In the Realtime with WebRTC, I noticed that a server event is triggered with the type “output_audio_buffer.stopped”. It occures when the assistant stops speaking. Why is it not documented in the API reference? Is it scheduled to remove it?
In the Realtime with WebRTC, I noticed that a server event is triggered with the type “output_audio_buffer.stopped”. It occures when the assistant stops speaking. Why is it not documented in the API reference? Is it scheduled to remove it?