Azure/azure-sdk-for-java com.azure+azure-ai-voicelive_1.1.0

on GitHub

1.1.0 (2026-08-03)

Features Added

• Streaming input text into a conversation item:
  • New ClientEventInputTextDelta (input_text.delta) and ClientEventInputTextDone (input_text.done), each with optional contentIndex
  • New ClientEventType.INPUT_TEXT_DELTA and ClientEventType.INPUT_TEXT_DONE values
• Echo cancellation reference source: AudioEchoCancellation is now @Fluent and exposes referenceSource (new EchoCancellationReferenceSource enum with SERVER / CLIENT) and channels for stereo input where channel 1 is the client-supplied echo reference
• Azure realtime native voices: new AzureRealtimeNativeVoice (type azure-realtime-native) and AzureRealtimeNativeVoiceName expandable enum (AARTI, ANDREW, AVA, DENISE, DIYA, ELSA, FLORIAN, FRANCISCA, MEERA, XIAOXIAO, YUNXI, XIMENA) for use with the azure-realtime model
• Parallel tool calls: VoiceLiveSessionOptions.setParallelToolCalls(Boolean) / isParallelToolCalls() (and matching accessors on VoiceLiveSessionResponse) to control whether the model may invoke tools in parallel
• Hosted agent invocation passthrough:
  • ResponseCreateParams.setInvokeInput(Map<String, BinaryData>) / getInvokeInput() to attach input data for a hosted agent invocation on a single response request
  • New ServerEventResponseInvocationDelta (response.invocation.delta) and ServerEventType.RESPONSE_INVOCATION_DELTA pass through non-speech SSE events from the hosted agent
• Session expiration: VoiceLiveSessionResponse.getExpiresAt() / setExpiresAt(OffsetDateTime) expose the server-assigned session expiration time (serialized on the wire as seconds since epoch)

Other Changes

• Regenerated against the 2026-07-15 (GA) VoiceLive TypeSpec.
• Added service API version V2026_07_15 and made it the default used by VoiceLiveClientBuilder. VoiceLiveServiceVersion.getLatest() now returns V2026_07_15. The previous default V2026_04_10 remains available for callers that pin to it.
• The following APIs introduced in 1.1.0-beta.1 are not part of this GA release: WebRTC SDP negotiation (ClientEventRtcCallSdpCreate, ServerEventRtcCallSdpCreated, ServerEventRtcCallError, RtcCallErrorDetails, and the rtc.call.* event-type values), output audio buffer playback lifecycle events (ServerEventOutputAudioBufferStarted, ServerEventOutputAudioBufferStopped, and the output_audio_buffer.started / output_audio_buffer.stopped ServerEventType values), and smart audio-based end-of-turn detection (SmartEndOfTurnDetection and the EouDetectionModel.SMART_END_OF_TURN_DETECTION value).