feat(api): adding new image model support

Author: stainless-app[bot]

.stats.yml

@@ -1,4 +1,4 @@
 configured_endpoints: 97
-openapi_spec_url: https://storage.googleapis.com/stainless-sdk-openapi-specs/openai%2Fopenai-5633633cc38734869cf7d993f7b549bb8e4d10e0ec45381ec2cd91507cd8eb8f.yml
-openapi_spec_hash: c855121b2b2324b99499c9244c21d24d
-config_hash: d20837393b73efdb19cd08e04c1cc9a1
+openapi_spec_url: https://storage.googleapis.com/stainless-sdk-openapi-specs/openai%2Fopenai-8b68ae6b807dca92e914da1dd9e835a20f69b075e79102a264367fd7fddddb33.yml
+openapi_spec_hash: b6ade5b1a6327339e6669e1134de2d03
+config_hash: b597cd9a31e9e5ec709e2eefb4c54122

api.md

@@ -249,7 +249,7 @@ Methods:
 
 - <code title="post /fine_tuning/checkpoints/{fine_tuned_model_checkpoint}/permissions">client.fineTuning.checkpoints.permissions.<a href="./src/resources/fine-tuning/checkpoints/permissions.ts">create</a>(fineTunedModelCheckpoint, { ...params }) -> PermissionCreateResponsesPage</code>
 - <code title="get /fine_tuning/checkpoints/{fine_tuned_model_checkpoint}/permissions">client.fineTuning.checkpoints.permissions.<a href="./src/resources/fine-tuning/checkpoints/permissions.ts">retrieve</a>(fineTunedModelCheckpoint, { ...params }) -> PermissionRetrieveResponse</code>
-- <code title="delete /fine_tuning/checkpoints/{fine_tuned_model_checkpoint}/permissions">client.fineTuning.checkpoints.permissions.<a href="./src/resources/fine-tuning/checkpoints/permissions.ts">del</a>(fineTunedModelCheckpoint) -> PermissionDeleteResponse</code>
+- <code title="delete /fine_tuning/checkpoints/{fine_tuned_model_checkpoint}/permissions/{permission_id}">client.fineTuning.checkpoints.permissions.<a href="./src/resources/fine-tuning/checkpoints/permissions.ts">del</a>(fineTunedModelCheckpoint, permissionId) -> PermissionDeleteResponse</code>
 
 # VectorStores
 
@@ -626,6 +626,10 @@ Types:
 - <code><a href="./src/resources/responses/responses.ts">ResponseOutputRefusal</a></code>
 - <code><a href="./src/resources/responses/responses.ts">ResponseOutputText</a></code>
 - <code><a href="./src/resources/responses/responses.ts">ResponseReasoningItem</a></code>
+- <code><a href="./src/resources/responses/responses.ts">ResponseReasoningSummaryPartAddedEvent</a></code>
+- <code><a href="./src/resources/responses/responses.ts">ResponseReasoningSummaryPartDoneEvent</a></code>
+- <code><a href="./src/resources/responses/responses.ts">ResponseReasoningSummaryTextDeltaEvent</a></code>
+- <code><a href="./src/resources/responses/responses.ts">ResponseReasoningSummaryTextDoneEvent</a></code>
 - <code><a href="./src/resources/responses/responses.ts">ResponseRefusalDeltaEvent</a></code>
 - <code><a href="./src/resources/responses/responses.ts">ResponseRefusalDoneEvent</a></code>
 - <code><a href="./src/resources/responses/responses.ts">ResponseStatus</a></code>

src/resources/beta/realtime/realtime.ts

@@ -915,12 +915,34 @@ export type RealtimeClientEvent =
   | 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-model-capabilities#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.
  */
@@ -1174,7 +1196,10 @@ export type RealtimeServerEvent =
   | ResponseTextDoneEvent
   | SessionCreatedEvent
   | SessionUpdatedEvent
-  | TranscriptionSessionUpdatedEvent;
+  | TranscriptionSessionUpdatedEvent
+  | RealtimeServerEvent.OutputAudioBufferStarted
+  | RealtimeServerEvent.OutputAudioBufferStopped
+  | RealtimeServerEvent.OutputAudioBufferCleared;
 
 export namespace RealtimeServerEvent {
   /**
@@ -1197,6 +1222,77 @@ export namespace RealtimeServerEvent {
      */
     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-model-capabilities#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-model-capabilities#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-model-capabilities#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';
+  }
 }
 
 /**

src/resources/beta/threads/threads.ts

@@ -686,9 +686,7 @@ export interface ThreadCreateAndRunParamsBase {
    * Override the tools the assistant can use for this run. This is useful for
    * modifying the behavior on a per-run basis.
    */
-  tools?: Array<
-    AssistantsAPI.CodeInterpreterTool | AssistantsAPI.FileSearchTool | AssistantsAPI.FunctionTool
-  > | null;
+  tools?: Array<AssistantsAPI.AssistantTool> | null;
 
   /**
    * An alternative to sampling with temperature, called nucleus sampling, where the