Constructor Detail

Method Detail

• sessionUpdate

   final Optional<RealtimeTranslationSessionUpdateEvent> sessionUpdate()

  Send this event to update the translation session configuration. Translation sessions support updates to audio.output.language, audio.input.transcription, and audio.input.noise_reduction.

• sessionInputAudioBufferAppend

   final Optional<RealtimeTranslationInputAudioBufferAppendEvent> sessionInputAudioBufferAppend()

  Send this event to append audio bytes to the translation session input audio buffer.

  WebSocket translation sessions accept base64-encoded 24 kHz PCM16 mono little-endian raw audio bytes. Unsupported websocket audio formats return a validation error because lower-quality audio materially degrades translation quality.

  Translation consumes 200 ms engine frames. For best realtime behavior, append audio in 200 ms chunks. If a chunk is shorter, the server buffers it until it has enough audio for one frame. If a chunk is longer, the server splits it into 200 ms frames and enqueues them back-to-back.

  Keep appending silence while the session is active. If a client stops sending audio and later resumes, model time treats the resumed audio as contiguous with the previous audio rather than as a real-world pause.

• sessionClose

   final Optional<RealtimeTranslationSessionCloseEvent> sessionClose()

  Gracefully close the realtime translation session. The server flushes pending input audio and emits any remaining translated output before closing the session.

• isSessionUpdate

   final Boolean isSessionUpdate()

• isSessionInputAudioBufferAppend

   final Boolean isSessionInputAudioBufferAppend()

• isSessionClose

   final Boolean isSessionClose()

• asSessionUpdate

   final RealtimeTranslationSessionUpdateEvent asSessionUpdate()

  Send this event to update the translation session configuration. Translation sessions support updates to audio.output.language, audio.input.transcription, and audio.input.noise_reduction.

• asSessionInputAudioBufferAppend

   final RealtimeTranslationInputAudioBufferAppendEvent asSessionInputAudioBufferAppend()

  Send this event to append audio bytes to the translation session input audio buffer.

  WebSocket translation sessions accept base64-encoded 24 kHz PCM16 mono little-endian raw audio bytes. Unsupported websocket audio formats return a validation error because lower-quality audio materially degrades translation quality.

  Translation consumes 200 ms engine frames. For best realtime behavior, append audio in 200 ms chunks. If a chunk is shorter, the server buffers it until it has enough audio for one frame. If a chunk is longer, the server splits it into 200 ms frames and enqueues them back-to-back.

  Keep appending silence while the session is active. If a client stops sending audio and later resumes, model time treats the resumed audio as contiguous with the previous audio rather than as a real-world pause.

• asSessionClose

   final RealtimeTranslationSessionCloseEvent asSessionClose()

  Gracefully close the realtime translation session. The server flushes pending input audio and emits any remaining translated output before closing the session.

• _json

   final Optional<JsonValue> _json()

• accept

   final <T extends Any> T accept(RealtimeTranslationClientEvent.Visitor<T> visitor)

  Maps this instance's current variant to a value of type T using the given visitor.

  Note that this method is not forwards compatible with new variants from the API, unless visitor overrides Visitor.unknown. To handle variants not known to this version of the SDK gracefully, consider overriding Visitor.unknown:

  import com.openai.core.JsonValue;
  import java.util.Optional;
  
  Optional<String> result = realtimeTranslationClientEvent.accept(new RealtimeTranslationClientEvent.Visitor<Optional<String>>() {
      @Override
      public Optional<String> visitSessionUpdate(RealtimeTranslationSessionUpdateEvent sessionUpdate) {
          return Optional.of(sessionUpdate.toString());
      }
  
      // ...
  
      @Override
      public Optional<String> unknown(JsonValue json) {
          // Or inspect the `json`.
          return Optional.empty();
      }
  });

• validate

   final RealtimeTranslationClientEvent validate()

  Validates that the types of all values in this object match their expected types recursively.

  This method is not forwards compatible with new types from the API for existing fields.

• isValid

   final Boolean isValid()

• equals

   Boolean equals(Object other)

• hashCode

   Integer hashCode()

• toString

   String toString()

• ofSessionUpdate

   final static RealtimeTranslationClientEvent ofSessionUpdate(RealtimeTranslationSessionUpdateEvent sessionUpdate)

  Send this event to update the translation session configuration. Translation sessions support updates to audio.output.language, audio.input.transcription, and audio.input.noise_reduction.

• ofSessionInputAudioBufferAppend

   final static RealtimeTranslationClientEvent ofSessionInputAudioBufferAppend(RealtimeTranslationInputAudioBufferAppendEvent sessionInputAudioBufferAppend)

  Send this event to append audio bytes to the translation session input audio buffer.

  WebSocket translation sessions accept base64-encoded 24 kHz PCM16 mono little-endian raw audio bytes. Unsupported websocket audio formats return a validation error because lower-quality audio materially degrades translation quality.

  Translation consumes 200 ms engine frames. For best realtime behavior, append audio in 200 ms chunks. If a chunk is shorter, the server buffers it until it has enough audio for one frame. If a chunk is longer, the server splits it into 200 ms frames and enqueues them back-to-back.

  Keep appending silence while the session is active. If a client stops sending audio and later resumes, model time treats the resumed audio as contiguous with the previous audio rather than as a real-world pause.

• ofSessionClose

   final static RealtimeTranslationClientEvent ofSessionClose(RealtimeTranslationSessionCloseEvent sessionClose)

  Gracefully close the realtime translation session. The server flushes pending input audio and emits any remaining translated output before closing the session.