@tiendung-36/openrouter-cli 1.0.0

package/node_modules/openai/src/resources/beta/realtime/realtime.ts

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+// File generated from our OpenAPI spec by Stainless. See CONTRIBUTING.md for details.
+
+import { APIResource } from '../../../resource';
+import * as RealtimeAPI from './realtime';
+import * as Shared from '../../shared';
+import * as SessionsAPI from './sessions';
+import {
+  Session as SessionsAPISession,
+  SessionCreateParams,
+  SessionCreateResponse,
+  Sessions,
+} from './sessions';
+import * as TranscriptionSessionsAPI from './transcription-sessions';
+import {
+  TranscriptionSession,
+  TranscriptionSessionCreateParams,
+  TranscriptionSessions,
+} from './transcription-sessions';
+
+export class Realtime extends APIResource {
+  sessions: SessionsAPI.Sessions = new SessionsAPI.Sessions(this._client);
+  transcriptionSessions: TranscriptionSessionsAPI.TranscriptionSessions =
+    new TranscriptionSessionsAPI.TranscriptionSessions(this._client);
+}
+
+/**
+ * Returned when a conversation is created. Emitted right after session creation.
+ */
+export interface ConversationCreatedEvent {
+  /**
+   * The conversation resource.
+   */
+  conversation: ConversationCreatedEvent.Conversation;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The event type, must be `conversation.created`.
+   */
+  type: 'conversation.created';
+}
+
+export namespace ConversationCreatedEvent {
+  /**
+   * The conversation resource.
+   */
+  export interface Conversation {
+    /**
+     * The unique ID of the conversation.
+     */
+    id?: string;
+
+    /**
+     * The object type, must be `realtime.conversation`.
+     */
+    object?: 'realtime.conversation';
+  }
+}
+
+/**
+ * The item to add to the conversation.
+ */
+export interface ConversationItem {
+  /**
+   * The unique ID of the item, this can be generated by the client to help manage
+   * server-side context, but is not required because the server will generate one if
+   * not provided.
+   */
+  id?: string;
+
+  /**
+   * The arguments of the function call (for `function_call` items).
+   */
+  arguments?: string;
+
+  /**
+   * The ID of the function call (for `function_call` and `function_call_output`
+   * items). If passed on a `function_call_output` item, the server will check that a
+   * `function_call` item with the same ID exists in the conversation history.
+   */
+  call_id?: string;
+
+  /**
+   * The content of the message, applicable for `message` items.
+   *
+   * - Message items of role `system` support only `input_text` content
+   * - Message items of role `user` support `input_text` and `input_audio` content
+   * - Message items of role `assistant` support `text` content.
+   */
+  content?: Array<ConversationItemContent>;
+
+  /**
+   * The name of the function being called (for `function_call` items).
+   */
+  name?: string;
+
+  /**
+   * Identifier for the API object being returned - always `realtime.item`.
+   */
+  object?: 'realtime.item';
+
+  /**
+   * The output of the function call (for `function_call_output` items).
+   */
+  output?: string;
+
+  /**
+   * The role of the message sender (`user`, `assistant`, `system`), only applicable
+   * for `message` items.
+   */
+  role?: 'user' | 'assistant' | 'system';
+
+  /**
+   * The status of the item (`completed`, `incomplete`). These have no effect on the
+   * conversation, but are accepted for consistency with the
+   * `conversation.item.created` event.
+   */
+  status?: 'completed' | 'incomplete';
+
+  /**
+   * The type of the item (`message`, `function_call`, `function_call_output`).
+   */
+  type?: 'message' | 'function_call' | 'function_call_output';
+}
+
+export interface ConversationItemContent {
+  /**
+   * ID of a previous conversation item to reference (for `item_reference` content
+   * types in `response.create` events). These can reference both client and server
+   * created items.
+   */
+  id?: string;
+
+  /**
+   * Base64-encoded audio bytes, used for `input_audio` content type.
+   */
+  audio?: string;
+
+  /**
+   * The text content, used for `input_text` and `text` content types.
+   */
+  text?: string;
+
+  /**
+   * The transcript of the audio, used for `input_audio` content type.
+   */
+  transcript?: string;
+
+  /**
+   * The content type (`input_text`, `input_audio`, `item_reference`, `text`).
+   */
+  type?: 'input_text' | 'input_audio' | 'item_reference' | 'text';
+}
+
+/**
+ * Add a new Item to the Conversation's context, including messages, function
+ * calls, and function call responses. This event can be used both to populate a
+ * "history" of the conversation and to add new items mid-stream, but has the
+ * current limitation that it cannot populate assistant audio messages.
+ *
+ * If successful, the server will respond with a `conversation.item.created` event,
+ * otherwise an `error` event will be sent.
+ */
+export interface ConversationItemCreateEvent {
+  /**
+   * The item to add to the conversation.
+   */
+  item: ConversationItem;
+
+  /**
+   * The event type, must be `conversation.item.create`.
+   */
+  type: 'conversation.item.create';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+
+  /**
+   * The ID of the preceding item after which the new item will be inserted. If not
+   * set, the new item will be appended to the end of the conversation. If set to
+   * `root`, the new item will be added to the beginning of the conversation. If set
+   * to an existing ID, it allows an item to be inserted mid-conversation. If the ID
+   * cannot be found, an error will be returned and the item will not be added.
+   */
+  previous_item_id?: string;
+}
+
+/**
+ * Returned when a conversation item is created. There are several scenarios that
+ * produce this event:
+ *
+ * - The server is generating a Response, which if successful will produce either
+ *   one or two Items, which will be of type `message` (role `assistant`) or type
+ *   `function_call`.
+ * - The input audio buffer has been committed, either by the client or the server
+ *   (in `server_vad` mode). The server will take the content of the input audio
+ *   buffer and add it to a new user message Item.
+ * - The client has sent a `conversation.item.create` event to add a new Item to
+ *   the Conversation.
+ */
+export interface ConversationItemCreatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The item to add to the conversation.
+   */
+  item: ConversationItem;
+
+  /**
+   * The ID of the preceding item in the Conversation context, allows the client to
+   * understand the order of the conversation.
+   */
+  previous_item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.created`.
+   */
+  type: 'conversation.item.created';
+}
+
+/**
+ * Send this event when you want to remove any item from the conversation history.
+ * The server will respond with a `conversation.item.deleted` event, unless the
+ * item does not exist in the conversation history, in which case the server will
+ * respond with an error.
+ */
+export interface ConversationItemDeleteEvent {
+  /**
+   * The ID of the item to delete.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.delete`.
+   */
+  type: 'conversation.item.delete';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Returned when an item in the conversation is deleted by the client with a
+ * `conversation.item.delete` event. This event is used to synchronize the server's
+ * understanding of the conversation history with the client's view.
+ */
+export interface ConversationItemDeletedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item that was deleted.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.deleted`.
+   */
+  type: 'conversation.item.deleted';
+}
+
+/**
+ * This event is the output of audio transcription for user audio written to the
+ * user audio buffer. Transcription begins when the input audio buffer is committed
+ * by the client or server (in `server_vad` mode). Transcription runs
+ * asynchronously with Response creation, so this event may come before or after
+ * the Response events.
+ *
+ * Realtime API models accept audio natively, and thus input transcription is a
+ * separate process run on a separate ASR (Automatic Speech Recognition) model,
+ * currently always `whisper-1`. Thus the transcript may diverge somewhat from the
+ * model's interpretation, and should be treated as a rough guide.
+ */
+export interface ConversationItemInputAudioTranscriptionCompletedEvent {
+  /**
+   * The index of the content part containing the audio.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the user message item containing the audio.
+   */
+  item_id: string;
+
+  /**
+   * The transcribed text.
+   */
+  transcript: string;
+
+  /**
+   * The event type, must be `conversation.item.input_audio_transcription.completed`.
+   */
+  type: 'conversation.item.input_audio_transcription.completed';
+
+  /**
+   * The log probabilities of the transcription.
+   */
+  logprobs?: Array<ConversationItemInputAudioTranscriptionCompletedEvent.Logprob> | null;
+}
+
+export namespace ConversationItemInputAudioTranscriptionCompletedEvent {
+  /**
+   * A log probability object.
+   */
+  export interface Logprob {
+    /**
+     * The token that was used to generate the log probability.
+     */
+    token: string;
+
+    /**
+     * The bytes that were used to generate the log probability.
+     */
+    bytes: Array<number>;
+
+    /**
+     * The log probability of the token.
+     */
+    logprob: number;
+  }
+}
+
+/**
+ * Returned when the text value of an input audio transcription content part is
+ * updated.
+ */
+export interface ConversationItemInputAudioTranscriptionDeltaEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.input_audio_transcription.delta`.
+   */
+  type: 'conversation.item.input_audio_transcription.delta';
+
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index?: number;
+
+  /**
+   * The text delta.
+   */
+  delta?: string;
+
+  /**
+   * The log probabilities of the transcription.
+   */
+  logprobs?: Array<ConversationItemInputAudioTranscriptionDeltaEvent.Logprob> | null;
+}
+
+export namespace ConversationItemInputAudioTranscriptionDeltaEvent {
+  /**
+   * A log probability object.
+   */
+  export interface Logprob {
+    /**
+     * The token that was used to generate the log probability.
+     */
+    token: string;
+
+    /**
+     * The bytes that were used to generate the log probability.
+     */
+    bytes: Array<number>;
+
+    /**
+     * The log probability of the token.
+     */
+    logprob: number;
+  }
+}
+
+/**
+ * Returned when input audio transcription is configured, and a transcription
+ * request for a user message failed. These events are separate from other `error`
+ * events so that the client can identify the related Item.
+ */
+export interface ConversationItemInputAudioTranscriptionFailedEvent {
+  /**
+   * The index of the content part containing the audio.
+   */
+  content_index: number;
+
+  /**
+   * Details of the transcription error.
+   */
+  error: ConversationItemInputAudioTranscriptionFailedEvent.Error;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the user message item.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.input_audio_transcription.failed`.
+   */
+  type: 'conversation.item.input_audio_transcription.failed';
+}
+
+export namespace ConversationItemInputAudioTranscriptionFailedEvent {
+  /**
+   * Details of the transcription error.
+   */
+  export interface Error {
+    /**
+     * Error code, if any.
+     */
+    code?: string;
+
+    /**
+     * A human-readable error message.
+     */
+    message?: string;
+
+    /**
+     * Parameter related to the error, if any.
+     */
+    param?: string;
+
+    /**
+     * The type of error.
+     */
+    type?: string;
+  }
+}
+
+/**
+ * Send this event when you want to retrieve the server's representation of a
+ * specific item in the conversation history. This is useful, for example, to
+ * inspect user audio after noise cancellation and VAD. The server will respond
+ * with a `conversation.item.retrieved` event, unless the item does not exist in
+ * the conversation history, in which case the server will respond with an error.
+ */
+export interface ConversationItemRetrieveEvent {
+  /**
+   * The ID of the item to retrieve.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.retrieve`.
+   */
+  type: 'conversation.item.retrieve';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Send this event to truncate a previous assistant message’s audio. The server
+ * will produce audio faster than realtime, so this event is useful when the user
+ * interrupts to truncate audio that has already been sent to the client but not
+ * yet played. This will synchronize the server's understanding of the audio with
+ * the client's playback.
+ *
+ * Truncating audio will delete the server-side text transcript to ensure there is
+ * not text in the context that hasn't been heard by the user.
+ *
+ * If successful, the server will respond with a `conversation.item.truncated`
+ * event.
+ */
+export interface ConversationItemTruncateEvent {
+  /**
+   * Inclusive duration up to which audio is truncated, in milliseconds. If the
+   * audio_end_ms is greater than the actual audio duration, the server will respond
+   * with an error.
+   */
+  audio_end_ms: number;
+
+  /**
+   * The index of the content part to truncate. Set this to 0.
+   */
+  content_index: number;
+
+  /**
+   * The ID of the assistant message item to truncate. Only assistant message items
+   * can be truncated.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.truncate`.
+   */
+  type: 'conversation.item.truncate';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Returned when an earlier assistant audio message item is truncated by the client
+ * with a `conversation.item.truncate` event. This event is used to synchronize the
+ * server's understanding of the audio with the client's playback.
+ *
+ * This action will truncate the audio and remove the server-side text transcript
+ * to ensure there is no text in the context that hasn't been heard by the user.
+ */
+export interface ConversationItemTruncatedEvent {
+  /**
+   * The duration up to which the audio was truncated, in milliseconds.
+   */
+  audio_end_ms: number;
+
+  /**
+   * The index of the content part that was truncated.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the assistant message item that was truncated.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `conversation.item.truncated`.
+   */
+  type: 'conversation.item.truncated';
+}
+
+/**
+ * The item to add to the conversation.
+ */
+export interface ConversationItemWithReference {
+  /**
+   * For an item of type (`message` | `function_call` | `function_call_output`) this
+   * field allows the client to assign the unique ID of the item. It is not required
+   * because the server will generate one if not provided.
+   *
+   * For an item of type `item_reference`, this field is required and is a reference
+   * to any item that has previously existed in the conversation.
+   */
+  id?: string;
+
+  /**
+   * The arguments of the function call (for `function_call` items).
+   */
+  arguments?: string;
+
+  /**
+   * The ID of the function call (for `function_call` and `function_call_output`
+   * items). If passed on a `function_call_output` item, the server will check that a
+   * `function_call` item with the same ID exists in the conversation history.
+   */
+  call_id?: string;
+
+  /**
+   * The content of the message, applicable for `message` items.
+   *
+   * - Message items of role `system` support only `input_text` content
+   * - Message items of role `user` support `input_text` and `input_audio` content
+   * - Message items of role `assistant` support `text` content.
+   */
+  content?: Array<ConversationItemContent>;
+
+  /**
+   * The name of the function being called (for `function_call` items).
+   */
+  name?: string;
+
+  /**
+   * Identifier for the API object being returned - always `realtime.item`.
+   */
+  object?: 'realtime.item';
+
+  /**
+   * The output of the function call (for `function_call_output` items).
+   */
+  output?: string;
+
+  /**
+   * The role of the message sender (`user`, `assistant`, `system`), only applicable
+   * for `message` items.
+   */
+  role?: 'user' | 'assistant' | 'system';
+
+  /**
+   * The status of the item (`completed`, `incomplete`). These have no effect on the
+   * conversation, but are accepted for consistency with the
+   * `conversation.item.created` event.
+   */
+  status?: 'completed' | 'incomplete';
+
+  /**
+   * The type of the item (`message`, `function_call`, `function_call_output`,
+   * `item_reference`).
+   */
+  type?: 'message' | 'function_call' | 'function_call_output' | 'item_reference';
+}
+
+/**
+ * Returned when an error occurs, which could be a client problem or a server
+ * problem. Most errors are recoverable and the session will stay open, we
+ * recommend to implementors to monitor and log error messages by default.
+ */
+export interface ErrorEvent {
+  /**
+   * Details of the error.
+   */
+  error: ErrorEvent.Error;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The event type, must be `error`.
+   */
+  type: 'error';
+}
+
+export namespace ErrorEvent {
+  /**
+   * Details of the error.
+   */
+  export interface Error {
+    /**
+     * A human-readable error message.
+     */
+    message: string;
+
+    /**
+     * The type of error (e.g., "invalid_request_error", "server_error").
+     */
+    type: string;
+
+    /**
+     * Error code, if any.
+     */
+    code?: string | null;
+
+    /**
+     * The event_id of the client event that caused the error, if applicable.
+     */
+    event_id?: string | null;
+
+    /**
+     * Parameter related to the error, if any.
+     */
+    param?: string | null;
+  }
+}
+
+/**
+ * Send this event to append audio bytes to the input audio buffer. The audio
+ * buffer is temporary storage you can write to and later commit. In Server VAD
+ * mode, the audio buffer is used to detect speech and the server will decide when
+ * to commit. When Server VAD is disabled, you must commit the audio buffer
+ * manually.
+ *
+ * The client may choose how much audio to place in each event up to a maximum of
+ * 15 MiB, for example streaming smaller chunks from the client may allow the VAD
+ * to be more responsive. Unlike made other client events, the server will not send
+ * a confirmation response to this event.
+ */
+export interface InputAudioBufferAppendEvent {
+  /**
+   * Base64-encoded audio bytes. This must be in the format specified by the
+   * `input_audio_format` field in the session configuration.
+   */
+  audio: string;
+
+  /**
+   * The event type, must be `input_audio_buffer.append`.
+   */
+  type: 'input_audio_buffer.append';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Send this event to clear the audio bytes in the buffer. The server will respond
+ * with an `input_audio_buffer.cleared` event.
+ */
+export interface InputAudioBufferClearEvent {
+  /**
+   * The event type, must be `input_audio_buffer.clear`.
+   */
+  type: 'input_audio_buffer.clear';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Returned when the input audio buffer is cleared by the client with a
+ * `input_audio_buffer.clear` event.
+ */
+export interface InputAudioBufferClearedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The event type, must be `input_audio_buffer.cleared`.
+   */
+  type: 'input_audio_buffer.cleared';
+}
+
+/**
+ * Send this event to commit the user input audio buffer, which will create a new
+ * user message item in the conversation. This event will produce an error if the
+ * input audio buffer is empty. When in Server VAD mode, the client does not need
+ * to send this event, the server will commit the audio buffer automatically.
+ *
+ * Committing the input audio buffer will trigger input audio transcription (if
+ * enabled in session configuration), but it will not create a response from the
+ * model. The server will respond with an `input_audio_buffer.committed` event.
+ */
+export interface InputAudioBufferCommitEvent {
+  /**
+   * The event type, must be `input_audio_buffer.commit`.
+   */
+  type: 'input_audio_buffer.commit';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+/**
+ * Returned when an input audio buffer is committed, either by the client or
+ * automatically in server VAD mode. The `item_id` property is the ID of the user
+ * message item that will be created, thus a `conversation.item.created` event will
+ * also be sent to the client.
+ */
+export interface InputAudioBufferCommittedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the user message item that will be created.
+   */
+  item_id: string;
+
+  /**
+   * The ID of the preceding item after which the new item will be inserted.
+   */
+  previous_item_id: string;
+
+  /**
+   * The event type, must be `input_audio_buffer.committed`.
+   */
+  type: 'input_audio_buffer.committed';
+}
+
+/**
+ * Sent by the server when in `server_vad` mode to indicate that speech has been
+ * detected in the audio buffer. This can happen any time audio is added to the
+ * buffer (unless speech is already detected). The client may want to use this
+ * event to interrupt audio playback or provide visual feedback to the user.
+ *
+ * The client should expect to receive a `input_audio_buffer.speech_stopped` event
+ * when speech stops. The `item_id` property is the ID of the user message item
+ * that will be created when speech stops and will also be included in the
+ * `input_audio_buffer.speech_stopped` event (unless the client manually commits
+ * the audio buffer during VAD activation).
+ */
+export interface InputAudioBufferSpeechStartedEvent {
+  /**
+   * Milliseconds from the start of all audio written to the buffer during the
+   * session when speech was first detected. This will correspond to the beginning of
+   * audio sent to the model, and thus includes the `prefix_padding_ms` configured in
+   * the Session.
+   */
+  audio_start_ms: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the user message item that will be created when speech stops.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `input_audio_buffer.speech_started`.
+   */
+  type: 'input_audio_buffer.speech_started';
+}
+
+/**
+ * Returned in `server_vad` mode when the server detects the end of speech in the
+ * audio buffer. The server will also send an `conversation.item.created` event
+ * with the user message item that is created from the audio buffer.
+ */
+export interface InputAudioBufferSpeechStoppedEvent {
+  /**
+   * Milliseconds since the session started when speech stopped. This will correspond
+   * to the end of audio sent to the model, and thus includes the
+   * `min_silence_duration_ms` configured in the Session.
+   */
+  audio_end_ms: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the user message item that will be created.
+   */
+  item_id: string;
+
+  /**
+   * The event type, must be `input_audio_buffer.speech_stopped`.
+   */
+  type: 'input_audio_buffer.speech_stopped';
+}
+
+/**
+ * Emitted at the beginning of a Response to indicate the updated rate limits. When
+ * a Response is created some tokens will be "reserved" for the output tokens, the
+ * rate limits shown here reflect that reservation, which is then adjusted
+ * accordingly once the Response is completed.
+ */
+export interface RateLimitsUpdatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * List of rate limit information.
+   */
+  rate_limits: Array<RateLimitsUpdatedEvent.RateLimit>;
+
+  /**
+   * The event type, must be `rate_limits.updated`.
+   */
+  type: 'rate_limits.updated';
+}
+
+export namespace RateLimitsUpdatedEvent {
+  export interface RateLimit {
+    /**
+     * The maximum allowed value for the rate limit.
+     */
+    limit?: number;
+
+    /**
+     * The name of the rate limit (`requests`, `tokens`).
+     */
+    name?: 'requests' | 'tokens';
+
+    /**
+     * The remaining value before the limit is reached.
+     */
+    remaining?: number;
+
+    /**
+     * Seconds until the rate limit resets.
+     */
+    reset_seconds?: number;
+  }
+}
+
+/**
+ * A realtime client event.
+ */
+export type RealtimeClientEvent =
+  | ConversationItemCreateEvent
+  | ConversationItemDeleteEvent
+  | ConversationItemRetrieveEvent
+  | ConversationItemTruncateEvent
+  | InputAudioBufferAppendEvent
+  | InputAudioBufferClearEvent
+  | RealtimeClientEvent.OutputAudioBufferClear
+  | InputAudioBufferCommitEvent
+  | ResponseCancelEvent
+  | ResponseCreateEvent
+  | SessionUpdateEvent
+  | TranscriptionSessionUpdate;
+
+export namespace RealtimeClientEvent {
+  /**
+   * **WebRTC Only:** Emit to cut off the current audio response. This will trigger
+   * the server to stop generating audio and emit a `output_audio_buffer.cleared`
+   * event. This event should be preceded by a `response.cancel` client event to stop
+   * the generation of the current response.
+   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
+   */
+  export interface OutputAudioBufferClear {
+    /**
+     * The event type, must be `output_audio_buffer.clear`.
+     */
+    type: 'output_audio_buffer.clear';
+
+    /**
+     * The unique ID of the client event used for error handling.
+     */
+    event_id?: string;
+  }
+}
+
+/**
+ * The response resource.
+ */
+export interface RealtimeResponse {
+  /**
+   * The unique ID of the response.
+   */
+  id?: string;
+
+  /**
+   * Which conversation the response is added to, determined by the `conversation`
+   * field in the `response.create` event. If `auto`, the response will be added to
+   * the default conversation and the value of `conversation_id` will be an id like
+   * `conv_1234`. If `none`, the response will not be added to any conversation and
+   * the value of `conversation_id` will be `null`. If responses are being triggered
+   * by server VAD, the response will be added to the default conversation, thus the
+   * `conversation_id` will be an id like `conv_1234`.
+   */
+  conversation_id?: string;
+
+  /**
+   * Maximum number of output tokens for a single assistant response, inclusive of
+   * tool calls, that was used in this response.
+   */
+  max_output_tokens?: number | 'inf';
+
+  /**
+   * Set of 16 key-value pairs that can be attached to an object. This can be useful
+   * for storing additional information about the object in a structured format, and
+   * querying for objects via API or the dashboard.
+   *
+   * Keys are strings with a maximum length of 64 characters. Values are strings with
+   * a maximum length of 512 characters.
+   */
+  metadata?: Shared.Metadata | null;
+
+  /**
+   * The set of modalities the model used to respond. If there are multiple
+   * modalities, the model will pick one, for example if `modalities` is
+   * `["text", "audio"]`, the model could be responding in either text or audio.
+   */
+  modalities?: Array<'text' | 'audio'>;
+
+  /**
+   * The object type, must be `realtime.response`.
+   */
+  object?: 'realtime.response';
+
+  /**
+   * The list of output items generated by the response.
+   */
+  output?: Array<ConversationItem>;
+
+  /**
+   * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
+   */
+  output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
+
+  /**
+   * The final status of the response (`completed`, `cancelled`, `failed`, or
+   * `incomplete`).
+   */
+  status?: 'completed' | 'cancelled' | 'failed' | 'incomplete';
+
+  /**
+   * Additional details about the status.
+   */
+  status_details?: RealtimeResponseStatus;
+
+  /**
+   * Sampling temperature for the model, limited to [0.6, 1.2]. Defaults to 0.8.
+   */
+  temperature?: number;
+
+  /**
+   * Usage statistics for the Response, this will correspond to billing. A Realtime
+   * API session will maintain a conversation context and append new Items to the
+   * Conversation, thus output from previous turns (text and audio tokens) will
+   * become the input for later turns.
+   */
+  usage?: RealtimeResponseUsage;
+
+  /**
+   * The voice the model used to respond. Current voice options are `alloy`, `ash`,
+   * `ballad`, `coral`, `echo`, `fable`, `onyx`, `nova`, `sage`, `shimmer`, and
+   * `verse`.
+   */
+  voice?:
+    | (string & {})
+    | 'alloy'
+    | 'ash'
+    | 'ballad'
+    | 'coral'
+    | 'echo'
+    | 'fable'
+    | 'onyx'
+    | 'nova'
+    | 'sage'
+    | 'shimmer'
+    | 'verse';
+}
+
+/**
+ * Additional details about the status.
+ */
+export interface RealtimeResponseStatus {
+  /**
+   * A description of the error that caused the response to fail, populated when the
+   * `status` is `failed`.
+   */
+  error?: RealtimeResponseStatus.Error;
+
+  /**
+   * The reason the Response did not complete. For a `cancelled` Response, one of
+   * `turn_detected` (the server VAD detected a new start of speech) or
+   * `client_cancelled` (the client sent a cancel event). For an `incomplete`
+   * Response, one of `max_output_tokens` or `content_filter` (the server-side safety
+   * filter activated and cut off the response).
+   */
+  reason?: 'turn_detected' | 'client_cancelled' | 'max_output_tokens' | 'content_filter';
+
+  /**
+   * The type of error that caused the response to fail, corresponding with the
+   * `status` field (`completed`, `cancelled`, `incomplete`, `failed`).
+   */
+  type?: 'completed' | 'cancelled' | 'incomplete' | 'failed';
+}
+
+export namespace RealtimeResponseStatus {
+  /**
+   * A description of the error that caused the response to fail, populated when the
+   * `status` is `failed`.
+   */
+  export interface Error {
+    /**
+     * Error code, if any.
+     */
+    code?: string;
+
+    /**
+     * The type of error.
+     */
+    type?: string;
+  }
+}
+
+/**
+ * Usage statistics for the Response, this will correspond to billing. A Realtime
+ * API session will maintain a conversation context and append new Items to the
+ * Conversation, thus output from previous turns (text and audio tokens) will
+ * become the input for later turns.
+ */
+export interface RealtimeResponseUsage {
+  /**
+   * Details about the input tokens used in the Response.
+   */
+  input_token_details?: RealtimeResponseUsage.InputTokenDetails;
+
+  /**
+   * The number of input tokens used in the Response, including text and audio
+   * tokens.
+   */
+  input_tokens?: number;
+
+  /**
+   * Details about the output tokens used in the Response.
+   */
+  output_token_details?: RealtimeResponseUsage.OutputTokenDetails;
+
+  /**
+   * The number of output tokens sent in the Response, including text and audio
+   * tokens.
+   */
+  output_tokens?: number;
+
+  /**
+   * The total number of tokens in the Response including input and output text and
+   * audio tokens.
+   */
+  total_tokens?: number;
+}
+
+export namespace RealtimeResponseUsage {
+  /**
+   * Details about the input tokens used in the Response.
+   */
+  export interface InputTokenDetails {
+    /**
+     * The number of audio tokens used in the Response.
+     */
+    audio_tokens?: number;
+
+    /**
+     * The number of cached tokens used in the Response.
+     */
+    cached_tokens?: number;
+
+    /**
+     * The number of text tokens used in the Response.
+     */
+    text_tokens?: number;
+  }
+
+  /**
+   * Details about the output tokens used in the Response.
+   */
+  export interface OutputTokenDetails {
+    /**
+     * The number of audio tokens used in the Response.
+     */
+    audio_tokens?: number;
+
+    /**
+     * The number of text tokens used in the Response.
+     */
+    text_tokens?: number;
+  }
+}
+
+/**
+ * A realtime server event.
+ */
+export type RealtimeServerEvent =
+  | ConversationCreatedEvent
+  | ConversationItemCreatedEvent
+  | ConversationItemDeletedEvent
+  | ConversationItemInputAudioTranscriptionCompletedEvent
+  | ConversationItemInputAudioTranscriptionDeltaEvent
+  | ConversationItemInputAudioTranscriptionFailedEvent
+  | RealtimeServerEvent.ConversationItemRetrieved
+  | ConversationItemTruncatedEvent
+  | ErrorEvent
+  | InputAudioBufferClearedEvent
+  | InputAudioBufferCommittedEvent
+  | InputAudioBufferSpeechStartedEvent
+  | InputAudioBufferSpeechStoppedEvent
+  | RateLimitsUpdatedEvent
+  | ResponseAudioDeltaEvent
+  | ResponseAudioDoneEvent
+  | ResponseAudioTranscriptDeltaEvent
+  | ResponseAudioTranscriptDoneEvent
+  | ResponseContentPartAddedEvent
+  | ResponseContentPartDoneEvent
+  | ResponseCreatedEvent
+  | ResponseDoneEvent
+  | ResponseFunctionCallArgumentsDeltaEvent
+  | ResponseFunctionCallArgumentsDoneEvent
+  | ResponseOutputItemAddedEvent
+  | ResponseOutputItemDoneEvent
+  | ResponseTextDeltaEvent
+  | ResponseTextDoneEvent
+  | SessionCreatedEvent
+  | SessionUpdatedEvent
+  | TranscriptionSessionUpdatedEvent
+  | RealtimeServerEvent.OutputAudioBufferStarted
+  | RealtimeServerEvent.OutputAudioBufferStopped
+  | RealtimeServerEvent.OutputAudioBufferCleared;
+
+export namespace RealtimeServerEvent {
+  /**
+   * Returned when a conversation item is retrieved with
+   * `conversation.item.retrieve`.
+   */
+  export interface ConversationItemRetrieved {
+    /**
+     * The unique ID of the server event.
+     */
+    event_id: string;
+
+    /**
+     * The item to add to the conversation.
+     */
+    item: RealtimeAPI.ConversationItem;
+
+    /**
+     * The event type, must be `conversation.item.retrieved`.
+     */
+    type: 'conversation.item.retrieved';
+  }
+
+  /**
+   * **WebRTC Only:** Emitted when the server begins streaming audio to the client.
+   * This event is emitted after an audio content part has been added
+   * (`response.content_part.added`) to the response.
+   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
+   */
+  export interface OutputAudioBufferStarted {
+    /**
+     * The unique ID of the server event.
+     */
+    event_id: string;
+
+    /**
+     * The unique ID of the response that produced the audio.
+     */
+    response_id: string;
+
+    /**
+     * The event type, must be `output_audio_buffer.started`.
+     */
+    type: 'output_audio_buffer.started';
+  }
+
+  /**
+   * **WebRTC Only:** Emitted when the output audio buffer has been completely
+   * drained on the server, and no more audio is forthcoming. This event is emitted
+   * after the full response data has been sent to the client (`response.done`).
+   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
+   */
+  export interface OutputAudioBufferStopped {
+    /**
+     * The unique ID of the server event.
+     */
+    event_id: string;
+
+    /**
+     * The unique ID of the response that produced the audio.
+     */
+    response_id: string;
+
+    /**
+     * The event type, must be `output_audio_buffer.stopped`.
+     */
+    type: 'output_audio_buffer.stopped';
+  }
+
+  /**
+   * **WebRTC Only:** Emitted when the output audio buffer is cleared. This happens
+   * either in VAD mode when the user has interrupted
+   * (`input_audio_buffer.speech_started`), or when the client has emitted the
+   * `output_audio_buffer.clear` event to manually cut off the current audio
+   * response.
+   * [Learn more](https://platform.openai.com/docs/guides/realtime-conversations#client-and-server-events-for-audio-in-webrtc).
+   */
+  export interface OutputAudioBufferCleared {
+    /**
+     * The unique ID of the server event.
+     */
+    event_id: string;
+
+    /**
+     * The unique ID of the response that produced the audio.
+     */
+    response_id: string;
+
+    /**
+     * The event type, must be `output_audio_buffer.cleared`.
+     */
+    type: 'output_audio_buffer.cleared';
+  }
+}
+
+/**
+ * Returned when the model-generated audio is updated.
+ */
+export interface ResponseAudioDeltaEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * Base64-encoded audio data delta.
+   */
+  delta: string;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.audio.delta`.
+   */
+  type: 'response.audio.delta';
+}
+
+/**
+ * Returned when the model-generated audio is done. Also emitted when a Response is
+ * interrupted, incomplete, or cancelled.
+ */
+export interface ResponseAudioDoneEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.audio.done`.
+   */
+  type: 'response.audio.done';
+}
+
+/**
+ * Returned when the model-generated transcription of audio output is updated.
+ */
+export interface ResponseAudioTranscriptDeltaEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The transcript delta.
+   */
+  delta: string;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.audio_transcript.delta`.
+   */
+  type: 'response.audio_transcript.delta';
+}
+
+/**
+ * Returned when the model-generated transcription of audio output is done
+ * streaming. Also emitted when a Response is interrupted, incomplete, or
+ * cancelled.
+ */
+export interface ResponseAudioTranscriptDoneEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The final transcript of the audio.
+   */
+  transcript: string;
+
+  /**
+   * The event type, must be `response.audio_transcript.done`.
+   */
+  type: 'response.audio_transcript.done';
+}
+
+/**
+ * Send this event to cancel an in-progress response. The server will respond with
+ * a `response.cancelled` event or an error if there is no response to cancel.
+ */
+export interface ResponseCancelEvent {
+  /**
+   * The event type, must be `response.cancel`.
+   */
+  type: 'response.cancel';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+
+  /**
+   * A specific response ID to cancel - if not provided, will cancel an in-progress
+   * response in the default conversation.
+   */
+  response_id?: string;
+}
+
+/**
+ * Returned when a new content part is added to an assistant message item during
+ * response generation.
+ */
+export interface ResponseContentPartAddedEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item to which the content part was added.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The content part that was added.
+   */
+  part: ResponseContentPartAddedEvent.Part;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.content_part.added`.
+   */
+  type: 'response.content_part.added';
+}
+
+export namespace ResponseContentPartAddedEvent {
+  /**
+   * The content part that was added.
+   */
+  export interface Part {
+    /**
+     * Base64-encoded audio data (if type is "audio").
+     */
+    audio?: string;
+
+    /**
+     * The text content (if type is "text").
+     */
+    text?: string;
+
+    /**
+     * The transcript of the audio (if type is "audio").
+     */
+    transcript?: string;
+
+    /**
+     * The content type ("text", "audio").
+     */
+    type?: 'text' | 'audio';
+  }
+}
+
+/**
+ * Returned when a content part is done streaming in an assistant message item.
+ * Also emitted when a Response is interrupted, incomplete, or cancelled.
+ */
+export interface ResponseContentPartDoneEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The content part that is done.
+   */
+  part: ResponseContentPartDoneEvent.Part;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.content_part.done`.
+   */
+  type: 'response.content_part.done';
+}
+
+export namespace ResponseContentPartDoneEvent {
+  /**
+   * The content part that is done.
+   */
+  export interface Part {
+    /**
+     * Base64-encoded audio data (if type is "audio").
+     */
+    audio?: string;
+
+    /**
+     * The text content (if type is "text").
+     */
+    text?: string;
+
+    /**
+     * The transcript of the audio (if type is "audio").
+     */
+    transcript?: string;
+
+    /**
+     * The content type ("text", "audio").
+     */
+    type?: 'text' | 'audio';
+  }
+}
+
+/**
+ * This event instructs the server to create a Response, which means triggering
+ * model inference. When in Server VAD mode, the server will create Responses
+ * automatically.
+ *
+ * A Response will include at least one Item, and may have two, in which case the
+ * second will be a function call. These Items will be appended to the conversation
+ * history.
+ *
+ * The server will respond with a `response.created` event, events for Items and
+ * content created, and finally a `response.done` event to indicate the Response is
+ * complete.
+ *
+ * The `response.create` event includes inference configuration like
+ * `instructions`, and `temperature`. These fields will override the Session's
+ * configuration for this Response only.
+ */
+export interface ResponseCreateEvent {
+  /**
+   * The event type, must be `response.create`.
+   */
+  type: 'response.create';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+
+  /**
+   * Create a new Realtime response with these parameters
+   */
+  response?: ResponseCreateEvent.Response;
+}
+
+export namespace ResponseCreateEvent {
+  /**
+   * Create a new Realtime response with these parameters
+   */
+  export interface Response {
+    /**
+     * Controls which conversation the response is added to. Currently supports `auto`
+     * and `none`, with `auto` as the default value. The `auto` value means that the
+     * contents of the response will be added to the default conversation. Set this to
+     * `none` to create an out-of-band response which will not add items to default
+     * conversation.
+     */
+    conversation?: (string & {}) | 'auto' | 'none';
+
+    /**
+     * Input items to include in the prompt for the model. Using this field creates a
+     * new context for this Response instead of using the default conversation. An
+     * empty array `[]` will clear the context for this Response. Note that this can
+     * include references to items from the default conversation.
+     */
+    input?: Array<RealtimeAPI.ConversationItemWithReference>;
+
+    /**
+     * The default system instructions (i.e. system message) prepended to model calls.
+     * This field allows the client to guide the model on desired responses. The model
+     * can be instructed on response content and format, (e.g. "be extremely succinct",
+     * "act friendly", "here are examples of good responses") and on audio behavior
+     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
+     * instructions are not guaranteed to be followed by the model, but they provide
+     * guidance to the model on the desired behavior.
+     *
+     * Note that the server sets default instructions which will be used if this field
+     * is not set and are visible in the `session.created` event at the start of the
+     * session.
+     */
+    instructions?: string;
+
+    /**
+     * Maximum number of output tokens for a single assistant response, inclusive of
+     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
+     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
+     */
+    max_response_output_tokens?: number | 'inf';
+
+    /**
+     * Set of 16 key-value pairs that can be attached to an object. This can be useful
+     * for storing additional information about the object in a structured format, and
+     * querying for objects via API or the dashboard.
+     *
+     * Keys are strings with a maximum length of 64 characters. Values are strings with
+     * a maximum length of 512 characters.
+     */
+    metadata?: Shared.Metadata | null;
+
+    /**
+     * The set of modalities the model can respond with. To disable audio, set this to
+     * ["text"].
+     */
+    modalities?: Array<'text' | 'audio'>;
+
+    /**
+     * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
+     */
+    output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
+
+    /**
+     * Sampling temperature for the model, limited to [0.6, 1.2]. Defaults to 0.8.
+     */
+    temperature?: number;
+
+    /**
+     * How the model chooses tools. Options are `auto`, `none`, `required`, or specify
+     * a function, like `{"type": "function", "function": {"name": "my_function"}}`.
+     */
+    tool_choice?: string;
+
+    /**
+     * Tools (functions) available to the model.
+     */
+    tools?: Array<Response.Tool>;
+
+    /**
+     * The voice the model uses to respond. Voice cannot be changed during the session
+     * once the model has responded with audio at least once. Current voice options are
+     * `alloy`, `ash`, `ballad`, `coral`, `echo`, `fable`, `onyx`, `nova`, `sage`,
+     * `shimmer`, and `verse`.
+     */
+    voice?:
+      | (string & {})
+      | 'alloy'
+      | 'ash'
+      | 'ballad'
+      | 'coral'
+      | 'echo'
+      | 'fable'
+      | 'onyx'
+      | 'nova'
+      | 'sage'
+      | 'shimmer'
+      | 'verse';
+  }
+
+  export namespace Response {
+    export interface Tool {
+      /**
+       * The description of the function, including guidance on when and how to call it,
+       * and guidance about what to tell the user when calling (if anything).
+       */
+      description?: string;
+
+      /**
+       * The name of the function.
+       */
+      name?: string;
+
+      /**
+       * Parameters of the function in JSON Schema.
+       */
+      parameters?: unknown;
+
+      /**
+       * The type of the tool, i.e. `function`.
+       */
+      type?: 'function';
+    }
+  }
+}
+
+/**
+ * Returned when a new Response is created. The first event of response creation,
+ * where the response is in an initial state of `in_progress`.
+ */
+export interface ResponseCreatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The response resource.
+   */
+  response: RealtimeResponse;
+
+  /**
+   * The event type, must be `response.created`.
+   */
+  type: 'response.created';
+}
+
+/**
+ * Returned when a Response is done streaming. Always emitted, no matter the final
+ * state. The Response object included in the `response.done` event will include
+ * all output Items in the Response but will omit the raw audio data.
+ */
+export interface ResponseDoneEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The response resource.
+   */
+  response: RealtimeResponse;
+
+  /**
+   * The event type, must be `response.done`.
+   */
+  type: 'response.done';
+}
+
+/**
+ * Returned when the model-generated function call arguments are updated.
+ */
+export interface ResponseFunctionCallArgumentsDeltaEvent {
+  /**
+   * The ID of the function call.
+   */
+  call_id: string;
+
+  /**
+   * The arguments delta as a JSON string.
+   */
+  delta: string;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the function call item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.function_call_arguments.delta`.
+   */
+  type: 'response.function_call_arguments.delta';
+}
+
+/**
+ * Returned when the model-generated function call arguments are done streaming.
+ * Also emitted when a Response is interrupted, incomplete, or cancelled.
+ */
+export interface ResponseFunctionCallArgumentsDoneEvent {
+  /**
+   * The final arguments as a JSON string.
+   */
+  arguments: string;
+
+  /**
+   * The ID of the function call.
+   */
+  call_id: string;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the function call item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.function_call_arguments.done`.
+   */
+  type: 'response.function_call_arguments.done';
+}
+
+/**
+ * Returned when a new Item is created during Response generation.
+ */
+export interface ResponseOutputItemAddedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The item to add to the conversation.
+   */
+  item: ConversationItem;
+
+  /**
+   * The index of the output item in the Response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the Response to which the item belongs.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.output_item.added`.
+   */
+  type: 'response.output_item.added';
+}
+
+/**
+ * Returned when an Item is done streaming. Also emitted when a Response is
+ * interrupted, incomplete, or cancelled.
+ */
+export interface ResponseOutputItemDoneEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The item to add to the conversation.
+   */
+  item: ConversationItem;
+
+  /**
+   * The index of the output item in the Response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the Response to which the item belongs.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.output_item.done`.
+   */
+  type: 'response.output_item.done';
+}
+
+/**
+ * Returned when the text value of a "text" content part is updated.
+ */
+export interface ResponseTextDeltaEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The text delta.
+   */
+  delta: string;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The event type, must be `response.text.delta`.
+   */
+  type: 'response.text.delta';
+}
+
+/**
+ * Returned when the text value of a "text" content part is done streaming. Also
+ * emitted when a Response is interrupted, incomplete, or cancelled.
+ */
+export interface ResponseTextDoneEvent {
+  /**
+   * The index of the content part in the item's content array.
+   */
+  content_index: number;
+
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * The ID of the item.
+   */
+  item_id: string;
+
+  /**
+   * The index of the output item in the response.
+   */
+  output_index: number;
+
+  /**
+   * The ID of the response.
+   */
+  response_id: string;
+
+  /**
+   * The final text content.
+   */
+  text: string;
+
+  /**
+   * The event type, must be `response.text.done`.
+   */
+  type: 'response.text.done';
+}
+
+/**
+ * Returned when a Session is created. Emitted automatically when a new connection
+ * is established as the first server event. This event will contain the default
+ * Session configuration.
+ */
+export interface SessionCreatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * Realtime session object configuration.
+   */
+  session: SessionsAPI.Session;
+
+  /**
+   * The event type, must be `session.created`.
+   */
+  type: 'session.created';
+}
+
+/**
+ * Send this event to update the session’s default configuration. The client may
+ * send this event at any time to update any field, except for `voice`. However,
+ * note that once a session has been initialized with a particular `model`, it
+ * can’t be changed to another model using `session.update`.
+ *
+ * When the server receives a `session.update`, it will respond with a
+ * `session.updated` event showing the full, effective configuration. Only the
+ * fields that are present are updated. To clear a field like `instructions`, pass
+ * an empty string.
+ */
+export interface SessionUpdateEvent {
+  /**
+   * Realtime session object configuration.
+   */
+  session: SessionUpdateEvent.Session;
+
+  /**
+   * The event type, must be `session.update`.
+   */
+  type: 'session.update';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+export namespace SessionUpdateEvent {
+  /**
+   * Realtime session object configuration.
+   */
+  export interface Session {
+    /**
+     * Configuration options for the generated client secret.
+     */
+    client_secret?: Session.ClientSecret;
+
+    /**
+     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`. For
+     * `pcm16`, input audio must be 16-bit PCM at a 24kHz sample rate, single channel
+     * (mono), and little-endian byte order.
+     */
+    input_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
+
+    /**
+     * Configuration for input audio noise reduction. This can be set to `null` to turn
+     * off. Noise reduction filters audio added to the input audio buffer before it is
+     * sent to VAD and the model. Filtering the audio can improve VAD and turn
+     * detection accuracy (reducing false positives) and model performance by improving
+     * perception of the input audio.
+     */
+    input_audio_noise_reduction?: Session.InputAudioNoiseReduction;
+
+    /**
+     * Configuration for input audio transcription, defaults to off and can be set to
+     * `null` to turn off once on. Input audio transcription is not native to the
+     * model, since the model consumes audio directly. Transcription runs
+     * asynchronously through
+     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
+     * and should be treated as guidance of input audio content rather than precisely
+     * what the model heard. The client can optionally set the language and prompt for
+     * transcription, these offer additional guidance to the transcription service.
+     */
+    input_audio_transcription?: Session.InputAudioTranscription;
+
+    /**
+     * The default system instructions (i.e. system message) prepended to model calls.
+     * This field allows the client to guide the model on desired responses. The model
+     * can be instructed on response content and format, (e.g. "be extremely succinct",
+     * "act friendly", "here are examples of good responses") and on audio behavior
+     * (e.g. "talk quickly", "inject emotion into your voice", "laugh frequently"). The
+     * instructions are not guaranteed to be followed by the model, but they provide
+     * guidance to the model on the desired behavior.
+     *
+     * Note that the server sets default instructions which will be used if this field
+     * is not set and are visible in the `session.created` event at the start of the
+     * session.
+     */
+    instructions?: string;
+
+    /**
+     * Maximum number of output tokens for a single assistant response, inclusive of
+     * tool calls. Provide an integer between 1 and 4096 to limit output tokens, or
+     * `inf` for the maximum available tokens for a given model. Defaults to `inf`.
+     */
+    max_response_output_tokens?: number | 'inf';
+
+    /**
+     * The set of modalities the model can respond with. To disable audio, set this to
+     * ["text"].
+     */
+    modalities?: Array<'text' | 'audio'>;
+
+    /**
+     * The Realtime model used for this session.
+     */
+    model?:
+      | 'gpt-4o-realtime-preview'
+      | 'gpt-4o-realtime-preview-2024-10-01'
+      | 'gpt-4o-realtime-preview-2024-12-17'
+      | 'gpt-4o-mini-realtime-preview'
+      | 'gpt-4o-mini-realtime-preview-2024-12-17';
+
+    /**
+     * The format of output audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`.
+     * For `pcm16`, output audio is sampled at a rate of 24kHz.
+     */
+    output_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
+
+    /**
+     * Sampling temperature for the model, limited to [0.6, 1.2]. For audio models a
+     * temperature of 0.8 is highly recommended for best performance.
+     */
+    temperature?: number;
+
+    /**
+     * How the model chooses tools. Options are `auto`, `none`, `required`, or specify
+     * a function.
+     */
+    tool_choice?: string;
+
+    /**
+     * Tools (functions) available to the model.
+     */
+    tools?: Array<Session.Tool>;
+
+    /**
+     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
+     * set to `null` to turn off, in which case the client must manually trigger model
+     * response. Server VAD means that the model will detect the start and end of
+     * speech based on audio volume and respond at the end of user speech. Semantic VAD
+     * is more advanced and uses a turn detection model (in conjuction with VAD) to
+     * semantically estimate whether the user has finished speaking, then dynamically
+     * sets a timeout based on this probability. For example, if user audio trails off
+     * with "uhhm", the model will score a low probability of turn end and wait longer
+     * for the user to continue speaking. This can be useful for more natural
+     * conversations, but may have a higher latency.
+     */
+    turn_detection?: Session.TurnDetection;
+
+    /**
+     * The voice the model uses to respond. Voice cannot be changed during the session
+     * once the model has responded with audio at least once. Current voice options are
+     * `alloy`, `ash`, `ballad`, `coral`, `echo`, `fable`, `onyx`, `nova`, `sage`,
+     * `shimmer`, and `verse`.
+     */
+    voice?:
+      | (string & {})
+      | 'alloy'
+      | 'ash'
+      | 'ballad'
+      | 'coral'
+      | 'echo'
+      | 'fable'
+      | 'onyx'
+      | 'nova'
+      | 'sage'
+      | 'shimmer'
+      | 'verse';
+  }
+
+  export namespace Session {
+    /**
+     * Configuration options for the generated client secret.
+     */
+    export interface ClientSecret {
+      /**
+       * Configuration for the ephemeral token expiration.
+       */
+      expires_at?: ClientSecret.ExpiresAt;
+    }
+
+    export namespace ClientSecret {
+      /**
+       * Configuration for the ephemeral token expiration.
+       */
+      export interface ExpiresAt {
+        /**
+         * The anchor point for the ephemeral token expiration. Only `created_at` is
+         * currently supported.
+         */
+        anchor?: 'created_at';
+
+        /**
+         * The number of seconds from the anchor point to the expiration. Select a value
+         * between `10` and `7200`.
+         */
+        seconds?: number;
+      }
+    }
+
+    /**
+     * Configuration for input audio noise reduction. This can be set to `null` to turn
+     * off. Noise reduction filters audio added to the input audio buffer before it is
+     * sent to VAD and the model. Filtering the audio can improve VAD and turn
+     * detection accuracy (reducing false positives) and model performance by improving
+     * perception of the input audio.
+     */
+    export interface InputAudioNoiseReduction {
+      /**
+       * Type of noise reduction. `near_field` is for close-talking microphones such as
+       * headphones, `far_field` is for far-field microphones such as laptop or
+       * conference room microphones.
+       */
+      type?: 'near_field' | 'far_field';
+    }
+
+    /**
+     * Configuration for input audio transcription, defaults to off and can be set to
+     * `null` to turn off once on. Input audio transcription is not native to the
+     * model, since the model consumes audio directly. Transcription runs
+     * asynchronously through
+     * [the /audio/transcriptions endpoint](https://platform.openai.com/docs/api-reference/audio/createTranscription)
+     * and should be treated as guidance of input audio content rather than precisely
+     * what the model heard. The client can optionally set the language and prompt for
+     * transcription, these offer additional guidance to the transcription service.
+     */
+    export interface InputAudioTranscription {
+      /**
+       * The language of the input audio. Supplying the input language in
+       * [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`)
+       * format will improve accuracy and latency.
+       */
+      language?: string;
+
+      /**
+       * The model to use for transcription, current options are `gpt-4o-transcribe`,
+       * `gpt-4o-mini-transcribe`, and `whisper-1`.
+       */
+      model?: string;
+
+      /**
+       * An optional text to guide the model's style or continue a previous audio
+       * segment. For `whisper-1`, the
+       * [prompt is a list of keywords](https://platform.openai.com/docs/guides/speech-to-text#prompting).
+       * For `gpt-4o-transcribe` models, the prompt is a free text string, for example
+       * "expect words related to technology".
+       */
+      prompt?: string;
+    }
+
+    export interface Tool {
+      /**
+       * The description of the function, including guidance on when and how to call it,
+       * and guidance about what to tell the user when calling (if anything).
+       */
+      description?: string;
+
+      /**
+       * The name of the function.
+       */
+      name?: string;
+
+      /**
+       * Parameters of the function in JSON Schema.
+       */
+      parameters?: unknown;
+
+      /**
+       * The type of the tool, i.e. `function`.
+       */
+      type?: 'function';
+    }
+
+    /**
+     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
+     * set to `null` to turn off, in which case the client must manually trigger model
+     * response. Server VAD means that the model will detect the start and end of
+     * speech based on audio volume and respond at the end of user speech. Semantic VAD
+     * is more advanced and uses a turn detection model (in conjuction with VAD) to
+     * semantically estimate whether the user has finished speaking, then dynamically
+     * sets a timeout based on this probability. For example, if user audio trails off
+     * with "uhhm", the model will score a low probability of turn end and wait longer
+     * for the user to continue speaking. This can be useful for more natural
+     * conversations, but may have a higher latency.
+     */
+    export interface TurnDetection {
+      /**
+       * Whether or not to automatically generate a response when a VAD stop event
+       * occurs.
+       */
+      create_response?: boolean;
+
+      /**
+       * Used only for `semantic_vad` mode. The eagerness of the model to respond. `low`
+       * will wait longer for the user to continue speaking, `high` will respond more
+       * quickly. `auto` is the default and is equivalent to `medium`.
+       */
+      eagerness?: 'low' | 'medium' | 'high' | 'auto';
+
+      /**
+       * Whether or not to automatically interrupt any ongoing response with output to
+       * the default conversation (i.e. `conversation` of `auto`) when a VAD start event
+       * occurs.
+       */
+      interrupt_response?: boolean;
+
+      /**
+       * Used only for `server_vad` mode. Amount of audio to include before the VAD
+       * detected speech (in milliseconds). Defaults to 300ms.
+       */
+      prefix_padding_ms?: number;
+
+      /**
+       * Used only for `server_vad` mode. Duration of silence to detect speech stop (in
+       * milliseconds). Defaults to 500ms. With shorter values the model will respond
+       * more quickly, but may jump in on short pauses from the user.
+       */
+      silence_duration_ms?: number;
+
+      /**
+       * Used only for `server_vad` mode. Activation threshold for VAD (0.0 to 1.0), this
+       * defaults to 0.5. A higher threshold will require louder audio to activate the
+       * model, and thus might perform better in noisy environments.
+       */
+      threshold?: number;
+
+      /**
+       * Type of turn detection.
+       */
+      type?: 'server_vad' | 'semantic_vad';
+    }
+  }
+}
+
+/**
+ * Returned when a session is updated with a `session.update` event, unless there
+ * is an error.
+ */
+export interface SessionUpdatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * Realtime session object configuration.
+   */
+  session: SessionsAPI.Session;
+
+  /**
+   * The event type, must be `session.updated`.
+   */
+  type: 'session.updated';
+}
+
+/**
+ * Send this event to update a transcription session.
+ */
+export interface TranscriptionSessionUpdate {
+  /**
+   * Realtime transcription session object configuration.
+   */
+  session: TranscriptionSessionUpdate.Session;
+
+  /**
+   * The event type, must be `transcription_session.update`.
+   */
+  type: 'transcription_session.update';
+
+  /**
+   * Optional client-generated ID used to identify this event.
+   */
+  event_id?: string;
+}
+
+export namespace TranscriptionSessionUpdate {
+  /**
+   * Realtime transcription session object configuration.
+   */
+  export interface Session {
+    /**
+     * Configuration options for the generated client secret.
+     */
+    client_secret?: Session.ClientSecret;
+
+    /**
+     * The set of items to include in the transcription. Current available items are:
+     *
+     * - `item.input_audio_transcription.logprobs`
+     */
+    include?: Array<string>;
+
+    /**
+     * The format of input audio. Options are `pcm16`, `g711_ulaw`, or `g711_alaw`. For
+     * `pcm16`, input audio must be 16-bit PCM at a 24kHz sample rate, single channel
+     * (mono), and little-endian byte order.
+     */
+    input_audio_format?: 'pcm16' | 'g711_ulaw' | 'g711_alaw';
+
+    /**
+     * Configuration for input audio noise reduction. This can be set to `null` to turn
+     * off. Noise reduction filters audio added to the input audio buffer before it is
+     * sent to VAD and the model. Filtering the audio can improve VAD and turn
+     * detection accuracy (reducing false positives) and model performance by improving
+     * perception of the input audio.
+     */
+    input_audio_noise_reduction?: Session.InputAudioNoiseReduction;
+
+    /**
+     * Configuration for input audio transcription. The client can optionally set the
+     * language and prompt for transcription, these offer additional guidance to the
+     * transcription service.
+     */
+    input_audio_transcription?: Session.InputAudioTranscription;
+
+    /**
+     * The set of modalities the model can respond with. To disable audio, set this to
+     * ["text"].
+     */
+    modalities?: Array<'text' | 'audio'>;
+
+    /**
+     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
+     * set to `null` to turn off, in which case the client must manually trigger model
+     * response. Server VAD means that the model will detect the start and end of
+     * speech based on audio volume and respond at the end of user speech. Semantic VAD
+     * is more advanced and uses a turn detection model (in conjuction with VAD) to
+     * semantically estimate whether the user has finished speaking, then dynamically
+     * sets a timeout based on this probability. For example, if user audio trails off
+     * with "uhhm", the model will score a low probability of turn end and wait longer
+     * for the user to continue speaking. This can be useful for more natural
+     * conversations, but may have a higher latency.
+     */
+    turn_detection?: Session.TurnDetection;
+  }
+
+  export namespace Session {
+    /**
+     * Configuration options for the generated client secret.
+     */
+    export interface ClientSecret {
+      /**
+       * Configuration for the ephemeral token expiration.
+       */
+      expires_at?: ClientSecret.ExpiresAt;
+    }
+
+    export namespace ClientSecret {
+      /**
+       * Configuration for the ephemeral token expiration.
+       */
+      export interface ExpiresAt {
+        /**
+         * The anchor point for the ephemeral token expiration. Only `created_at` is
+         * currently supported.
+         */
+        anchor?: 'created_at';
+
+        /**
+         * The number of seconds from the anchor point to the expiration. Select a value
+         * between `10` and `7200`.
+         */
+        seconds?: number;
+      }
+    }
+
+    /**
+     * Configuration for input audio noise reduction. This can be set to `null` to turn
+     * off. Noise reduction filters audio added to the input audio buffer before it is
+     * sent to VAD and the model. Filtering the audio can improve VAD and turn
+     * detection accuracy (reducing false positives) and model performance by improving
+     * perception of the input audio.
+     */
+    export interface InputAudioNoiseReduction {
+      /**
+       * Type of noise reduction. `near_field` is for close-talking microphones such as
+       * headphones, `far_field` is for far-field microphones such as laptop or
+       * conference room microphones.
+       */
+      type?: 'near_field' | 'far_field';
+    }
+
+    /**
+     * Configuration for input audio transcription. The client can optionally set the
+     * language and prompt for transcription, these offer additional guidance to the
+     * transcription service.
+     */
+    export interface InputAudioTranscription {
+      /**
+       * The language of the input audio. Supplying the input language in
+       * [ISO-639-1](https://en.wikipedia.org/wiki/List_of_ISO_639-1_codes) (e.g. `en`)
+       * format will improve accuracy and latency.
+       */
+      language?: string;
+
+      /**
+       * The model to use for transcription, current options are `gpt-4o-transcribe`,
+       * `gpt-4o-mini-transcribe`, and `whisper-1`.
+       */
+      model?: 'gpt-4o-transcribe' | 'gpt-4o-mini-transcribe' | 'whisper-1';
+
+      /**
+       * An optional text to guide the model's style or continue a previous audio
+       * segment. For `whisper-1`, the
+       * [prompt is a list of keywords](https://platform.openai.com/docs/guides/speech-to-text#prompting).
+       * For `gpt-4o-transcribe` models, the prompt is a free text string, for example
+       * "expect words related to technology".
+       */
+      prompt?: string;
+    }
+
+    /**
+     * Configuration for turn detection, ether Server VAD or Semantic VAD. This can be
+     * set to `null` to turn off, in which case the client must manually trigger model
+     * response. Server VAD means that the model will detect the start and end of
+     * speech based on audio volume and respond at the end of user speech. Semantic VAD
+     * is more advanced and uses a turn detection model (in conjuction with VAD) to
+     * semantically estimate whether the user has finished speaking, then dynamically
+     * sets a timeout based on this probability. For example, if user audio trails off
+     * with "uhhm", the model will score a low probability of turn end and wait longer
+     * for the user to continue speaking. This can be useful for more natural
+     * conversations, but may have a higher latency.
+     */
+    export interface TurnDetection {
+      /**
+       * Whether or not to automatically generate a response when a VAD stop event
+       * occurs. Not available for transcription sessions.
+       */
+      create_response?: boolean;
+
+      /**
+       * Used only for `semantic_vad` mode. The eagerness of the model to respond. `low`
+       * will wait longer for the user to continue speaking, `high` will respond more
+       * quickly. `auto` is the default and is equivalent to `medium`.
+       */
+      eagerness?: 'low' | 'medium' | 'high' | 'auto';
+
+      /**
+       * Whether or not to automatically interrupt any ongoing response with output to
+       * the default conversation (i.e. `conversation` of `auto`) when a VAD start event
+       * occurs. Not available for transcription sessions.
+       */
+      interrupt_response?: boolean;
+
+      /**
+       * Used only for `server_vad` mode. Amount of audio to include before the VAD
+       * detected speech (in milliseconds). Defaults to 300ms.
+       */
+      prefix_padding_ms?: number;
+
+      /**
+       * Used only for `server_vad` mode. Duration of silence to detect speech stop (in
+       * milliseconds). Defaults to 500ms. With shorter values the model will respond
+       * more quickly, but may jump in on short pauses from the user.
+       */
+      silence_duration_ms?: number;
+
+      /**
+       * Used only for `server_vad` mode. Activation threshold for VAD (0.0 to 1.0), this
+       * defaults to 0.5. A higher threshold will require louder audio to activate the
+       * model, and thus might perform better in noisy environments.
+       */
+      threshold?: number;
+
+      /**
+       * Type of turn detection.
+       */
+      type?: 'server_vad' | 'semantic_vad';
+    }
+  }
+}
+
+/**
+ * Returned when a transcription session is updated with a
+ * `transcription_session.update` event, unless there is an error.
+ */
+export interface TranscriptionSessionUpdatedEvent {
+  /**
+   * The unique ID of the server event.
+   */
+  event_id: string;
+
+  /**
+   * A new Realtime transcription session configuration.
+   *
+   * When a session is created on the server via REST API, the session object also
+   * contains an ephemeral key. Default TTL for keys is 10 minutes. This property is
+   * not present when a session is updated via the WebSocket API.
+   */
+  session: TranscriptionSessionsAPI.TranscriptionSession;
+
+  /**
+   * The event type, must be `transcription_session.updated`.
+   */
+  type: 'transcription_session.updated';
+}
+
+Realtime.Sessions = Sessions;
+Realtime.TranscriptionSessions = TranscriptionSessions;
+
+export declare namespace Realtime {
+  export {
+    type ConversationCreatedEvent as ConversationCreatedEvent,
+    type ConversationItem as ConversationItem,
+    type ConversationItemContent as ConversationItemContent,
+    type ConversationItemCreateEvent as ConversationItemCreateEvent,
+    type ConversationItemCreatedEvent as ConversationItemCreatedEvent,
+    type ConversationItemDeleteEvent as ConversationItemDeleteEvent,
+    type ConversationItemDeletedEvent as ConversationItemDeletedEvent,
+    type ConversationItemInputAudioTranscriptionCompletedEvent as ConversationItemInputAudioTranscriptionCompletedEvent,
+    type ConversationItemInputAudioTranscriptionDeltaEvent as ConversationItemInputAudioTranscriptionDeltaEvent,
+    type ConversationItemInputAudioTranscriptionFailedEvent as ConversationItemInputAudioTranscriptionFailedEvent,
+    type ConversationItemRetrieveEvent as ConversationItemRetrieveEvent,
+    type ConversationItemTruncateEvent as ConversationItemTruncateEvent,
+    type ConversationItemTruncatedEvent as ConversationItemTruncatedEvent,
+    type ConversationItemWithReference as ConversationItemWithReference,
+    type ErrorEvent as ErrorEvent,
+    type InputAudioBufferAppendEvent as InputAudioBufferAppendEvent,
+    type InputAudioBufferClearEvent as InputAudioBufferClearEvent,
+    type InputAudioBufferClearedEvent as InputAudioBufferClearedEvent,
+    type InputAudioBufferCommitEvent as InputAudioBufferCommitEvent,
+    type InputAudioBufferCommittedEvent as InputAudioBufferCommittedEvent,
+    type InputAudioBufferSpeechStartedEvent as InputAudioBufferSpeechStartedEvent,
+    type InputAudioBufferSpeechStoppedEvent as InputAudioBufferSpeechStoppedEvent,
+    type RateLimitsUpdatedEvent as RateLimitsUpdatedEvent,
+    type RealtimeClientEvent as RealtimeClientEvent,
+    type RealtimeResponse as RealtimeResponse,
+    type RealtimeResponseStatus as RealtimeResponseStatus,
+    type RealtimeResponseUsage as RealtimeResponseUsage,
+    type RealtimeServerEvent as RealtimeServerEvent,
+    type ResponseAudioDeltaEvent as ResponseAudioDeltaEvent,
+    type ResponseAudioDoneEvent as ResponseAudioDoneEvent,
+    type ResponseAudioTranscriptDeltaEvent as ResponseAudioTranscriptDeltaEvent,
+    type ResponseAudioTranscriptDoneEvent as ResponseAudioTranscriptDoneEvent,
+    type ResponseCancelEvent as ResponseCancelEvent,
+    type ResponseContentPartAddedEvent as ResponseContentPartAddedEvent,
+    type ResponseContentPartDoneEvent as ResponseContentPartDoneEvent,
+    type ResponseCreateEvent as ResponseCreateEvent,
+    type ResponseCreatedEvent as ResponseCreatedEvent,
+    type ResponseDoneEvent as ResponseDoneEvent,
+    type ResponseFunctionCallArgumentsDeltaEvent as ResponseFunctionCallArgumentsDeltaEvent,
+    type ResponseFunctionCallArgumentsDoneEvent as ResponseFunctionCallArgumentsDoneEvent,
+    type ResponseOutputItemAddedEvent as ResponseOutputItemAddedEvent,
+    type ResponseOutputItemDoneEvent as ResponseOutputItemDoneEvent,
+    type ResponseTextDeltaEvent as ResponseTextDeltaEvent,
+    type ResponseTextDoneEvent as ResponseTextDoneEvent,
+    type SessionCreatedEvent as SessionCreatedEvent,
+    type SessionUpdateEvent as SessionUpdateEvent,
+    type SessionUpdatedEvent as SessionUpdatedEvent,
+    type TranscriptionSessionUpdate as TranscriptionSessionUpdate,
+    type TranscriptionSessionUpdatedEvent as TranscriptionSessionUpdatedEvent,
+  };
+
+  export {
+    Sessions as Sessions,
+    type SessionsAPISession as Session,
+    type SessionCreateResponse as SessionCreateResponse,
+    type SessionCreateParams as SessionCreateParams,
+  };
+
+  export {
+    TranscriptionSessions as TranscriptionSessions,
+    type TranscriptionSession as TranscriptionSession,
+    type TranscriptionSessionCreateParams as TranscriptionSessionCreateParams,
+  };
+}