | 'connector_googlecalendar'
+      | 'connector_googledrive'
+      | 'connector_microsoftteams'
+      | 'connector_outlookcalendar'
+      | 'connector_outlookemail'
+      | 'connector_sharepoint';
+
+    /**
+     * Optional HTTP headers to send to the MCP server. Use for authentication or other
+     * purposes.
+     */
+    headers?: { [key: string]: string } | null;
+
+    /**
+     * Specify which of the MCP server's tools require approval.
+     */
+    require_approval?: McpTool.McpToolApprovalFilter | 'always' | 'never' | null;
+
+    /**
+     * Optional description of the MCP server, used to provide more context.
+     */
+    server_description?: string;
+
+    /**
+     * The URL for the MCP server. One of `server_url` or `connector_id` must be
+     * provided.
+     */
+    server_url?: string;
+  }
+
+  export namespace McpTool {
+    /**
+     * A filter object to specify which tools are allowed.
+     */
+    export interface McpToolFilter {
+      /**
+       * Indicates whether or not a tool modifies data or is read-only. If an MCP server
+       * is
+       * [annotated with `readOnlyHint`](https://modelcontextprotocol.io/specification/2025-06-18/schema#toolannotations-readonlyhint),
+       * it will match this filter.
+       */
+      read_only?: boolean;
+
+      /**
+       * List of allowed tool names.
+       */
+      tool_names?: Array<string>;
+    }
+
+    /**
+     * Specify which of the MCP server's tools require approval. Can be `always`,
+     * `never`, or a filter object associated with tools that require approval.
+     */
+    export interface McpToolApprovalFilter {
+      /**
+       * A filter object to specify which tools are allowed.
+       */
+      always?: McpToolApprovalFilter.Always;
+
+      /**
+       * A filter object to specify which tools are allowed.
+       */
+      never?: McpToolApprovalFilter.Never;
+    }
+
+    export namespace McpToolApprovalFilter {
+      /**
+       * A filter object to specify which tools are allowed.
+       */
+      export interface Always {
+        /**
+         * Indicates whether or not a tool modifies data or is read-only. If an MCP server
+         * is
+         * [annotated with `readOnlyHint`](https://modelcontextprotocol.io/specification/2025-06-18/schema#toolannotations-readonlyhint),
+         * it will match this filter.
+         */
+        read_only?: boolean;
+
+        /**
+         * List of allowed tool names.
+         */
+        tool_names?: Array<string>;
+      }
+
+      /**
+       * A filter object to specify which tools are allowed.
+       */
+      export interface Never {
+        /**
+         * Indicates whether or not a tool modifies data or is read-only. If an MCP server
+         * is
+         * [annotated with `readOnlyHint`](https://modelcontextprotocol.io/specification/2025-06-18/schema#toolannotations-readonlyhint),
+         * it will match this filter.
+         */
+        read_only?: boolean;
+
+        /**
+         * List of allowed tool names.
+         */
+        tool_names?: Array<string>;
+      }
+    }
+  }
+
+  /**
+   * Granular configuration for tracing.
+   */
+  export interface TracingConfiguration {
+    /**
+     * The group id to attach to this trace to enable filtering and grouping in the
+     * Traces Dashboard.
+     */
+    group_id?: string;
+
+    /**
+     * The arbitrary metadata to attach to this trace to enable filtering in the Traces
+     * Dashboard.
+     */
+    metadata?: unknown;
+
+    /**
+     * The name of the workflow to attach to this trace. This is used to name the trace
+     * in the Traces Dashboard.
+     */
+    workflow_name?: string;
+  }
+}
+
+/**
+ * A Realtime transcription session configuration object.
+ */
+export interface RealtimeTranscriptionSessionCreateResponse {
+  /**
+   * Unique identifier for the session that looks like `sess_1234567890abcdef`.
+   */
+  id: string;
+
+  /**
+   * The object type. Always `realtime.transcription_session`.
+   */
+  object: string;
+
+  /**
+   * The type of session. Always `transcription` for transcription sessions.
+   */
+  type: 'transcription';
+
+  /**
+   * Configuration for input audio for the session.
+   */
+  audio?: RealtimeTranscriptionSessionCreateResponse.Audio;
+
+  /**
+   * Expiration timestamp for the session, in seconds since epoch.
+   */
+  expires_at?: number;
+
+  /**
+   * Additional fields to include in server outputs.
+   *
+   * - `item.input_audio_transcription.logprobs`: Include logprobs for input audio
+   *   transcription.
+   */
+  include?: Array<'item.input_audio_transcription.logprobs'>;
+}
+
+export namespace RealtimeTranscriptionSessionCreateResponse {
+  /**
+   * Configuration for input audio for the session.
+   */
+  export interface Audio {
+    input?: Audio.Input;
+  }
+
+  export namespace Audio {
+    export interface Input {
+      /**
+       * The PCM audio format. Only a 24kHz sample rate is supported.
+       */
+      format?: RealtimeAPI.RealtimeAudioFormats;
+
+      /**
+       * Configuration for input audio noise reduction.
+       */
+      noise_reduction?: Input.NoiseReduction;
+
+      /**
+       * Configuration of the transcription model.
+       */
+      transcription?: RealtimeAPI.AudioTranscription;
+
+      /**
+       * 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?: ClientSecretsAPI.RealtimeTranscriptionSessionTurnDetection;
+    }
+
+    export namespace Input {
+      /**
+       * Configuration for input audio noise reduction.
+       */
+      export interface NoiseReduction {
+        /**
+         * 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?: RealtimeAPI.NoiseReductionType;
+      }
+    }
+  }
+}
+
+/**
+ * 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.
+ */
+export interface RealtimeTranscriptionSessionTurnDetection {
+  /**
+   * 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;
+}
+
+/**
+ * Response from creating a session and client secret for the Realtime API.
+ */
+export interface ClientSecretCreateResponse {
+  /**
+   * Expiration timestamp for the client secret, in seconds since epoch.
+   */
+  expires_at: number;
+
+  /**
+   * The session configuration for either a realtime or transcription session.
+   */
+  session: RealtimeSessionCreateResponse | RealtimeTranscriptionSessionCreateResponse;
+
+  /**
+   * The generated client secret value.
+   */
+  value: string;
+}
+
+export interface ClientSecretCreateParams {
+  /**
+   * Configuration for the client secret expiration. Expiration refers to the time
+   * after which a client secret will no longer be valid for creating sessions. The
+   * session itself may continue after that time once started. A secret can be used
+   * to create multiple sessions until it expires.
+   */
+  expires_after?: ClientSecretCreateParams.ExpiresAfter;
+
+  /**
+   * Session configuration to use for the client secret. Choose either a realtime
+   * session or a transcription session.
+   */
+  session?: RealtimeAPI.RealtimeSessionCreateRequest | RealtimeAPI.RealtimeTranscriptionSessionCreateRequest;
+}
+
+export namespace ClientSecretCreateParams {
+  /**
+   * Configuration for the client secret expiration. Expiration refers to the time
+   * after which a client secret will no longer be valid for creating sessions. The
+   * session itself may continue after that time once started. A secret can be used
+   * to create multiple sessions until it expires.
+   */
+  export interface ExpiresAfter {
+    /**
+     * The anchor point for the client secret expiration, meaning that `seconds` will
+     * be added to the `created_at` time of the client secret to produce an expiration
+     * timestamp. 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` (2 hours). This default to 600 seconds (10 minutes) if
+     * not specified.
+     */
+    seconds?: number;
+  }
+}
+
+export declare namespace ClientSecrets {
+  export {
+    type RealtimeSessionClientSecret as RealtimeSessionClientSecret,
+    type RealtimeSessionCreateResponse as RealtimeSessionCreateResponse,
+    type RealtimeTranscriptionSessionCreateResponse as RealtimeTranscriptionSessionCreateResponse,
+    type RealtimeTranscriptionSessionTurnDetection as RealtimeTranscriptionSessionTurnDetection,
+    type ClientSecretCreateResponse as ClientSecretCreateResponse,
+    type ClientSecretCreateParams as ClientSecretCreateParams,
+  };
+}