@jsonstudio/rcc 0.89.555 → 0.89.682

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

@@ -1,4823 +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 CallsAPI from './calls';
-import { CallAcceptParams, CallReferParams, CallRejectParams, Calls } from './calls';
-import * as ClientSecretsAPI from './client-secrets';
-import {
-  ClientSecretCreateParams,
-  ClientSecretCreateResponse,
-  ClientSecrets,
-  RealtimeSessionClientSecret,
-  RealtimeSessionCreateResponse,
-  RealtimeTranscriptionSessionCreateResponse,
-  RealtimeTranscriptionSessionTurnDetection,
-} from './client-secrets';
-import * as ResponsesAPI from '../responses/responses';
-
-export class Realtime extends APIResource {
-  clientSecrets: ClientSecretsAPI.ClientSecrets = new ClientSecretsAPI.ClientSecrets(this._client);
-  calls: CallsAPI.Calls = new CallsAPI.Calls(this._client);
-}
-
-export interface AudioTranscription {
-  /**
-   * 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 `whisper-1`,
-   * `gpt-4o-mini-transcribe`, `gpt-4o-transcribe`, and `gpt-4o-transcribe-diarize`.
-   * Use `gpt-4o-transcribe-diarize` when you need diarization with speaker labels.
-   */
-  model?: 'whisper-1' | 'gpt-4o-mini-transcribe' | 'gpt-4o-transcribe' | 'gpt-4o-transcribe-diarize';
-
-  /**
-   * 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 (excluding `gpt-4o-transcribe-diarize`), the
-   * prompt is a free text string, for example "expect words related to technology".
-   */
-  prompt?: string;
-}
-
-/**
- * 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';
-  }
-}
-
-/**
- * A single item within a Realtime conversation.
- */
-export type ConversationItem =
-  | RealtimeConversationItemSystemMessage
-  | RealtimeConversationItemUserMessage
-  | RealtimeConversationItemAssistantMessage
-  | RealtimeConversationItemFunctionCall
-  | RealtimeConversationItemFunctionCallOutput
-  | RealtimeMcpApprovalResponse
-  | RealtimeMcpListTools
-  | RealtimeMcpToolCall
-  | RealtimeMcpApprovalRequest;
-
-/**
- * Sent by the server when an Item is added to the default Conversation. This can
- * happen in several cases:
- *
- * - When the client sends a `conversation.item.create` event.
- * - When the input audio buffer is committed. In this case the item will be a user
- *   message containing the audio from the buffer.
- * - When the model is generating a Response. In this case the
- *   `conversation.item.added` event will be sent when the model starts generating
- *   a specific Item, and thus it will not yet have any content (and `status` will
- *   be `in_progress`).
- *
- * The event will include the full content of the Item (except when model is
- * generating a Response) except for audio data, which can be retrieved separately
- * with a `conversation.item.retrieve` event if necessary.
- */
-export interface ConversationItemAdded {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * A single item within a Realtime conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The event type, must be `conversation.item.added`.
-   */
-  type: 'conversation.item.added';
-
-  /**
-   * The ID of the item that precedes this one, if any. This is used to maintain
-   * ordering when items are inserted.
-   */
-  previous_item_id?: string | null;
-}
-
-/**
- * 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 {
-  /**
-   * A single item within a Realtime 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;
-
-  /**
-   * A single item within a Realtime 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';
-}
-
-/**
- * Returned when a conversation item is finalized.
- *
- * The event will include the full content of the Item except for audio data, which
- * can be retrieved separately with a `conversation.item.retrieve` event if needed.
- */
-export interface ConversationItemDone {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * A single item within a Realtime conversation.
-   */
-  item: ConversationItem;
-
-  /**
-   * The event type, must be `conversation.item.done`.
-   */
-  type: 'conversation.item.done';
-
-  /**
-   * The ID of the item that precedes this one, if any. This is used to maintain
-   * ordering when items are inserted.
-   */
-  previous_item_id?: string | null;
-}
-
-/**
- * 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 (when VAD is enabled). 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 item containing the audio that is being transcribed.
-   */
-  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, this is billed according to the ASR
-   * model's pricing rather than the realtime model's pricing.
-   */
-  usage:
-    | ConversationItemInputAudioTranscriptionCompletedEvent.TranscriptTextUsageTokens
-    | ConversationItemInputAudioTranscriptionCompletedEvent.TranscriptTextUsageDuration;
-
-  /**
-   * The log probabilities of the transcription.
-   */
-  logprobs?: Array<LogProbProperties> | 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';
-  }
-}
-
-/**
- * Returned when the text value of an input audio transcription content part is
- * updated with incremental transcription results.
- */
-export interface ConversationItemInputAudioTranscriptionDeltaEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item containing the audio that is being transcribed.
-   */
-  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. These can be enabled by
-   * configurating the session with
-   * `"include": ["item.input_audio_transcription.logprobs"]`. Each entry in the
-   * array corresponds a log probability of which token would be selected for this
-   * chunk of transcription. This can help to identify if it was possible there were
-   * multiple valid options for a given chunk of transcription.
-   */
-  logprobs?: Array<LogProbProperties> | null;
-}
-
-/**
- * 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;
-  }
-}
-
-/**
- * Returned when an input audio transcription segment is identified for an item.
- */
-export interface ConversationItemInputAudioTranscriptionSegment {
-  /**
-   * The segment identifier.
-   */
-  id: string;
-
-  /**
-   * The index of the input audio content part within the item.
-   */
-  content_index: number;
-
-  /**
-   * End time of the segment in seconds.
-   */
-  end: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item containing the input audio content.
-   */
-  item_id: string;
-
-  /**
-   * The detected speaker label for this segment.
-   */
-  speaker: string;
-
-  /**
-   * Start time of the segment in seconds.
-   */
-  start: number;
-
-  /**
-   * The text for this segment.
-   */
-  text: string;
-
-  /**
-   * The event type, must be `conversation.item.input_audio_transcription.segment`.
-   */
-  type: 'conversation.item.input_audio_transcription.segment';
-}
-
-/**
- * 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';
-  }
-}
-
-/**
- * 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. A "commit" will
- * create a new user message item in the conversation history from the buffer
- * content and clear the buffer. Input audio transcription (if enabled) will be
- * generated when the buffer is committed.
- *
- * If VAD is enabled 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. Input audio noise reduction operates on writes to the audio
- * buffer.
- *
- * 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 most 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;
-}
-
-/**
- * **SIP Only:** Returned when an DTMF event is received. A DTMF event is a message
- * that represents a telephone keypad press (0–9, \*, #, A–D). The `event` property
- * is the keypad that the user press. The `received_at` is the UTC Unix Timestamp
- * that the server received the event.
- */
-export interface InputAudioBufferDtmfEventReceivedEvent {
-  /**
-   * The telephone keypad that was pressed by the user.
-   */
-  event: string;
-
-  /**
-   * UTC Unix Timestamp when DTMF Event was received by server.
-   */
-  received_at: number;
-
-  /**
-   * The event type, must be `input_audio_buffer.dtmf_event_received`.
-   */
-  type: 'input_audio_buffer.dtmf_event_received';
-}
-
-/**
- * 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';
-}
-
-/**
- * Returned when the Server VAD timeout is triggered for the input audio buffer.
- * This is configured with `idle_timeout_ms` in the `turn_detection` settings of
- * the session, and it indicates that there hasn't been any speech detected for the
- * configured duration.
- *
- * The `audio_start_ms` and `audio_end_ms` fields indicate the segment of audio
- * after the last model response up to the triggering time, as an offset from the
- * beginning of audio written to the input audio buffer. This means it demarcates
- * the segment of audio that was silent and the difference between the start and
- * end values will roughly match the configured timeout.
- *
- * The empty audio will be committed to the conversation as an `input_audio` item
- * (there will be a `input_audio_buffer.committed` event) and a model response will
- * be generated. There may be speech that didn't trigger VAD but is still detected
- * by the model, so the model may respond with something relevant to the
- * conversation or a prompt to continue speaking.
- */
-export interface InputAudioBufferTimeoutTriggered {
-  /**
-   * Millisecond offset of audio written to the input audio buffer at the time the
-   * timeout was triggered.
-   */
-  audio_end_ms: number;
-
-  /**
-   * Millisecond offset of audio written to the input audio buffer that was after the
-   * playback time of the last model response.
-   */
-  audio_start_ms: number;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the item associated with this segment.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `input_audio_buffer.timeout_triggered`.
-   */
-  type: 'input_audio_buffer.timeout_triggered';
-}
-
-/**
- * A log probability object.
- */
-export interface LogProbProperties {
-  /**
-   * 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 listing MCP tools has completed for an item.
- */
-export interface McpListToolsCompleted {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP list tools item.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `mcp_list_tools.completed`.
-   */
-  type: 'mcp_list_tools.completed';
-}
-
-/**
- * Returned when listing MCP tools has failed for an item.
- */
-export interface McpListToolsFailed {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP list tools item.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `mcp_list_tools.failed`.
-   */
-  type: 'mcp_list_tools.failed';
-}
-
-/**
- * Returned when listing MCP tools is in progress for an item.
- */
-export interface McpListToolsInProgress {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP list tools item.
-   */
-  item_id: string;
-
-  /**
-   * The event type, must be `mcp_list_tools.in_progress`.
-   */
-  type: 'mcp_list_tools.in_progress';
-}
-
-/**
- * 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.
- */
-export type NoiseReductionType = 'near_field' | 'far_field';
-
-/**
- * **WebRTC/SIP 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 OutputAudioBufferClearEvent {
-  /**
-   * 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;
-}
-
-/**
- * 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;
-  }
-}
-
-/**
- * Configuration for input and output audio.
- */
-export interface RealtimeAudioConfig {
-  input?: RealtimeAudioConfigInput;
-
-  output?: RealtimeAudioConfigOutput;
-}
-
-export interface RealtimeAudioConfigInput {
-  /**
-   * The format of the input audio.
-   */
-  format?: 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?: RealtimeAudioConfigInput.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?: 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?: RealtimeAudioInputTurnDetection | null;
-}
-
-export namespace RealtimeAudioConfigInput {
-  /**
-   * 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;
-  }
-}
-
-export interface RealtimeAudioConfigOutput {
-  /**
-   * The format of the output audio.
-   */
-  format?: 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';
-}
-
-/**
- * The PCM audio format. Only a 24kHz sample rate is supported.
- */
-export type RealtimeAudioFormats =
-  | RealtimeAudioFormats.AudioPCM
-  | RealtimeAudioFormats.AudioPCMU
-  | RealtimeAudioFormats.AudioPCMA;
-
-export namespace RealtimeAudioFormats {
-  /**
-   * The PCM audio format. Only a 24kHz sample rate is supported.
-   */
-  export interface AudioPCM {
-    /**
-     * The sample rate of the audio. Always `24000`.
-     */
-    rate?: 24000;
-
-    /**
-     * The audio format. Always `audio/pcm`.
-     */
-    type?: 'audio/pcm';
-  }
-
-  /**
-   * The G.711 μ-law format.
-   */
-  export interface AudioPCMU {
-    /**
-     * The audio format. Always `audio/pcmu`.
-     */
-    type?: 'audio/pcmu';
-  }
-
-  /**
-   * The G.711 A-law format.
-   */
-  export interface AudioPCMA {
-    /**
-     * The audio format. Always `audio/pcma`.
-     */
-    type?: 'audio/pcma';
-  }
-}
-
-/**
- * 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 type RealtimeAudioInputTurnDetection =
-  | RealtimeAudioInputTurnDetection.ServerVad
-  | RealtimeAudioInputTurnDetection.SemanticVad;
-
-export namespace RealtimeAudioInputTurnDetection {
-  /**
-   * 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;
-  }
-}
-
-/**
- * A realtime client event.
- */
-export type RealtimeClientEvent =
-  | ConversationItemCreateEvent
-  | ConversationItemDeleteEvent
-  | ConversationItemRetrieveEvent
-  | ConversationItemTruncateEvent
-  | InputAudioBufferAppendEvent
-  | InputAudioBufferClearEvent
-  | OutputAudioBufferClearEvent
-  | InputAudioBufferCommitEvent
-  | ResponseCancelEvent
-  | ResponseCreateEvent
-  | SessionUpdateEvent;
-
-/**
- * An assistant message item in a Realtime conversation.
- */
-export interface RealtimeConversationItemAssistantMessage {
-  /**
-   * The content of the message.
-   */
-  content: Array<RealtimeConversationItemAssistantMessage.Content>;
-
-  /**
-   * The role of the message sender. Always `assistant`.
-   */
-  role: 'assistant';
-
-  /**
-   * The type of the item. Always `message`.
-   */
-  type: 'message';
-
-  /**
-   * The unique ID of the item. This may be provided by the client or generated by
-   * the server.
-   */
-  id?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`. Optional
-   * when creating a new item.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The status of the item. Has no effect on the conversation.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-}
-
-export namespace RealtimeConversationItemAssistantMessage {
-  export interface Content {
-    /**
-     * Base64-encoded audio bytes, these will be parsed as the format specified in the
-     * session output audio type configuration. This defaults to PCM 16-bit 24kHz mono
-     * if not specified.
-     */
-    audio?: string;
-
-    /**
-     * The text content.
-     */
-    text?: string;
-
-    /**
-     * The transcript of the audio content, this will always be present if the output
-     * type is `audio`.
-     */
-    transcript?: string;
-
-    /**
-     * The content type, `output_text` or `output_audio` depending on the session
-     * `output_modalities` configuration.
-     */
-    type?: 'output_text' | 'output_audio';
-  }
-}
-
-/**
- * A function call item in a Realtime conversation.
- */
-export interface RealtimeConversationItemFunctionCall {
-  /**
-   * The arguments of the function call. This is a JSON-encoded string representing
-   * the arguments passed to the function, for example
-   * `{"arg1": "value1", "arg2": 42}`.
-   */
-  arguments: string;
-
-  /**
-   * The name of the function being called.
-   */
-  name: string;
-
-  /**
-   * The type of the item. Always `function_call`.
-   */
-  type: 'function_call';
-
-  /**
-   * The unique ID of the item. This may be provided by the client or generated by
-   * the server.
-   */
-  id?: string;
-
-  /**
-   * The ID of the function call.
-   */
-  call_id?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`. Optional
-   * when creating a new item.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The status of the item. Has no effect on the conversation.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-}
-
-/**
- * A function call output item in a Realtime conversation.
- */
-export interface RealtimeConversationItemFunctionCallOutput {
-  /**
-   * The ID of the function call this output is for.
-   */
-  call_id: string;
-
-  /**
-   * The output of the function call, this is free text and can contain any
-   * information or simply be empty.
-   */
-  output: string;
-
-  /**
-   * The type of the item. Always `function_call_output`.
-   */
-  type: 'function_call_output';
-
-  /**
-   * The unique ID of the item. This may be provided by the client or generated by
-   * the server.
-   */
-  id?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`. Optional
-   * when creating a new item.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The status of the item. Has no effect on the conversation.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-}
-
-/**
- * A system message in a Realtime conversation can be used to provide additional
- * context or instructions to the model. This is similar but distinct from the
- * instruction prompt provided at the start of a conversation, as system messages
- * can be added at any point in the conversation. For major changes to the
- * conversation's behavior, use instructions, but for smaller updates (e.g. "the
- * user is now asking about a different topic"), use system messages.
- */
-export interface RealtimeConversationItemSystemMessage {
-  /**
-   * The content of the message.
-   */
-  content: Array<RealtimeConversationItemSystemMessage.Content>;
-
-  /**
-   * The role of the message sender. Always `system`.
-   */
-  role: 'system';
-
-  /**
-   * The type of the item. Always `message`.
-   */
-  type: 'message';
-
-  /**
-   * The unique ID of the item. This may be provided by the client or generated by
-   * the server.
-   */
-  id?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`. Optional
-   * when creating a new item.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The status of the item. Has no effect on the conversation.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-}
-
-export namespace RealtimeConversationItemSystemMessage {
-  export interface Content {
-    /**
-     * The text content.
-     */
-    text?: string;
-
-    /**
-     * The content type. Always `input_text` for system messages.
-     */
-    type?: 'input_text';
-  }
-}
-
-/**
- * A user message item in a Realtime conversation.
- */
-export interface RealtimeConversationItemUserMessage {
-  /**
-   * The content of the message.
-   */
-  content: Array<RealtimeConversationItemUserMessage.Content>;
-
-  /**
-   * The role of the message sender. Always `user`.
-   */
-  role: 'user';
-
-  /**
-   * The type of the item. Always `message`.
-   */
-  type: 'message';
-
-  /**
-   * The unique ID of the item. This may be provided by the client or generated by
-   * the server.
-   */
-  id?: string;
-
-  /**
-   * Identifier for the API object being returned - always `realtime.item`. Optional
-   * when creating a new item.
-   */
-  object?: 'realtime.item';
-
-  /**
-   * The status of the item. Has no effect on the conversation.
-   */
-  status?: 'completed' | 'incomplete' | 'in_progress';
-}
-
-export namespace RealtimeConversationItemUserMessage {
-  export interface Content {
-    /**
-     * Base64-encoded audio bytes (for `input_audio`), these will be parsed as the
-     * format specified in the session input audio type configuration. This defaults to
-     * PCM 16-bit 24kHz mono if not specified.
-     */
-    audio?: string;
-
-    /**
-     * The detail level of the image (for `input_image`). `auto` will default to
-     * `high`.
-     */
-    detail?: 'auto' | 'low' | 'high';
-
-    /**
-     * Base64-encoded image bytes (for `input_image`) as a data URI. For example
-     * `data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAA...`. Supported formats are PNG
-     * and JPEG.
-     */
-    image_url?: string;
-
-    /**
-     * The text content (for `input_text`).
-     */
-    text?: string;
-
-    /**
-     * Transcript of the audio (for `input_audio`). This is not sent to the model, but
-     * will be attached to the message item for reference.
-     */
-    transcript?: string;
-
-    /**
-     * The content type (`input_text`, `input_audio`, or `input_image`).
-     */
-    type?: 'input_text' | 'input_audio' | 'input_image';
-  }
-}
-
-/**
- * Details of the error.
- */
-export interface RealtimeError {
-  /**
-   * 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;
-}
-
-/**
- * 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 RealtimeErrorEvent {
-  /**
-   * Details of the error.
-   */
-  error: RealtimeError;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The event type, must be `error`.
-   */
-  type: 'error';
-}
-
-export interface RealtimeFunctionTool {
-  /**
-   * 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';
-}
-
-/**
- * A Realtime item requesting human approval of a tool invocation.
- */
-export interface RealtimeMcpApprovalRequest {
-  /**
-   * The unique ID of the approval request.
-   */
-  id: string;
-
-  /**
-   * A JSON string of arguments for the tool.
-   */
-  arguments: string;
-
-  /**
-   * The name of the tool to run.
-   */
-  name: string;
-
-  /**
-   * The label of the MCP server making the request.
-   */
-  server_label: string;
-
-  /**
-   * The type of the item. Always `mcp_approval_request`.
-   */
-  type: 'mcp_approval_request';
-}
-
-/**
- * A Realtime item responding to an MCP approval request.
- */
-export interface RealtimeMcpApprovalResponse {
-  /**
-   * The unique ID of the approval response.
-   */
-  id: string;
-
-  /**
-   * The ID of the approval request being answered.
-   */
-  approval_request_id: string;
-
-  /**
-   * Whether the request was approved.
-   */
-  approve: boolean;
-
-  /**
-   * The type of the item. Always `mcp_approval_response`.
-   */
-  type: 'mcp_approval_response';
-
-  /**
-   * Optional reason for the decision.
-   */
-  reason?: string | null;
-}
-
-/**
- * A Realtime item listing tools available on an MCP server.
- */
-export interface RealtimeMcpListTools {
-  /**
-   * The label of the MCP server.
-   */
-  server_label: string;
-
-  /**
-   * The tools available on the server.
-   */
-  tools: Array<RealtimeMcpListTools.Tool>;
-
-  /**
-   * The type of the item. Always `mcp_list_tools`.
-   */
-  type: 'mcp_list_tools';
-
-  /**
-   * The unique ID of the list.
-   */
-  id?: string;
-}
-
-export namespace RealtimeMcpListTools {
-  /**
-   * A tool available on an MCP server.
-   */
-  export interface Tool {
-    /**
-     * The JSON schema describing the tool's input.
-     */
-    input_schema: unknown;
-
-    /**
-     * The name of the tool.
-     */
-    name: string;
-
-    /**
-     * Additional annotations about the tool.
-     */
-    annotations?: unknown | null;
-
-    /**
-     * The description of the tool.
-     */
-    description?: string | null;
-  }
-}
-
-export interface RealtimeMcpProtocolError {
-  code: number;
-
-  message: string;
-
-  type: 'protocol_error';
-}
-
-/**
- * A Realtime item representing an invocation of a tool on an MCP server.
- */
-export interface RealtimeMcpToolCall {
-  /**
-   * The unique ID of the tool call.
-   */
-  id: string;
-
-  /**
-   * A JSON string of the arguments passed to the tool.
-   */
-  arguments: string;
-
-  /**
-   * The name of the tool that was run.
-   */
-  name: string;
-
-  /**
-   * The label of the MCP server running the tool.
-   */
-  server_label: string;
-
-  /**
-   * The type of the item. Always `mcp_call`.
-   */
-  type: 'mcp_call';
-
-  /**
-   * The ID of an associated approval request, if any.
-   */
-  approval_request_id?: string | null;
-
-  /**
-   * The error from the tool call, if any.
-   */
-  error?: RealtimeMcpProtocolError | RealtimeMcpToolExecutionError | RealtimeMcphttpError | null;
-
-  /**
-   * The output from the tool call.
-   */
-  output?: string | null;
-}
-
-export interface RealtimeMcpToolExecutionError {
-  message: string;
-
-  type: 'tool_execution_error';
-}
-
-export interface RealtimeMcphttpError {
-  code: number;
-
-  message: string;
-
-  type: 'http_error';
-}
-
-/**
- * The response resource.
- */
-export interface RealtimeResponse {
-  /**
-   * The unique ID of the response, will look like `resp_1234`.
-   */
-  id?: string;
-
-  /**
-   * Configuration for audio output.
-   */
-  audio?: RealtimeResponse.Audio;
-
-  /**
-   * 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
-   * automatically by VAD the response will be added to the default conversation
-   */
-  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 object type, must be `realtime.response`.
-   */
-  object?: 'realtime.response';
-
-  /**
-   * The list of output items generated by the response.
-   */
-  output?: Array<ConversationItem>;
-
-  /**
-   * The set of modalities the model used to respond, currently the only possible
-   * values are `[\"audio\"]`, `[\"text\"]`. Audio output always include a text
-   * transcript. Setting the output to mode `text` will disable audio output from the
-   * model.
-   */
-  output_modalities?: Array<'text' | 'audio'>;
-
-  /**
-   * 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;
-
-  /**
-   * 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;
-}
-
-export namespace RealtimeResponse {
-  /**
-   * Configuration for audio output.
-   */
-  export interface Audio {
-    output?: Audio.Output;
-  }
-
-  export namespace Audio {
-    export interface Output {
-      /**
-       * The format of the output audio.
-       */
-      format?: RealtimeAPI.RealtimeAudioFormats;
-
-      /**
-       * 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';
-    }
-  }
-}
-
-/**
- * Configuration for audio input and output.
- */
-export interface RealtimeResponseCreateAudioOutput {
-  output?: RealtimeResponseCreateAudioOutput.Output;
-}
-
-export namespace RealtimeResponseCreateAudioOutput {
-  export interface Output {
-    /**
-     * The format of the output audio.
-     */
-    format?: RealtimeAPI.RealtimeAudioFormats;
-
-    /**
-     * 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 RealtimeResponseCreateMcpTool {
-  /**
-   * 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> | RealtimeResponseCreateMcpTool.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?: RealtimeResponseCreateMcpTool.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 RealtimeResponseCreateMcpTool {
-  /**
-   * 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>;
-    }
-  }
-}
-
-/**
- * Create a new Realtime response with these parameters
- */
-export interface RealtimeResponseCreateParams {
-  /**
-   * Configuration for audio input and output.
-   */
-  audio?: RealtimeResponseCreateAudioOutput;
-
-  /**
-   * 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 that previously appeared in the session using their
-   * id.
-   */
-  input?: Array<ConversationItem>;
-
-  /**
-   * 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';
-
-  /**
-   * 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, currently the only possible
-   * values are `[\"audio\"]`, `[\"text\"]`. Audio output always include a text
-   * transcript. Setting the output to mode `text` will disable audio output from the
-   * model.
-   */
-  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<RealtimeFunctionTool | RealtimeResponseCreateMcpTool>;
-}
-
-/**
- * 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. Cached tokens are tokens
-   * from previous turns in the conversation that are included as context for the
-   * current response. Cached tokens here are counted as a subset of input tokens,
-   * meaning input tokens will include cached and uncached tokens.
-   */
-  input_token_details?: RealtimeResponseUsageInputTokenDetails;
-
-  /**
-   * 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?: RealtimeResponseUsageOutputTokenDetails;
-
-  /**
-   * 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;
-}
-
-/**
- * Details about the input tokens used in the Response. Cached tokens are tokens
- * from previous turns in the conversation that are included as context for the
- * current response. Cached tokens here are counted as a subset of input tokens,
- * meaning input tokens will include cached and uncached tokens.
- */
-export interface RealtimeResponseUsageInputTokenDetails {
-  /**
-   * The number of audio tokens used as input for the Response.
-   */
-  audio_tokens?: number;
-
-  /**
-   * The number of cached tokens used as input for the Response.
-   */
-  cached_tokens?: number;
-
-  /**
-   * Details about the cached tokens used as input for the Response.
-   */
-  cached_tokens_details?: RealtimeResponseUsageInputTokenDetails.CachedTokensDetails;
-
-  /**
-   * The number of image tokens used as input for the Response.
-   */
-  image_tokens?: number;
-
-  /**
-   * The number of text tokens used as input for the Response.
-   */
-  text_tokens?: number;
-}
-
-export namespace RealtimeResponseUsageInputTokenDetails {
-  /**
-   * Details about the cached tokens used as input for the Response.
-   */
-  export interface CachedTokensDetails {
-    /**
-     * The number of cached audio tokens used as input for the Response.
-     */
-    audio_tokens?: number;
-
-    /**
-     * The number of cached image tokens used as input for the Response.
-     */
-    image_tokens?: number;
-
-    /**
-     * The number of cached text tokens used as input for the Response.
-     */
-    text_tokens?: number;
-  }
-}
-
-/**
- * Details about the output tokens used in the Response.
- */
-export interface RealtimeResponseUsageOutputTokenDetails {
-  /**
-   * 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
-  | RealtimeErrorEvent
-  | InputAudioBufferClearedEvent
-  | InputAudioBufferCommittedEvent
-  | InputAudioBufferDtmfEventReceivedEvent
-  | InputAudioBufferSpeechStartedEvent
-  | InputAudioBufferSpeechStoppedEvent
-  | RateLimitsUpdatedEvent
-  | ResponseAudioDeltaEvent
-  | ResponseAudioDoneEvent
-  | ResponseAudioTranscriptDeltaEvent
-  | ResponseAudioTranscriptDoneEvent
-  | ResponseContentPartAddedEvent
-  | ResponseContentPartDoneEvent
-  | ResponseCreatedEvent
-  | ResponseDoneEvent
-  | ResponseFunctionCallArgumentsDeltaEvent
-  | ResponseFunctionCallArgumentsDoneEvent
-  | ResponseOutputItemAddedEvent
-  | ResponseOutputItemDoneEvent
-  | ResponseTextDeltaEvent
-  | ResponseTextDoneEvent
-  | SessionCreatedEvent
-  | SessionUpdatedEvent
-  | RealtimeServerEvent.OutputAudioBufferStarted
-  | RealtimeServerEvent.OutputAudioBufferStopped
-  | RealtimeServerEvent.OutputAudioBufferCleared
-  | ConversationItemAdded
-  | ConversationItemDone
-  | InputAudioBufferTimeoutTriggered
-  | ConversationItemInputAudioTranscriptionSegment
-  | McpListToolsInProgress
-  | McpListToolsCompleted
-  | McpListToolsFailed
-  | ResponseMcpCallArgumentsDelta
-  | ResponseMcpCallArgumentsDone
-  | ResponseMcpCallInProgress
-  | ResponseMcpCallCompleted
-  | ResponseMcpCallFailed;
-
-export namespace RealtimeServerEvent {
-  /**
-   * Returned when a conversation item is retrieved with
-   * `conversation.item.retrieve`. This is provided as a way to fetch the server's
-   * representation of an item, for example to get access to the post-processed audio
-   * data after noise cancellation and VAD. It includes the full content of the Item,
-   * including audio data.
-   */
-  export interface ConversationItemRetrieved {
-    /**
-     * The unique ID of the server event.
-     */
-    event_id: string;
-
-    /**
-     * A single item within a Realtime conversation.
-     */
-    item: RealtimeAPI.ConversationItem;
-
-    /**
-     * The event type, must be `conversation.item.retrieved`.
-     */
-    type: 'conversation.item.retrieved';
-  }
-
-  /**
-   * **WebRTC/SIP 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/SIP 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/SIP 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';
-  }
-}
-
-/**
- * Realtime session object for the beta interface.
- */
-export interface RealtimeSession {
-  /**
-   * Unique identifier for the session that looks like `sess_1234567890abcdef`.
-   */
-  id?: string;
-
-  /**
-   * 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'> | null;
-
-  /**
-   * 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?: RealtimeSession.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?: AudioTranscription | null;
-
-  /**
-   * 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-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-audio-mini'
-    | 'gpt-audio-mini-2025-10-06';
-
-  /**
-   * The object type. Always `realtime.session`.
-   */
-  object?: 'realtime.session';
-
-  /**
-   * 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';
-
-  /**
-   * 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;
-
-  /**
-   * 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<RealtimeFunctionTool>;
-
-  /**
-   * 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' | RealtimeSession.TracingConfiguration | null;
-
-  /**
-   * 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?: RealtimeSession.ServerVad | RealtimeSession.SemanticVad | null;
-
-  /**
-   * 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'
-    | 'marin'
-    | 'cedar';
-}
-
-export namespace RealtimeSession {
-  /**
-   * 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?: RealtimeAPI.NoiseReductionType;
-  }
-
-  /**
-   * 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;
-  }
-
-  /**
-   * 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;
-  }
-}
-
-/**
- * Realtime session object configuration.
- */
-export interface RealtimeSessionCreateRequest {
-  /**
-   * The type of session to create. Always `realtime` for the Realtime API.
-   */
-  type: 'realtime';
-
-  /**
-   * Configuration for input and output audio.
-   */
-  audio?: RealtimeAudioConfig;
-
-  /**
-   * 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-audio-mini'
-    | 'gpt-audio-mini-2025-10-06';
-
-  /**
-   * 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?: RealtimeToolChoiceConfig;
-
-  /**
-   * Tools available to the model.
-   */
-  tools?: RealtimeToolsConfig;
-
-  /**
-   * 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?: RealtimeTracingConfig | 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?: RealtimeTruncation;
-}
-
-/**
- * How the model chooses tools. Provide one of the string modes or force a specific
- * function/MCP tool.
- */
-export type RealtimeToolChoiceConfig =
-  | ResponsesAPI.ToolChoiceOptions
-  | ResponsesAPI.ToolChoiceFunction
-  | ResponsesAPI.ToolChoiceMcp;
-
-/**
- * Tools available to the model.
- */
-export type RealtimeToolsConfig = Array<RealtimeToolsConfigUnion>;
-
-/**
- * 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 type RealtimeToolsConfigUnion = RealtimeFunctionTool | RealtimeToolsConfigUnion.Mcp;
-
-export namespace RealtimeToolsConfigUnion {
-  /**
-   * 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 Mcp {
-    /**
-     * 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> | Mcp.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?: Mcp.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 Mcp {
-    /**
-     * 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>;
-      }
-    }
-  }
-}
-
-/**
- * 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.
- */
-export type RealtimeTracingConfig = 'auto' | RealtimeTracingConfig.TracingConfiguration;
-
-export namespace RealtimeTracingConfig {
-  /**
-   * 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 input and output audio.
- */
-export interface RealtimeTranscriptionSessionAudio {
-  input?: RealtimeTranscriptionSessionAudioInput;
-}
-
-export interface RealtimeTranscriptionSessionAudioInput {
-  /**
-   * The PCM audio format. Only a 24kHz sample rate is supported.
-   */
-  format?: 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?: RealtimeTranscriptionSessionAudioInput.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?: 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?: RealtimeTranscriptionSessionAudioInputTurnDetection | null;
-}
-
-export namespace RealtimeTranscriptionSessionAudioInput {
-  /**
-   * 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;
-  }
-}
-
-/**
- * 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 type RealtimeTranscriptionSessionAudioInputTurnDetection =
-  | RealtimeTranscriptionSessionAudioInputTurnDetection.ServerVad
-  | RealtimeTranscriptionSessionAudioInputTurnDetection.SemanticVad;
-
-export namespace RealtimeTranscriptionSessionAudioInputTurnDetection {
-  /**
-   * 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;
-  }
-}
-
-/**
- * Realtime transcription session object configuration.
- */
-export interface RealtimeTranscriptionSessionCreateRequest {
-  /**
-   * The type of session to create. Always `transcription` for transcription
-   * sessions.
-   */
-  type: 'transcription';
-
-  /**
-   * Configuration for input and output audio.
-   */
-  audio?: RealtimeTranscriptionSessionAudio;
-
-  /**
-   * 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'>;
-}
-
-/**
- * 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.
- */
-export type RealtimeTruncation = 'auto' | 'disabled' | RealtimeTruncationRetentionRatio;
-
-/**
- * Retain a fraction of the conversation tokens when the conversation exceeds the
- * input token limit. This allows you to amortize truncations across multiple
- * turns, which can help improve cached token usage.
- */
-export interface RealtimeTruncationRetentionRatio {
-  /**
-   * Fraction of post-instruction conversation tokens to retain (`0.0` - `1.0`) when
-   * the conversation exceeds the input token limit. Setting this to `0.8` means that
-   * messages will be dropped until 80% of the maximum allowed tokens are used. This
-   * helps reduce the frequency of truncations and improve cache rates.
-   */
-  retention_ratio: number;
-
-  /**
-   * Use retention ratio truncation.
-   */
-  type: 'retention_ratio';
-
-  /**
-   * Optional custom token limits for this truncation strategy. If not provided, the
-   * model's default token limits will be used.
-   */
-  token_limits?: RealtimeTruncationRetentionRatio.TokenLimits;
-}
-
-export namespace RealtimeTruncationRetentionRatio {
-  /**
-   * Optional custom token limits for this truncation strategy. If not provided, the
-   * model's default token limits will be used.
-   */
-  export interface TokenLimits {
-    /**
-     * Maximum tokens allowed in the conversation after instructions (which including
-     * tool definitions). For example, setting this to 5,000 would mean that truncation
-     * would occur when the conversation exceeds 5,000 tokens after instructions. This
-     * cannot be higher than the model's context window size minus the maximum output
-     * tokens.
-     */
-    post_instructions?: number;
-  }
-}
-
-/**
- * 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.output_audio.delta`.
-   */
-  type: 'response.output_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.output_audio.done`.
-   */
-  type: 'response.output_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.output_audio_transcript.delta`.
-   */
-  type: 'response.output_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.output_audio_transcript.done`.
-   */
-  type: 'response.output_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. It's safe to
- * call `response.cancel` even if no response is in progress, an error will be
- * returned the session will remain unaffected.
- */
-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 by default.
- *
- * 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 `tools`. If these are set, they will override the Session's configuration
- * for this Response only.
- *
- * Responses can be created out-of-band of the default Conversation, meaning that
- * they can have arbitrary input, and it's possible to disable writing the output
- * to the Conversation. Only one Response can write to the default Conversation at
- * a time, but otherwise multiple Responses can be created in parallel. The
- * `metadata` field is a good way to disambiguate multiple simultaneous Responses.
- *
- * Clients can set `conversation` to `none` to create a Response that does not
- * write to the default Conversation. Arbitrary input can be provided with the
- * `input` field, which is an array accepting raw Items and references to existing
- * Items.
- */
-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?: RealtimeResponseCreateParams;
-}
-
-/**
- * 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.
- *
- * Clients should check the `status` field of the Response to determine if it was
- * successful (`completed`) or if there was another outcome: `cancelled`, `failed`,
- * or `incomplete`.
- *
- * A response will contain all output items that were generated during the
- * response, excluding any audio content.
- */
-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 MCP tool call arguments are updated during response generation.
- */
-export interface ResponseMcpCallArgumentsDelta {
-  /**
-   * The JSON-encoded arguments delta.
-   */
-  delta: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP tool 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.mcp_call_arguments.delta`.
-   */
-  type: 'response.mcp_call_arguments.delta';
-
-  /**
-   * If present, indicates the delta text was obfuscated.
-   */
-  obfuscation?: string | null;
-}
-
-/**
- * Returned when MCP tool call arguments are finalized during response generation.
- */
-export interface ResponseMcpCallArgumentsDone {
-  /**
-   * The final JSON-encoded arguments string.
-   */
-  arguments: string;
-
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP tool 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.mcp_call_arguments.done`.
-   */
-  type: 'response.mcp_call_arguments.done';
-}
-
-/**
- * Returned when an MCP tool call has completed successfully.
- */
-export interface ResponseMcpCallCompleted {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP tool call item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The event type, must be `response.mcp_call.completed`.
-   */
-  type: 'response.mcp_call.completed';
-}
-
-/**
- * Returned when an MCP tool call has failed.
- */
-export interface ResponseMcpCallFailed {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP tool call item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The event type, must be `response.mcp_call.failed`.
-   */
-  type: 'response.mcp_call.failed';
-}
-
-/**
- * Returned when an MCP tool call has started and is in progress.
- */
-export interface ResponseMcpCallInProgress {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * The ID of the MCP tool call item.
-   */
-  item_id: string;
-
-  /**
-   * The index of the output item in the response.
-   */
-  output_index: number;
-
-  /**
-   * The event type, must be `response.mcp_call.in_progress`.
-   */
-  type: 'response.mcp_call.in_progress';
-}
-
-/**
- * Returned when a new Item is created during Response generation.
- */
-export interface ResponseOutputItemAddedEvent {
-  /**
-   * The unique ID of the server event.
-   */
-  event_id: string;
-
-  /**
-   * A single item within a Realtime 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;
-
-  /**
-   * A single item within a Realtime 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 an "output_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.output_text.delta`.
-   */
-  type: 'response.output_text.delta';
-}
-
-/**
- * Returned when the text value of an "output_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.output_text.done`.
-   */
-  type: 'response.output_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;
-
-  /**
-   * The session configuration.
-   */
-  session: RealtimeSessionCreateRequest | RealtimeTranscriptionSessionCreateRequest;
-
-  /**
-   * The event type, must be `session.created`.
-   */
-  type: 'session.created';
-}
-
-/**
- * Send this event to update the session’s configuration. The client may send this
- * event at any time to update any field except for `voice` and `model`. `voice`
- * can be updated only if there have been no other audio outputs yet.
- *
- * 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 in the `session.update` are updated. To clear a field
- * like `instructions`, pass an empty string. To clear a field like `tools`, pass
- * an empty array. To clear a field like `turn_detection`, pass `null`.
- */
-export interface SessionUpdateEvent {
-  /**
-   * Update the Realtime session. Choose either a realtime session or a transcription
-   * session.
-   */
-  session: RealtimeSessionCreateRequest | RealtimeTranscriptionSessionCreateRequest;
-
-  /**
-   * The event type, must be `session.update`.
-   */
-  type: 'session.update';
-
-  /**
-   * Optional client-generated ID used to identify this event. This is an arbitrary
-   * string that a client may assign. It will be passed back if there is an error
-   * with the event, but the corresponding `session.updated` event will not include
-   * it.
-   */
-  event_id?: string;
-}
-
-/**
- * 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;
-
-  /**
-   * The session configuration.
-   */
-  session: RealtimeSessionCreateRequest | RealtimeTranscriptionSessionCreateRequest;
-
-  /**
-   * 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 {
-    /**
-     * The set of items to include in the transcription. Current available items are:
-     * `item.input_audio_transcription.logprobs`
-     */
-    include?: Array<'item.input_audio_transcription.logprobs'>;
-
-    /**
-     * 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?: 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?: Session.TurnDetection;
-  }
-
-  export namespace Session {
-    /**
-     * Configuration for input audio noise reduction. This can be set to `null` to turn
-     * off. Noise reduction filters audio added to the input audio buffer before it is
-     * sent to VAD and the model. Filtering the audio can improve VAD and turn
-     * detection accuracy (reducing false positives) and model performance by improving
-     * perception of the input audio.
-     */
-    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?: 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 TurnDetection {
-      /**
-       * Amount of audio to include before the VAD detected speech (in milliseconds).
-       * Defaults to 300ms.
-       */
-      prefix_padding_ms?: number;
-
-      /**
-       * Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms.
-       * With shorter values the model will respond more quickly, but may jump in on
-       * short pauses from the user.
-       */
-      silence_duration_ms?: number;
-
-      /**
-       * Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher
-       * threshold will require louder audio to activate the model, and thus might
-       * perform better in noisy environments.
-       */
-      threshold?: number;
-
-      /**
-       * Type of turn detection. Only `server_vad` is currently supported for
-       * transcription sessions.
-       */
-      type?: 'server_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: TranscriptionSessionUpdatedEvent.Session;
-
-  /**
-   * The event type, must be `transcription_session.updated`.
-   */
-  type: 'transcription_session.updated';
-}
-
-export namespace TranscriptionSessionUpdatedEvent {
-  /**
-   * A new Realtime transcription session configuration.
-   *
-   * When a session is created on the server via REST API, the session object also
-   * contains an ephemeral key. Default TTL for keys is 10 minutes. This property is
-   * not present when a session is updated via the WebSocket API.
-   */
-  export interface Session {
-    /**
-     * Ephemeral key returned by the API. Only present when the session is created on
-     * the server via REST API.
-     */
-    client_secret: Session.ClientSecret;
-
-    /**
-     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
-     */
-    input_audio_format?: string;
-
-    /**
-     * Configuration of the transcription model.
-     */
-    input_audio_transcription?: RealtimeAPI.AudioTranscription;
-
-    /**
-     * The set of modalities the model can respond with. To disable audio, set this to
-     * ["text"].
-     */
-    modalities?: Array<'text' | 'audio'>;
-
-    /**
-     * Configuration for turn detection. Can be set to `null` to turn off. Server VAD
-     * means that the model will detect the start and end of speech based on audio
-     * volume and respond at the end of user speech.
-     */
-    turn_detection?: Session.TurnDetection;
-  }
-
-  export namespace Session {
-    /**
-     * Ephemeral key returned by the API. Only present when the session is created on
-     * the server via REST API.
-     */
-    export interface ClientSecret {
-      /**
-       * Timestamp for when the token expires. Currently, all tokens expire after one
-       * minute.
-       */
-      expires_at: number;
-
-      /**
-       * Ephemeral key usable in client environments to authenticate connections to the
-       * Realtime API. Use this in client-side environments rather than a standard API
-       * token, which should only be used server-side.
-       */
-      value: string;
-    }
-
-    /**
-     * Configuration for 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 TurnDetection {
-      /**
-       * Amount of audio to include before the VAD detected speech (in milliseconds).
-       * Defaults to 300ms.
-       */
-      prefix_padding_ms?: number;
-
-      /**
-       * Duration of silence to detect speech stop (in milliseconds). Defaults to 500ms.
-       * With shorter values the model will respond more quickly, but may jump in on
-       * short pauses from the user.
-       */
-      silence_duration_ms?: number;
-
-      /**
-       * Activation threshold for VAD (0.0 to 1.0), this defaults to 0.5. A higher
-       * threshold will require louder audio to activate the model, and thus might
-       * perform better in noisy environments.
-       */
-      threshold?: number;
-
-      /**
-       * Type of turn detection, only `server_vad` is currently supported.
-       */
-      type?: string;
-    }
-  }
-}
-
-Realtime.ClientSecrets = ClientSecrets;
-Realtime.Calls = Calls;
-
-export declare namespace Realtime {
-  export {
-    type AudioTranscription as AudioTranscription,
-    type ConversationCreatedEvent as ConversationCreatedEvent,
-    type ConversationItem as ConversationItem,
-    type ConversationItemAdded as ConversationItemAdded,
-    type ConversationItemCreateEvent as ConversationItemCreateEvent,
-    type ConversationItemCreatedEvent as ConversationItemCreatedEvent,
-    type ConversationItemDeleteEvent as ConversationItemDeleteEvent,
-    type ConversationItemDeletedEvent as ConversationItemDeletedEvent,
-    type ConversationItemDone as ConversationItemDone,
-    type ConversationItemInputAudioTranscriptionCompletedEvent as ConversationItemInputAudioTranscriptionCompletedEvent,
-    type ConversationItemInputAudioTranscriptionDeltaEvent as ConversationItemInputAudioTranscriptionDeltaEvent,
-    type ConversationItemInputAudioTranscriptionFailedEvent as ConversationItemInputAudioTranscriptionFailedEvent,
-    type ConversationItemInputAudioTranscriptionSegment as ConversationItemInputAudioTranscriptionSegment,
-    type ConversationItemRetrieveEvent as ConversationItemRetrieveEvent,
-    type ConversationItemTruncateEvent as ConversationItemTruncateEvent,
-    type ConversationItemTruncatedEvent as ConversationItemTruncatedEvent,
-    type ConversationItemWithReference as ConversationItemWithReference,
-    type InputAudioBufferAppendEvent as InputAudioBufferAppendEvent,
-    type InputAudioBufferClearEvent as InputAudioBufferClearEvent,
-    type InputAudioBufferClearedEvent as InputAudioBufferClearedEvent,
-    type InputAudioBufferCommitEvent as InputAudioBufferCommitEvent,
-    type InputAudioBufferCommittedEvent as InputAudioBufferCommittedEvent,
-    type InputAudioBufferDtmfEventReceivedEvent as InputAudioBufferDtmfEventReceivedEvent,
-    type InputAudioBufferSpeechStartedEvent as InputAudioBufferSpeechStartedEvent,
-    type InputAudioBufferSpeechStoppedEvent as InputAudioBufferSpeechStoppedEvent,
-    type InputAudioBufferTimeoutTriggered as InputAudioBufferTimeoutTriggered,
-    type LogProbProperties as LogProbProperties,
-    type McpListToolsCompleted as McpListToolsCompleted,
-    type McpListToolsFailed as McpListToolsFailed,
-    type McpListToolsInProgress as McpListToolsInProgress,
-    type NoiseReductionType as NoiseReductionType,
-    type OutputAudioBufferClearEvent as OutputAudioBufferClearEvent,
-    type RateLimitsUpdatedEvent as RateLimitsUpdatedEvent,
-    type RealtimeAudioConfig as RealtimeAudioConfig,
-    type RealtimeAudioConfigInput as RealtimeAudioConfigInput,
-    type RealtimeAudioConfigOutput as RealtimeAudioConfigOutput,
-    type RealtimeAudioFormats as RealtimeAudioFormats,
-    type RealtimeAudioInputTurnDetection as RealtimeAudioInputTurnDetection,
-    type RealtimeClientEvent as RealtimeClientEvent,
-    type RealtimeConversationItemAssistantMessage as RealtimeConversationItemAssistantMessage,
-    type RealtimeConversationItemFunctionCall as RealtimeConversationItemFunctionCall,
-    type RealtimeConversationItemFunctionCallOutput as RealtimeConversationItemFunctionCallOutput,
-    type RealtimeConversationItemSystemMessage as RealtimeConversationItemSystemMessage,
-    type RealtimeConversationItemUserMessage as RealtimeConversationItemUserMessage,
-    type RealtimeError as RealtimeError,
-    type RealtimeErrorEvent as RealtimeErrorEvent,
-    type RealtimeFunctionTool as RealtimeFunctionTool,
-    type RealtimeMcpApprovalRequest as RealtimeMcpApprovalRequest,
-    type RealtimeMcpApprovalResponse as RealtimeMcpApprovalResponse,
-    type RealtimeMcpListTools as RealtimeMcpListTools,
-    type RealtimeMcpProtocolError as RealtimeMcpProtocolError,
-    type RealtimeMcpToolCall as RealtimeMcpToolCall,
-    type RealtimeMcpToolExecutionError as RealtimeMcpToolExecutionError,
-    type RealtimeMcphttpError as RealtimeMcphttpError,
-    type RealtimeResponse as RealtimeResponse,
-    type RealtimeResponseCreateAudioOutput as RealtimeResponseCreateAudioOutput,
-    type RealtimeResponseCreateMcpTool as RealtimeResponseCreateMcpTool,
-    type RealtimeResponseCreateParams as RealtimeResponseCreateParams,
-    type RealtimeResponseStatus as RealtimeResponseStatus,
-    type RealtimeResponseUsage as RealtimeResponseUsage,
-    type RealtimeResponseUsageInputTokenDetails as RealtimeResponseUsageInputTokenDetails,
-    type RealtimeResponseUsageOutputTokenDetails as RealtimeResponseUsageOutputTokenDetails,
-    type RealtimeServerEvent as RealtimeServerEvent,
-    type RealtimeSession as RealtimeSession,
-    type RealtimeSessionCreateRequest as RealtimeSessionCreateRequest,
-    type RealtimeToolChoiceConfig as RealtimeToolChoiceConfig,
-    type RealtimeToolsConfig as RealtimeToolsConfig,
-    type RealtimeToolsConfigUnion as RealtimeToolsConfigUnion,
-    type RealtimeTracingConfig as RealtimeTracingConfig,
-    type RealtimeTranscriptionSessionAudio as RealtimeTranscriptionSessionAudio,
-    type RealtimeTranscriptionSessionAudioInput as RealtimeTranscriptionSessionAudioInput,
-    type RealtimeTranscriptionSessionAudioInputTurnDetection as RealtimeTranscriptionSessionAudioInputTurnDetection,
-    type RealtimeTranscriptionSessionCreateRequest as RealtimeTranscriptionSessionCreateRequest,
-    type RealtimeTruncation as RealtimeTruncation,
-    type RealtimeTruncationRetentionRatio as RealtimeTruncationRetentionRatio,
-    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 ResponseMcpCallArgumentsDelta as ResponseMcpCallArgumentsDelta,
-    type ResponseMcpCallArgumentsDone as ResponseMcpCallArgumentsDone,
-    type ResponseMcpCallCompleted as ResponseMcpCallCompleted,
-    type ResponseMcpCallFailed as ResponseMcpCallFailed,
-    type ResponseMcpCallInProgress as ResponseMcpCallInProgress,
-    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 {
-    ClientSecrets as ClientSecrets,
-    type RealtimeSessionClientSecret as RealtimeSessionClientSecret,
-    type RealtimeSessionCreateResponse as RealtimeSessionCreateResponse,
-    type RealtimeTranscriptionSessionCreateResponse as RealtimeTranscriptionSessionCreateResponse,
-    type RealtimeTranscriptionSessionTurnDetection as RealtimeTranscriptionSessionTurnDetection,
-    type ClientSecretCreateResponse as ClientSecretCreateResponse,
-    type ClientSecretCreateParams as ClientSecretCreateParams,
-  };
-
-  export {
-    Calls as Calls,
-    type CallAcceptParams as CallAcceptParams,
-    type CallReferParams as CallReferParams,
-    type CallRejectParams as CallRejectParams,
-  };
-}