Speech to Text

client.asr().transcribe(...) uploads a complete audio file to luchs-1 and returns the transcript. See Speech to Text for the full wire contract, supported language, audio limits, and the WebSocket streaming protocol — the SDK does not yet wrap streaming.

Transcribe a file

import com.kugelaudio.sdk.KugelAudio;
import com.kugelaudio.sdk.KugelAudioOptions;
import com.kugelaudio.sdk.TranscriptionResponse;

import java.nio.file.Files;
import java.nio.file.Path;

KugelAudio client = new KugelAudio(
    KugelAudioOptions.builder("your_api_key").build()
);

byte[] audio = Files.readAllBytes(Path.of("speech.wav"));
TranscriptionResponse result = client.asr().transcribe(audio);

System.out.println(result.getText());          // OpenAI-compatible alias
System.out.println(result.getTranscript());    // KugelAudio's own field
System.out.println(result.getModel());         // "luchs-1"
System.out.println(result.getModelRevision()); // immutable checkpoint revision

client.close();

The single-argument transcribe(audio) overload defaults to ASRResource.MODEL_ID ("luchs-1"). The five-argument overload lets you set filename, contentType and language explicitly, but model must still equal ASRResource.MODEL_ID — any other value throws ValidationException before any HTTP call:

client.asr().transcribe(audio, "call.wav", "audio/wav", "en", "qwen3-asr");
// throws ValidationException: Unsupported ASR model; use luchs-1

language is forwarded but currently has no effect on recognition — see Supported language.

Migrating from `qwen3-asr`

Passing "qwen3-asr" as the model argument now throws ValidationException before any network call. Use ASRResource.MODEL_ID, or the literal "luchs-1". Anything reading the response model field must also account for it now always being "luchs-1"; use getModelRevision() if you need the underlying artifact identity. This changed in the next major release of the Java SDK.

Streaming

The Java SDK does not currently include a WebSocket streaming client for speech-to-text; StreamingTranscriptionResult exists only so callers who open their own WebSocket session can deserialize frames. Use the raw WebSocket protocol in Speech to Text directly for live transcription.

Get started

Build a voice agent

Generate audio

Voices

Deploy

SDK Reference

API Reference

Transcribe a file

Migrating from `qwen3-asr`

Streaming

​Transcribe a file

​Migrating from qwen3-asr

​Streaming

Transcribe a file

Migrating from `qwen3-asr`

Streaming