@jsonstudio/rcc 0.89.611 → 0.89.682

package/node_modules/@jsonstudio/llms/node_modules/openai/src/resources/beta/realtime/realtime.ts

@@ -1,2829 +0,0 @@
-// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
-
-import { APIResource } from '../../../core/resource';
-import * as RealtimeAPI from './realtime';
-import * as Shared from '../../shared';
-import * as SessionsAPI from './sessions';
-import {
-  Session as SessionsAPISession,
-  SessionCreateParams,
-  SessionCreateResponse,
-  Sessions,
-} from './sessions';
-import * as TranscriptionSessionsAPI from './transcription-sessions';
-import {
-  TranscriptionSession,
-  TranscriptionSessionCreateParams,
-  TranscriptionSessions,
-} from './transcription-sessions';
-
-/**
- * @deprecated Realtime has now launched and is generally available. The old beta API is now deprecated.
- */
-export class Realtime extends APIResource {
-  sessions: SessionsAPI.Sessions = new SessionsAPI.Sessions(this._client);
-  transcriptionSessions: TranscriptionSessionsAPI.TranscriptionSessions =
-    new TranscriptionSessionsAPI.TranscriptionSessions(this._client);
-}
-
-/**
- * Returned when a conversation is created. Emitted right after session creation.
- */
-export interface ConversationCreatedEvent {
-  /**
-   * The conversation resource.
-   */
-  conversation: ConversationCreatedEvent.Conversation;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The event type, must be `conversation.created`.
-   */
-  type: 'conversation.created';
-}
-
-export namespace ConversationCreatedEvent {
-  /**
-   * The conversation resource.
-   */
-  export interface Conversation {
-    /**
-     * The unique ID of the conversation.
-     */
-    id?: string;
-
-    /**
-     * The object type, must be `realtime.conversation`.
-     */
-    object?: 'realtime.conversation';
-  }
-}
-
-/**
- * The item to add to the conversation.
- */
-export interface ConversationItem {
-  /**
-   * The unique ID of the item, this can be generated by the client to help manage
-   * server-side context, but is not required because the server will generate one if
-   * not provided.
-   */
-  id?: string;
-
-  /**
-   * The arguments of the function call (for `function_call` items).
-   */
-  arguments?: string;
-
-  /**
-   * The ID of the function call (for `function_call` and `function_call_output`
-   * items). If passed on a `function_call_output` item, the server will check that a
-   * `function_call` item with the same ID exists in the conversation history.
-   */
-  call_id?: string;
-
-  /**
-   * The content of the message, applicable for `message` items.
-   *
-   * - Message items of role `system` support only `input_text` content
-   * - Message items of role `user` support `input_text` and `input_audio` content
-   * - Message items of role `assistant` support `text` content.
-   */
-  content?: Array<ConversationItemContent>;
-
-  /**
-   * The name of the function being called (for `function_call` items).
-   */
-  name?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The output of the function call (for `function_call_output` items).
-   */
-  output?: string;
-
-  /**
-   * The role of the message sender (`user`, `assistant`, `system`), only applicable
-   * for `message` items.
-   */
-  role?: 'user' | 'assistant' | 'system';
-
-  /**
-   * The status of the item (`completed`, `incomplete`, `in_progress`). These have no
-   * effect on the conversation, but are accepted for consistency with the
-   * `conversation.item.created` event.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-
-  /**
-   * The type of the item (`message`, `function_call`, `function_call_output`).
-   */
-  type?: 'message' | 'function_call' | 'function_call_output';
-}
-
-export interface ConversationItemContent {
-  /**
-   * ID of a previous conversation item to reference (for `item_reference` content
-   * types in `response.create` events). These can reference both client and server
-   * created items.
-   */
-  id?: string;
-
-  /**
-   * Base64-encoded audio bytes, used for `input_audio` content type.
-   */
-  audio?: string;
-
-  /**
-   * The text content, used for `input_text` and `text` content types.
-   */
-  text?: string;
-
-  /**
-   * The transcript of the audio, used for `input_audio` and `audio` content types.
-   */
-  transcript?: string;
-
-  /**
-   * The content type (`input_text`, `input_audio`, `item_reference`, `text`,
-   * `audio`).
-   */
-  type?: 'input_text' | 'input_audio' | 'item_reference' | 'text' | 'audio';
-}
-
-/**
- * Add a new Item to the Conversation's context, including messages, function
- * calls, and function call responses. This event can be used both to populate a
- * "history" of the conversation and to add new items mid-stream, but has the
- * current limitation that it cannot populate assistant audio messages.
- *
- * If successful, the server will respond with a `conversation.item.created` event,
- * otherwise an `error` event will be sent.
- */
-export interface ConversationItemCreateEvent {
-  /**
-   * The item to add to the conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The event type, must be `conversation.item.create`.
-   */
-  type: 'conversation.item.create';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-
-  /**
-   * The ID of the preceding item after which the new item will be inserted. If not
-   * set, the new item will be appended to the end of the conversation. If set to
-   * `root`, the new item will be added to the beginning of the conversation. If set
-   * to an existing ID, it allows an item to be inserted mid-conversation. If the ID
-   * cannot be found, an error will be returned and the item will not be added.
-   */
-  previous_item_id?: string;
-}
-
-/**
- * Returned when a conversation item is created. There are several scenarios that
- * produce this event:
- *
- * - The server is generating a Response, which if successful will produce either
- *   one or two Items, which will be of type `message` (role `assistant`) or type
- *   `function_call`.
- * - The input audio buffer has been committed, either by the client or the server
- *   (in `server_vad` mode). The server will take the content of the input audio
- *   buffer and add it to a new user message Item.
- * - The client has sent a `conversation.item.create` event to add a new Item to
- *   the Conversation.
- */
-export interface ConversationItemCreatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The item to add to the conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The event type, must be `conversation.item.created`.
-   */
-  type: 'conversation.item.created';
-
-  /**
-   * The ID of the preceding item in the Conversation context, allows the client to
-   * understand the order of the conversation. Can be `null` if the item has no
-   * predecessor.
-   */
-  previous_item_id?: string | null;
-}
-
-/**
- * Send this event when you want to remove any item from the conversation history.
- * The server will respond with a `conversation.item.deleted` event, unless the
- * item does not exist in the conversation history, in which case the server will
- * respond with an error.
- */
-export interface ConversationItemDeleteEvent {
-  /**
-   * The ID of the item to delete.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.delete`.
-   */
-  type: 'conversation.item.delete';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Returned when an item in the conversation is deleted by the client with a
- * `conversation.item.delete` event. This event is used to synchronize the server's
- * understanding of the conversation history with the client's view.
- */
-export interface ConversationItemDeletedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item that was deleted.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.deleted`.
-   */
-  type: 'conversation.item.deleted';
-}
-
-/**
- * This event is the output of audio transcription for user audio written to the
- * user audio buffer. Transcription begins when the input audio buffer is committed
- * by the client or server (in `server_vad` mode). Transcription runs
- * asynchronously with Response creation, so this event may come before or after
- * the Response events.
- *
- * Realtime API models accept audio natively, and thus input transcription is a
- * separate process run on a separate ASR (Automatic Speech Recognition) model. The
- * transcript may diverge somewhat from the model's interpretation, and should be
- * treated as a rough guide.
- */
-export interface ConversationItemInputAudioTranscriptionCompletedEvent {
-  /**
-   * The index of the content part containing the audio.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the user message item containing the audio.
-   */
-  item_id: string;
-
-  /**
-   * The transcribed text.
-   */
-  transcript: string;
-
-  /**
-   * The event type, must be `conversation.item.input_audio_transcription.completed`.
-   */
-  type: 'conversation.item.input_audio_transcription.completed';
-
-  /**
-   * Usage statistics for the transcription.
-   */
-  usage:
-    | ConversationItemInputAudioTranscriptionCompletedEvent.TranscriptTextUsageTokens
-    | ConversationItemInputAudioTranscriptionCompletedEvent.TranscriptTextUsageDuration;
-
-  /**
-   * The log probabilities of the transcription.
-   */
-  logprobs?: Array<ConversationItemInputAudioTranscriptionCompletedEvent.Logprob> | null;
-}
-
-export namespace ConversationItemInputAudioTranscriptionCompletedEvent {
-  /**
-   * Usage statistics for models billed by token usage.
-   */
-  export interface TranscriptTextUsageTokens {
-    /**
-     * Number of input tokens billed for this request.
-     */
-    input_tokens: number;
-
-    /**
-     * Number of output tokens generated.
-     */
-    output_tokens: number;
-
-    /**
-     * Total number of tokens used (input + output).
-     */
-    total_tokens: number;
-
-    /**
-     * The type of the usage object. Always `tokens` for this variant.
-     */
-    type: 'tokens';
-
-    /**
-     * Details about the input tokens billed for this request.
-     */
-    input_token_details?: TranscriptTextUsageTokens.InputTokenDetails;
-  }
-
-  export namespace TranscriptTextUsageTokens {
-    /**
-     * Details about the input tokens billed for this request.
-     */
-    export interface InputTokenDetails {
-      /**
-       * Number of audio tokens billed for this request.
-       */
-      audio_tokens?: number;
-
-      /**
-       * Number of text tokens billed for this request.
-       */
-      text_tokens?: number;
-    }
-  }
-
-  /**
-   * Usage statistics for models billed by audio input duration.
-   */
-  export interface TranscriptTextUsageDuration {
-    /**
-     * Duration of the input audio in seconds.
-     */
-    seconds: number;
-
-    /**
-     * The type of the usage object. Always `duration` for this variant.
-     */
-    type: 'duration';
-  }
-
-  /**
-   * A log probability object.
-   */
-  export interface Logprob {
-    /**
-     * The token that was used to generate the log probability.
-     */
-    token: string;
-
-    /**
-     * The bytes that were used to generate the log probability.
-     */
-    bytes: Array<number>;
-
-    /**
-     * The log probability of the token.
-     */
-    logprob: number;
-  }
-}
-
-/**
- * Returned when the text value of an input audio transcription content part is
- * updated.
- */
-export interface ConversationItemInputAudioTranscriptionDeltaEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.input_audio_transcription.delta`.
-   */
-  type: 'conversation.item.input_audio_transcription.delta';
-
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index?: number;
-
-  /**
-   * The text delta.
-   */
-  delta?: string;
-
-  /**
-   * The log probabilities of the transcription.
-   */
-  logprobs?: Array<ConversationItemInputAudioTranscriptionDeltaEvent.Logprob> | null;
-}
-
-export namespace ConversationItemInputAudioTranscriptionDeltaEvent {
-  /**
-   * A log probability object.
-   */
-  export interface Logprob {
-    /**
-     * The token that was used to generate the log probability.
-     */
-    token: string;
-
-    /**
-     * The bytes that were used to generate the log probability.
-     */
-    bytes: Array<number>;
-
-    /**
-     * The log probability of the token.
-     */
-    logprob: number;
-  }
-}
-
-/**
- * Returned when input audio transcription is configured, and a transcription
- * request for a user message failed. These events are separate from other `error`
- * events so that the client can identify the related Item.
- */
-export interface ConversationItemInputAudioTranscriptionFailedEvent {
-  /**
-   * The index of the content part containing the audio.
-   */
-  content_index: number;
-
-  /**
-   * Details of the transcription error.
-   */
-  error: ConversationItemInputAudioTranscriptionFailedEvent.Error;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the user message item.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.input_audio_transcription.failed`.
-   */
-  type: 'conversation.item.input_audio_transcription.failed';
-}
-
-export namespace ConversationItemInputAudioTranscriptionFailedEvent {
-  /**
-   * Details of the transcription error.
-   */
-  export interface Error {
-    /**
-     * Error code, if any.
-     */
-    code?: string;
-
-    /**
-     * A human-readable error message.
-     */
-    message?: string;
-
-    /**
-     * Parameter related to the error, if any.
-     */
-    param?: string;
-
-    /**
-     * The type of error.
-     */
-    type?: string;
-  }
-}
-
-/**
- * Send this event when you want to retrieve the server's representation of a
- * specific item in the conversation history. This is useful, for example, to
- * inspect user audio after noise cancellation and VAD. The server will respond
- * with a `conversation.item.retrieved` event, unless the item does not exist in
- * the conversation history, in which case the server will respond with an error.
- */
-export interface ConversationItemRetrieveEvent {
-  /**
-   * The ID of the item to retrieve.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.retrieve`.
-   */
-  type: 'conversation.item.retrieve';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Send this event to truncate a previous assistant message’s audio. The server
- * will produce audio faster than realtime, so this event is useful when the user
- * interrupts to truncate audio that has already been sent to the client but not
- * yet played. This will synchronize the server's understanding of the audio with
- * the client's playback.
- *
- * Truncating audio will delete the server-side text transcript to ensure there is
- * not text in the context that hasn't been heard by the user.
- *
- * If successful, the server will respond with a `conversation.item.truncated`
- * event.
- */
-export interface ConversationItemTruncateEvent {
-  /**
-   * Inclusive duration up to which audio is truncated, in milliseconds. If the
-   * audio_end_ms is greater than the actual audio duration, the server will respond
-   * with an error.
-   */
-  audio_end_ms: number;
-
-  /**
-   * The index of the content part to truncate. Set this to 0.
-   */
-  content_index: number;
-
-  /**
-   * The ID of the assistant message item to truncate. Only assistant message items
-   * can be truncated.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.truncate`.
-   */
-  type: 'conversation.item.truncate';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Returned when an earlier assistant audio message item is truncated by the client
- * with a `conversation.item.truncate` event. This event is used to synchronize the
- * server's understanding of the audio with the client's playback.
- *
- * This action will truncate the audio and remove the server-side text transcript
- * to ensure there is no text in the context that hasn't been heard by the user.
- */
-export interface ConversationItemTruncatedEvent {
-  /**
-   * The duration up to which the audio was truncated, in milliseconds.
-   */
-  audio_end_ms: number;
-
-  /**
-   * The index of the content part that was truncated.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the assistant message item that was truncated.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `conversation.item.truncated`.
-   */
-  type: 'conversation.item.truncated';
-}
-
-/**
- * The item to add to the conversation.
- */
-export interface ConversationItemWithReference {
-  /**
-   * For an item of type (`message` | `function_call` | `function_call_output`) this
-   * field allows the client to assign the unique ID of the item. It is not required
-   * because the server will generate one if not provided.
-   *
-   * For an item of type `item_reference`, this field is required and is a reference
-   * to any item that has previously existed in the conversation.
-   */
-  id?: string;
-
-  /**
-   * The arguments of the function call (for `function_call` items).
-   */
-  arguments?: string;
-
-  /**
-   * The ID of the function call (for `function_call` and `function_call_output`
-   * items). If passed on a `function_call_output` item, the server will check that a
-   * `function_call` item with the same ID exists in the conversation history.
-   */
-  call_id?: string;
-
-  /**
-   * The content of the message, applicable for `message` items.
-   *
-   * - Message items of role `system` support only `input_text` content
-   * - Message items of role `user` support `input_text` and `input_audio` content
-   * - Message items of role `assistant` support `text` content.
-   */
-  content?: Array<ConversationItemWithReference.Content>;
-
-  /**
-   * The name of the function being called (for `function_call` items).
-   */
-  name?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The output of the function call (for `function_call_output` items).
-   */
-  output?: string;
-
-  /**
-   * The role of the message sender (`user`, `assistant`, `system`), only applicable
-   * for `message` items.
-   */
-  role?: 'user' | 'assistant' | 'system';
-
-  /**
-   * The status of the item (`completed`, `incomplete`, `in_progress`). These have no
-   * effect on the conversation, but are accepted for consistency with the
-   * `conversation.item.created` event.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-
-  /**
-   * The type of the item (`message`, `function_call`, `function_call_output`,
-   * `item_reference`).
-   */
-  type?: 'message' | 'function_call' | 'function_call_output' | 'item_reference';
-}
-
-export namespace ConversationItemWithReference {
-  export interface Content {
-    /**
-     * ID of a previous conversation item to reference (for `item_reference` content
-     * types in `response.create` events). These can reference both client and server
-     * created items.
-     */
-    id?: string;
-
-    /**
-     * Base64-encoded audio bytes, used for `input_audio` content type.
-     */
-    audio?: string;
-
-    /**
-     * The text content, used for `input_text` and `text` content types.
-     */
-    text?: string;
-
-    /**
-     * The transcript of the audio, used for `input_audio` content type.
-     */
-    transcript?: string;
-
-    /**
-     * The content type (`input_text`, `input_audio`, `item_reference`, `text`).
-     */
-    type?: 'input_text' | 'input_audio' | 'item_reference' | 'text';
-  }
-}
-
-/**
- * Returned when an error occurs, which could be a client problem or a server
- * problem. Most errors are recoverable and the session will stay open, we
- * recommend to implementors to monitor and log error messages by default.
- */
-export interface ErrorEvent {
-  /**
-   * Details of the error.
-   */
-  error: ErrorEvent.Error;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The event type, must be `error`.
-   */
-  type: 'error';
-}
-
-export namespace ErrorEvent {
-  /**
-   * Details of the error.
-   */
-  export interface Error {
-    /**
-     * A human-readable error message.
-     */
-    message: string;
-
-    /**
-     * The type of error (e.g., "invalid_request_error", "server_error").
-     */
-    type: string;
-
-    /**
-     * Error code, if any.
-     */
-    code?: string | null;
-
-    /**
-     * The event_id of the client event that caused the error, if applicable.
-     */
-    event_id?: string | null;
-
-    /**
-     * Parameter related to the error, if any.
-     */
-    param?: string | null;
-  }
-}
-
-/**
- * Send this event to append audio bytes to the input audio buffer. The audio
- * buffer is temporary storage you can write to and later commit. In Server VAD
- * mode, the audio buffer is used to detect speech and the server will decide when
- * to commit. When Server VAD is disabled, you must commit the audio buffer
- * manually.
- *
- * The client may choose how much audio to place in each event up to a maximum of
- * 15 MiB, for example streaming smaller chunks from the client may allow the VAD
- * to be more responsive. Unlike made other client events, the server will not send
- * a confirmation response to this event.
- */
-export interface InputAudioBufferAppendEvent {
-  /**
-   * Base64-encoded audio bytes. This must be in the format specified by the
-   * `input_audio_format` field in the session configuration.
-   */
-  audio: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.append`.
-   */
-  type: 'input_audio_buffer.append';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Send this event to clear the audio bytes in the buffer. The server will respond
- * with an `input_audio_buffer.cleared` event.
- */
-export interface InputAudioBufferClearEvent {
-  /**
-   * The event type, must be `input_audio_buffer.clear`.
-   */
-  type: 'input_audio_buffer.clear';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Returned when the input audio buffer is cleared by the client with a
- * `input_audio_buffer.clear` event.
- */
-export interface InputAudioBufferClearedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.cleared`.
-   */
-  type: 'input_audio_buffer.cleared';
-}
-
-/**
- * Send this event to commit the user input audio buffer, which will create a new
- * user message item in the conversation. This event will produce an error if the
- * input audio buffer is empty. When in Server VAD mode, the client does not need
- * to send this event, the server will commit the audio buffer automatically.
- *
- * Committing the input audio buffer will trigger input audio transcription (if
- * enabled in session configuration), but it will not create a response from the
- * model. The server will respond with an `input_audio_buffer.committed` event.
- */
-export interface InputAudioBufferCommitEvent {
-  /**
-   * The event type, must be `input_audio_buffer.commit`.
-   */
-  type: 'input_audio_buffer.commit';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-/**
- * Returned when an input audio buffer is committed, either by the client or
- * automatically in server VAD mode. The `item_id` property is the ID of the user
- * message item that will be created, thus a `conversation.item.created` event will
- * also be sent to the client.
- */
-export interface InputAudioBufferCommittedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the user message item that will be created.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.committed`.
-   */
-  type: 'input_audio_buffer.committed';
-
-  /**
-   * The ID of the preceding item after which the new item will be inserted. Can be
-   * `null` if the item has no predecessor.
-   */
-  previous_item_id?: string | null;
-}
-
-/**
- * Sent by the server when in `server_vad` mode to indicate that speech has been
- * detected in the audio buffer. This can happen any time audio is added to the
- * buffer (unless speech is already detected). The client may want to use this
- * event to interrupt audio playback or provide visual feedback to the user.
- *
- * The client should expect to receive a `input_audio_buffer.speech_stopped` event
- * when speech stops. The `item_id` property is the ID of the user message item
- * that will be created when speech stops and will also be included in the
- * `input_audio_buffer.speech_stopped` event (unless the client manually commits
- * the audio buffer during VAD activation).
- */
-export interface InputAudioBufferSpeechStartedEvent {
-  /**
-   * Milliseconds from the start of all audio written to the buffer during the
-   * session when speech was first detected. This will correspond to the beginning of
-   * audio sent to the model, and thus includes the `prefix_padding_ms` configured in
-   * the Session.
-   */
-  audio_start_ms: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the user message item that will be created when speech stops.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.speech_started`.
-   */
-  type: 'input_audio_buffer.speech_started';
-}
-
-/**
- * Returned in `server_vad` mode when the server detects the end of speech in the
- * audio buffer. The server will also send an `conversation.item.created` event
- * with the user message item that is created from the audio buffer.
- */
-export interface InputAudioBufferSpeechStoppedEvent {
-  /**
-   * Milliseconds since the session started when speech stopped. This will correspond
-   * to the end of audio sent to the model, and thus includes the
-   * `min_silence_duration_ms` configured in the Session.
-   */
-  audio_end_ms: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the user message item that will be created.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.speech_stopped`.
-   */
-  type: 'input_audio_buffer.speech_stopped';
-}
-
-/**
- * Emitted at the beginning of a Response to indicate the updated rate limits. When
- * a Response is created some tokens will be "reserved" for the output tokens, the
- * rate limits shown here reflect that reservation, which is then adjusted
- * accordingly once the Response is completed.
- */
-export interface RateLimitsUpdatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * List of rate limit information.
-   */
-  rate_limits: Array<RateLimitsUpdatedEvent.RateLimit>;
-
-  /**
-   * The event type, must be `rate_limits.updated`.
-   */
-  type: 'rate_limits.updated';
-}
-
-export namespace RateLimitsUpdatedEvent {
-  export interface RateLimit {
-    /**
-     * The maximum allowed value for the rate limit.
-     */
-    limit?: number;
-
-    /**
-     * The name of the rate limit (`requests`, `tokens`).
-     */
-    name?: 'requests' | 'tokens';
-
-    /**
-     * The remaining value before the limit is reached.
-     */
-    remaining?: number;
-
-    /**
-     * Seconds until the rate limit resets.
-     */
-    reset_seconds?: number;
-  }
-}
-
-/**
- * A realtime client event.
- */
-export type RealtimeClientEvent =
-  | ConversationItemCreateEvent
-  | ConversationItemDeleteEvent
-  | ConversationItemRetrieveEvent
-  | ConversationItemTruncateEvent
-  | InputAudioBufferAppendEvent
-  | InputAudioBufferClearEvent
-  | RealtimeClientEvent.OutputAudioBufferClear
-  | InputAudioBufferCommitEvent
-  | ResponseCancelEvent
-  | ResponseCreateEvent
-  | SessionUpdateEvent
-  | TranscriptionSessionUpdate;
-
-export namespace RealtimeClientEvent {
-  /**
-   * **WebRTC Only:** Emit to cut off the current audio response. This will trigger
-   * the server to stop generating audio and emit a `output_audio_buffer.cleared`
-   * event. This event should be preceded by a `response.cancel` client event to stop
-   * the generation of the current response.
-   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
-   */
-  export interface OutputAudioBufferClear {
-    /**
-     * The event type, must be `output_audio_buffer.clear`.
-     */
-    type: 'output_audio_buffer.clear';
-
-    /**
-     * The unique ID of the client event used for error handling.
-     */
-    event_id?: string;
-  }
-}
-
-/**
- * The response resource.
- */
-export interface RealtimeResponse {
-  /**
-   * The unique ID of the response.
-   */
-  id?: string;
-
-  /**
-   * Which conversation the response is added to, determined by the `conversation`
-   * field in the `response.create` event. If `auto`, the response will be added to
-   * the default conversation and the value of `conversation_id` will be an id like
-   * `conv_1234`. If `none`, the response will not be added to any conversation and
-   * the value of `conversation_id` will be `null`. If responses are being triggered
-   * by server VAD, the response will be added to the default conversation, thus the
-   * `conversation_id` will be an id like `conv_1234`.
-   */
-  conversation_id?: string;
-
-  /**
-   * Maximum number of output tokens for a single assistant response, inclusive of
-   * tool calls, that was used in this response.
-   */
-  max_output_tokens?: number | 'inf';
-
-  /**
-   * Set of 16 key-value pairs that can be attached to an object. This can be useful
-   * for storing additional information about the object in a structured format, and
-   * querying for objects via API or the dashboard.
-   *
-   * Keys are strings with a maximum length of 64 characters. Values are strings with
-   * a maximum length of 512 characters.
-   */
-  metadata?: Shared.Metadata | null;
-
-  /**
-   * The set of modalities the model used to respond. If there are multiple
-   * modalities, the model will pick one, for example if `modalities` is
-   * `["text", "audio"]`, the model could be responding in either text or audio.
-   */
-  modalities?: Array<'text' | 'audio'>;
-
-  /**
-   * The object type, must be `realtime.response`.
-   */
-  object?: 'realtime.response';
-
-  /**
-   * The list of output items generated by the response.
-   */
-  output?: Array<ConversationItem>;
-
-  /**
-   * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
-   */
-  output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
-
-  /**
-   * The final status of the response (`completed`, `cancelled`, `failed`, or
-   * `incomplete`, `in_progress`).
-   */
-  status?: 'completed' | 'cancelled' | 'failed' | 'incomplete' | 'in_progress';
-
-  /**
-   * Additional details about the status.
-   */
-  status_details?: RealtimeResponseStatus;
-
-  /**
-   * Sampling temperature for the model, limited to [0.6, 1.2]. Defaults to 0.8.
-   */
-  temperature?: number;
-
-  /**
-   * Usage statistics for the Response, this will correspond to billing. A Realtime
-   * API session will maintain a conversation context and append new Items to the
-   * Conversation, thus output from previous turns (text and audio tokens) will
-   * become the input for later turns.
-   */
-  usage?: RealtimeResponseUsage;
-
-  /**
-   * The voice the model used to respond. Current voice options are `alloy`, `ash`,
-   * `ballad`, `coral`, `echo`, `sage`, `shimmer`, and `verse`.
-   */
-  voice?: (string & {}) | 'alloy' | 'ash' | 'ballad' | 'coral' | 'echo' | 'sage' | 'shimmer' | 'verse';
-}
-
-/**
- * Additional details about the status.
- */
-export interface RealtimeResponseStatus {
-  /**
-   * A description of the error that caused the response to fail, populated when the
-   * `status` is `failed`.
-   */
-  error?: RealtimeResponseStatus.Error;
-
-  /**
-   * The reason the Response did not complete. For a `cancelled` Response, one of
-   * `turn_detected` (the server VAD detected a new start of speech) or
-   * `client_cancelled` (the client sent a cancel event). For an `incomplete`
-   * Response, one of `max_output_tokens` or `content_filter` (the server-side safety
-   * filter activated and cut off the response).
-   */
-  reason?: 'turn_detected' | 'client_cancelled' | 'max_output_tokens' | 'content_filter';
-
-  /**
-   * The type of error that caused the response to fail, corresponding with the
-   * `status` field (`completed`, `cancelled`, `incomplete`, `failed`).
-   */
-  type?: 'completed' | 'cancelled' | 'incomplete' | 'failed';
-}
-
-export namespace RealtimeResponseStatus {
-  /**
-   * A description of the error that caused the response to fail, populated when the
-   * `status` is `failed`.
-   */
-  export interface Error {
-    /**
-     * Error code, if any.
-     */
-    code?: string;
-
-    /**
-     * The type of error.
-     */
-    type?: string;
-  }
-}
-
-/**
- * Usage statistics for the Response, this will correspond to billing. A Realtime
- * API session will maintain a conversation context and append new Items to the
- * Conversation, thus output from previous turns (text and audio tokens) will
- * become the input for later turns.
- */
-export interface RealtimeResponseUsage {
-  /**
-   * Details about the input tokens used in the Response.
-   */
-  input_token_details?: RealtimeResponseUsage.InputTokenDetails;
-
-  /**
-   * The number of input tokens used in the Response, including text and audio
-   * tokens.
-   */
-  input_tokens?: number;
-
-  /**
-   * Details about the output tokens used in the Response.
-   */
-  output_token_details?: RealtimeResponseUsage.OutputTokenDetails;
-
-  /**
-   * The number of output tokens sent in the Response, including text and audio
-   * tokens.
-   */
-  output_tokens?: number;
-
-  /**
-   * The total number of tokens in the Response including input and output text and
-   * audio tokens.
-   */
-  total_tokens?: number;
-}
-
-export namespace RealtimeResponseUsage {
-  /**
-   * Details about the input tokens used in the Response.
-   */
-  export interface InputTokenDetails {
-    /**
-     * The number of audio tokens used in the Response.
-     */
-    audio_tokens?: number;
-
-    /**
-     * The number of cached tokens used in the Response.
-     */
-    cached_tokens?: number;
-
-    /**
-     * The number of text tokens used in the Response.
-     */
-    text_tokens?: number;
-  }
-
-  /**
-   * Details about the output tokens used in the Response.
-   */
-  export interface OutputTokenDetails {
-    /**
-     * The number of audio tokens used in the Response.
-     */
-    audio_tokens?: number;
-
-    /**
-     * The number of text tokens used in the Response.
-     */
-    text_tokens?: number;
-  }
-}
-
-/**
- * A realtime server event.
- */
-export type RealtimeServerEvent =
-  | ConversationCreatedEvent
-  | ConversationItemCreatedEvent
-  | ConversationItemDeletedEvent
-  | ConversationItemInputAudioTranscriptionCompletedEvent
-  | ConversationItemInputAudioTranscriptionDeltaEvent
-  | ConversationItemInputAudioTranscriptionFailedEvent
-  | RealtimeServerEvent.ConversationItemRetrieved
-  | ConversationItemTruncatedEvent
-  | ErrorEvent
-  | InputAudioBufferClearedEvent
-  | InputAudioBufferCommittedEvent
-  | InputAudioBufferSpeechStartedEvent
-  | InputAudioBufferSpeechStoppedEvent
-  | RateLimitsUpdatedEvent
-  | ResponseAudioDeltaEvent
-  | ResponseAudioDoneEvent
-  | ResponseAudioTranscriptDeltaEvent
-  | ResponseAudioTranscriptDoneEvent
-  | ResponseContentPartAddedEvent
-  | ResponseContentPartDoneEvent
-  | ResponseCreatedEvent
-  | ResponseDoneEvent
-  | ResponseFunctionCallArgumentsDeltaEvent
-  | ResponseFunctionCallArgumentsDoneEvent
-  | ResponseOutputItemAddedEvent
-  | ResponseOutputItemDoneEvent
-  | ResponseTextDeltaEvent
-  | ResponseTextDoneEvent
-  | SessionCreatedEvent
-  | SessionUpdatedEvent
-  | TranscriptionSessionUpdatedEvent
-  | RealtimeServerEvent.OutputAudioBufferStarted
-  | RealtimeServerEvent.OutputAudioBufferStopped
-  | RealtimeServerEvent.OutputAudioBufferCleared;
-
-export namespace RealtimeServerEvent {
-  /**
-   * Returned when a conversation item is retrieved with
-   * `conversation.item.retrieve`.
-   */
-  export interface ConversationItemRetrieved {
-    /**
-     * The unique ID of the server event.
-     */
-    event_id: string;
-
-    /**
-     * The item to add to the conversation.
-     */
-    item: RealtimeAPI.ConversationItem;
-
-    /**
-     * The event type, must be `conversation.item.retrieved`.
-     */
-    type: 'conversation.item.retrieved';
-  }
-
-  /**
-   * **WebRTC Only:** Emitted when the server begins streaming audio to the client.
-   * This event is emitted after an audio content part has been added
-   * (`response.content_part.added`) to the response.
-   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
-   */
-  export interface OutputAudioBufferStarted {
-    /**
-     * The unique ID of the server event.
-     */
-    event_id: string;
-
-    /**
-     * The unique ID of the response that produced the audio.
-     */
-    response_id: string;
-
-    /**
-     * The event type, must be `output_audio_buffer.started`.
-     */
-    type: 'output_audio_buffer.started';
-  }
-
-  /**
-   * **WebRTC Only:** Emitted when the output audio buffer has been completely
-   * drained on the server, and no more audio is forthcoming. This event is emitted
-   * after the full response data has been sent to the client (`response.done`).
-   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
-   */
-  export interface OutputAudioBufferStopped {
-    /**
-     * The unique ID of the server event.
-     */
-    event_id: string;
-
-    /**
-     * The unique ID of the response that produced the audio.
-     */
-    response_id: string;
-
-    /**
-     * The event type, must be `output_audio_buffer.stopped`.
-     */
-    type: 'output_audio_buffer.stopped';
-  }
-
-  /**
-   * **WebRTC Only:** Emitted when the output audio buffer is cleared. This happens
-   * either in VAD mode when the user has interrupted
-   * (`input_audio_buffer.speech_started`), or when the client has emitted the
-   * `output_audio_buffer.clear` event to manually cut off the current audio
-   * response.
-   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
-   */
-  export interface OutputAudioBufferCleared {
-    /**
-     * The unique ID of the server event.
-     */
-    event_id: string;
-
-    /**
-     * The unique ID of the response that produced the audio.
-     */
-    response_id: string;
-
-    /**
-     * The event type, must be `output_audio_buffer.cleared`.
-     */
-    type: 'output_audio_buffer.cleared';
-  }
-}
-
-/**
- * Returned when the model-generated audio is updated.
- */
-export interface ResponseAudioDeltaEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * Base64-encoded audio data delta.
-   */
-  delta: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.audio.delta`.
-   */
-  type: 'response.audio.delta';
-}
-
-/**
- * Returned when the model-generated audio is done. Also emitted when a Response is
- * interrupted, incomplete, or cancelled.
- */
-export interface ResponseAudioDoneEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.audio.done`.
-   */
-  type: 'response.audio.done';
-}
-
-/**
- * Returned when the model-generated transcription of audio output is updated.
- */
-export interface ResponseAudioTranscriptDeltaEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The transcript delta.
-   */
-  delta: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.audio_transcript.delta`.
-   */
-  type: 'response.audio_transcript.delta';
-}
-
-/**
- * Returned when the model-generated transcription of audio output is done
- * streaming. Also emitted when a Response is interrupted, incomplete, or
- * cancelled.
- */
-export interface ResponseAudioTranscriptDoneEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The final transcript of the audio.
-   */
-  transcript: string;
-
-  /**
-   * The event type, must be `response.audio_transcript.done`.
-   */
-  type: 'response.audio_transcript.done';
-}
-
-/**
- * Send this event to cancel an in-progress response. The server will respond with
- * a `response.done` event with a status of `response.status=cancelled`. If there
- * is no response to cancel, the server will respond with an error.
- */
-export interface ResponseCancelEvent {
-  /**
-   * The event type, must be `response.cancel`.
-   */
-  type: 'response.cancel';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-
-  /**
-   * A specific response ID to cancel - if not provided, will cancel an in-progress
-   * response in the default conversation.
-   */
-  response_id?: string;
-}
-
-/**
- * Returned when a new content part is added to an assistant message item during
- * response generation.
- */
-export interface ResponseContentPartAddedEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item to which the content part was added.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The content part that was added.
-   */
-  part: ResponseContentPartAddedEvent.Part;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.content_part.added`.
-   */
-  type: 'response.content_part.added';
-}
-
-export namespace ResponseContentPartAddedEvent {
-  /**
-   * The content part that was added.
-   */
-  export interface Part {
-    /**
-     * Base64-encoded audio data (if type is "audio").
-     */
-    audio?: string;
-
-    /**
-     * The text content (if type is "text").
-     */
-    text?: string;
-
-    /**
-     * The transcript of the audio (if type is "audio").
-     */
-    transcript?: string;
-
-    /**
-     * The content type ("text", "audio").
-     */
-    type?: 'text' | 'audio';
-  }
-}
-
-/**
- * Returned when a content part is done streaming in an assistant message item.
- * Also emitted when a Response is interrupted, incomplete, or cancelled.
- */
-export interface ResponseContentPartDoneEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The content part that is done.
-   */
-  part: ResponseContentPartDoneEvent.Part;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.content_part.done`.
-   */
-  type: 'response.content_part.done';
-}
-
-export namespace ResponseContentPartDoneEvent {
-  /**
-   * The content part that is done.
-   */
-  export interface Part {
-    /**
-     * Base64-encoded audio data (if type is "audio").
-     */
-    audio?: string;
-
-    /**
-     * The text content (if type is "text").
-     */
-    text?: string;
-
-    /**
-     * The transcript of the audio (if type is "audio").
-     */
-    transcript?: string;
-
-    /**
-     * The content type ("text", "audio").
-     */
-    type?: 'text' | 'audio';
-  }
-}
-
-/**
- * This event instructs the server to create a Response, which means triggering
- * model inference. When in Server VAD mode, the server will create Responses
- * automatically.
- *
- * A Response will include at least one Item, and may have two, in which case the
- * second will be a function call. These Items will be appended to the conversation
- * history.
- *
- * The server will respond with a `response.created` event, events for Items and
- * content created, and finally a `response.done` event to indicate the Response is
- * complete.
- *
- * The `response.create` event includes inference configuration like
- * `instructions`, and `temperature`. These fields will override the Session's
- * configuration for this Response only.
- */
-export interface ResponseCreateEvent {
-  /**
-   * The event type, must be `response.create`.
-   */
-  type: 'response.create';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-
-  /**
-   * Create a new Realtime response with these parameters
-   */
-  response?: ResponseCreateEvent.Response;
-}
-
-export namespace ResponseCreateEvent {
-  /**
-   * Create a new Realtime response with these parameters
-   */
-  export interface Response {
-    /**
-     * Controls which conversation the response is added to. Currently supports `auto`
-     * and `none`, with `auto` as the default value. The `auto` value means that the
-     * contents of the response will be added to the default conversation. Set this to
-     * `none` to create an out-of-band response which will not add items to default
-     * conversation.
-     */
-    conversation?: (string & {}) | 'auto' | 'none';
-
-    /**
-     * Input items to include in the prompt for the model. Using this field creates a
-     * new context for this Response instead of using the default conversation. An
-     * empty array `[]` will clear the context for this Response. Note that this can
-     * include references to items from the default conversation.
-     */
-    input?: Array<RealtimeAPI.ConversationItemWithReference>;
-
-    /**
-     * 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';
-
-    /**
-     * Set of 16 key-value pairs that can be attached to an object. This can be useful
-     * for storing additional information about the object in a structured format, and
-     * querying for objects via API or the dashboard.
-     *
-     * Keys are strings with a maximum length of 64 characters. Values are strings with
-     * a maximum length of 512 characters.
-     */
-    metadata?: Shared.Metadata | null;
-
-    /**
-     * 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?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
-
-    /**
-     * 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, like `{"type": "function", "function": {"name": "my_function"}}`.
-     */
-    tool_choice?: string;
-
-    /**
-     * Tools (functions) available to the model.
-     */
-    tools?: Array<Response.Tool>;
-
-    /**
-     * 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' | 'sage' | 'shimmer' | 'verse';
-  }
-
-  export namespace Response {
-    export 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';
-    }
-  }
-}
-
-/**
- * Returned when a new Response is created. The first event of response creation,
- * where the response is in an initial state of `in_progress`.
- */
-export interface ResponseCreatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The response resource.
-   */
-  response: RealtimeResponse;
-
-  /**
-   * The event type, must be `response.created`.
-   */
-  type: 'response.created';
-}
-
-/**
- * Returned when a Response is done streaming. Always emitted, no matter the final
- * state. The Response object included in the `response.done` event will include
- * all output Items in the Response but will omit the raw audio data.
- */
-export interface ResponseDoneEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The response resource.
-   */
-  response: RealtimeResponse;
-
-  /**
-   * The event type, must be `response.done`.
-   */
-  type: 'response.done';
-}
-
-/**
- * Returned when the model-generated function call arguments are updated.
- */
-export interface ResponseFunctionCallArgumentsDeltaEvent {
-  /**
-   * The ID of the function call.
-   */
-  call_id: string;
-
-  /**
-   * The arguments delta as a JSON string.
-   */
-  delta: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the function call item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.function_call_arguments.delta`.
-   */
-  type: 'response.function_call_arguments.delta';
-}
-
-/**
- * Returned when the model-generated function call arguments are done streaming.
- * Also emitted when a Response is interrupted, incomplete, or cancelled.
- */
-export interface ResponseFunctionCallArgumentsDoneEvent {
-  /**
-   * The final arguments as a JSON string.
-   */
-  arguments: string;
-
-  /**
-   * The ID of the function call.
-   */
-  call_id: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the function call item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.function_call_arguments.done`.
-   */
-  type: 'response.function_call_arguments.done';
-}
-
-/**
- * Returned when a new Item is created during Response generation.
- */
-export interface ResponseOutputItemAddedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The item to add to the conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The index of the output item in the Response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the Response to which the item belongs.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.output_item.added`.
-   */
-  type: 'response.output_item.added';
-}
-
-/**
- * Returned when an Item is done streaming. Also emitted when a Response is
- * interrupted, incomplete, or cancelled.
- */
-export interface ResponseOutputItemDoneEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The item to add to the conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The index of the output item in the Response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the Response to which the item belongs.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.output_item.done`.
-   */
-  type: 'response.output_item.done';
-}
-
-/**
- * Returned when the text value of a "text" content part is updated.
- */
-export interface ResponseTextDeltaEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The text delta.
-   */
-  delta: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The event type, must be `response.text.delta`.
-   */
-  type: 'response.text.delta';
-}
-
-/**
- * Returned when the text value of a "text" content part is done streaming. Also
- * emitted when a Response is interrupted, incomplete, or cancelled.
- */
-export interface ResponseTextDoneEvent {
-  /**
-   * The index of the content part in the item's content array.
-   */
-  content_index: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The ID of the response.
-   */
-  response_id: string;
-
-  /**
-   * The final text content.
-   */
-  text: string;
-
-  /**
-   * The event type, must be `response.text.done`.
-   */
-  type: 'response.text.done';
-}
-
-/**
- * Returned when a Session is created. Emitted automatically when a new connection
- * is established as the first server event. This event will contain the default
- * Session configuration.
- */
-export interface SessionCreatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * Realtime session object configuration.
-   */
-  session: SessionsAPI.Session;
-
-  /**
-   * The event type, must be `session.created`.
-   */
-  type: 'session.created';
-}
-
-/**
- * Send this event to update the session’s default configuration. The client may
- * send this event at any time to update any field, except for `voice`. However,
- * note that once a session has been initialized with a particular `model`, it
- * can’t be changed to another model using `session.update`.
- *
- * When the server receives a `session.update`, it will respond with a
- * `session.updated` event showing the full, effective configuration. Only the
- * fields that are present are updated. To clear a field like `instructions`, pass
- * an empty string.
- */
-export interface SessionUpdateEvent {
-  /**
-   * Realtime session object configuration.
-   */
-  session: SessionUpdateEvent.Session;
-
-  /**
-   * The event type, must be `session.update`.
-   */
-  type: 'session.update';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-export namespace SessionUpdateEvent {
-  /**
-   * Realtime session object configuration.
-   */
-  export interface Session {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    client_secret?: Session.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?: 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-realtime-preview-2025-06-03'
-      | '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';
-
-    /**
-     * The speed of the model's spoken response. 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.
-     */
-    speed?: number;
-
-    /**
-     * 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 options for tracing. 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' | Session.TracingConfiguration;
-
-    /**
-     * 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?: 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' | 'sage' | 'shimmer' | 'verse';
-  }
-
-  export namespace Session {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    export interface ClientSecret {
-      /**
-       * Configuration for the ephemeral token expiration.
-       */
-      expires_after?: ClientSecret.ExpiresAfter;
-    }
-
-    export namespace ClientSecret {
-      /**
-       * Configuration for the ephemeral token expiration.
-       */
-      export interface ExpiresAfter {
-        /**
-         * 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.
-     */
-    export 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.
-     */
-    export 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;
-    }
-
-    export 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';
-    }
-
-    /**
-     * 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;
-    }
-
-    /**
-     * 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.
-     */
-    export 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';
-    }
-  }
-}
-
-/**
- * Returned when a session is updated with a `session.update` event, unless there
- * is an error.
- */
-export interface SessionUpdatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * Realtime session object configuration.
-   */
-  session: SessionsAPI.Session;
-
-  /**
-   * The event type, must be `session.updated`.
-   */
-  type: 'session.updated';
-}
-
-/**
- * Send this event to update a transcription session.
- */
-export interface TranscriptionSessionUpdate {
-  /**
-   * Realtime transcription session object configuration.
-   */
-  session: TranscriptionSessionUpdate.Session;
-
-  /**
-   * The event type, must be `transcription_session.update`.
-   */
-  type: 'transcription_session.update';
-
-  /**
-   * Optional client-generated ID used to identify this event.
-   */
-  event_id?: string;
-}
-
-export namespace TranscriptionSessionUpdate {
-  /**
-   * Realtime transcription session object configuration.
-   */
-  export interface Session {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    client_secret?: Session.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?: Session.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?: Session.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?: Session.TurnDetection;
-  }
-
-  export namespace Session {
-    /**
-     * Configuration options for the generated client secret.
-     */
-    export interface ClientSecret {
-      /**
-       * Configuration for the ephemeral token expiration.
-       */
-      expires_at?: ClientSecret.ExpiresAt;
-    }
-
-    export namespace ClientSecret {
-      /**
-       * Configuration for the ephemeral token expiration.
-       */
-      export 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.
-     */
-    export 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.
-     */
-    export 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.
-     */
-    export 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';
-    }
-  }
-}
-
-/**
- * Returned when a transcription session is updated with a
- * `transcription_session.update` event, unless there is an error.
- */
-export interface TranscriptionSessionUpdatedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * 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.
-   */
-  session: TranscriptionSessionsAPI.TranscriptionSession;
-
-  /**
-   * The event type, must be `transcription_session.updated`.
-   */
-  type: 'transcription_session.updated';
-}
-
-Realtime.Sessions = Sessions;
-Realtime.TranscriptionSessions = TranscriptionSessions;
-
-export declare namespace Realtime {
-  export {
-    type ConversationCreatedEvent as ConversationCreatedEvent,
-    type ConversationItem as ConversationItem,
-    type ConversationItemContent as ConversationItemContent,
-    type ConversationItemCreateEvent as ConversationItemCreateEvent,
-    type ConversationItemCreatedEvent as ConversationItemCreatedEvent,
-    type ConversationItemDeleteEvent as ConversationItemDeleteEvent,
-    type ConversationItemDeletedEvent as ConversationItemDeletedEvent,
-    type ConversationItemInputAudioTranscriptionCompletedEvent as ConversationItemInputAudioTranscriptionCompletedEvent,
-    type ConversationItemInputAudioTranscriptionDeltaEvent as ConversationItemInputAudioTranscriptionDeltaEvent,
-    type ConversationItemInputAudioTranscriptionFailedEvent as ConversationItemInputAudioTranscriptionFailedEvent,
-    type ConversationItemRetrieveEvent as ConversationItemRetrieveEvent,
-    type ConversationItemTruncateEvent as ConversationItemTruncateEvent,
-    type ConversationItemTruncatedEvent as ConversationItemTruncatedEvent,
-    type ConversationItemWithReference as ConversationItemWithReference,
-    type ErrorEvent as ErrorEvent,
-    type InputAudioBufferAppendEvent as InputAudioBufferAppendEvent,
-    type InputAudioBufferClearEvent as InputAudioBufferClearEvent,
-    type InputAudioBufferClearedEvent as InputAudioBufferClearedEvent,
-    type InputAudioBufferCommitEvent as InputAudioBufferCommitEvent,
-    type InputAudioBufferCommittedEvent as InputAudioBufferCommittedEvent,
-    type InputAudioBufferSpeechStartedEvent as InputAudioBufferSpeechStartedEvent,
-    type InputAudioBufferSpeechStoppedEvent as InputAudioBufferSpeechStoppedEvent,
-    type RateLimitsUpdatedEvent as RateLimitsUpdatedEvent,
-    type RealtimeClientEvent as RealtimeClientEvent,
-    type RealtimeResponse as RealtimeResponse,
-    type RealtimeResponseStatus as RealtimeResponseStatus,
-    type RealtimeResponseUsage as RealtimeResponseUsage,
-    type RealtimeServerEvent as RealtimeServerEvent,
-    type ResponseAudioDeltaEvent as ResponseAudioDeltaEvent,
-    type ResponseAudioDoneEvent as ResponseAudioDoneEvent,
-    type ResponseAudioTranscriptDeltaEvent as ResponseAudioTranscriptDeltaEvent,
-    type ResponseAudioTranscriptDoneEvent as ResponseAudioTranscriptDoneEvent,
-    type ResponseCancelEvent as ResponseCancelEvent,
-    type ResponseContentPartAddedEvent as ResponseContentPartAddedEvent,
-    type ResponseContentPartDoneEvent as ResponseContentPartDoneEvent,
-    type ResponseCreateEvent as ResponseCreateEvent,
-    type ResponseCreatedEvent as ResponseCreatedEvent,
-    type ResponseDoneEvent as ResponseDoneEvent,
-    type ResponseFunctionCallArgumentsDeltaEvent as ResponseFunctionCallArgumentsDeltaEvent,
-    type ResponseFunctionCallArgumentsDoneEvent as ResponseFunctionCallArgumentsDoneEvent,
-    type ResponseOutputItemAddedEvent as ResponseOutputItemAddedEvent,
-    type ResponseOutputItemDoneEvent as ResponseOutputItemDoneEvent,
-    type ResponseTextDeltaEvent as ResponseTextDeltaEvent,
-    type ResponseTextDoneEvent as ResponseTextDoneEvent,
-    type SessionCreatedEvent as SessionCreatedEvent,
-    type SessionUpdateEvent as SessionUpdateEvent,
-    type SessionUpdatedEvent as SessionUpdatedEvent,
-    type TranscriptionSessionUpdate as TranscriptionSessionUpdate,
-    type TranscriptionSessionUpdatedEvent as TranscriptionSessionUpdatedEvent,
-  };
-
-  export {
-    Sessions as Sessions,
-    type SessionsAPISession as Session,
-    type SessionCreateResponse as SessionCreateResponse,
-    type SessionCreateParams as SessionCreateParams,
-  };
-
-  export {
-    TranscriptionSessions as TranscriptionSessions,
-    type TranscriptionSession as TranscriptionSession,
-    type TranscriptionSessionCreateParams as TranscriptionSessionCreateParams,
-  };
-}