hviske

Results on CoRal v3 full test sets

Evaluated on the complete test splits (17,560 samples). Two normalization conventions:

• raw: jiwer on un-normalized references and hypotheses
• strict: lowercase + punctuation strip + Danish digit-to-word (num2words(lang="da")) — the apples-to-apples normalization for comparing against published Whisper-style numbers

Greedy decoding (num_beams=1)

Beam search (num_beams=5, length_penalty=1.0)

Beam search costs ~75% more inference time but lowers avg WER by 0.4 pp.

Versus other Danish ASR models on CoRal v3 (CER)

The CoRal team publishes CER numbers on the same test splits. hviske-v5.3 numbers are evaluated on the full test sets. Other entries reproduced from the roest-v3-whisper-1.5b model card.

Conversation split

Read-aloud split

The CoRal team's published numbers do not specify the normalization used; both raw and strict CER are shown for hviske-v5.3 to make the comparison fair. capacit-ai/saga was evaluated with the same methodology used here (full test splits via greedy vllm serve + /v1/audio/transcriptions); raw CER is 8.26% (read_aloud) and 17.49% (conversation). ElevenLabs scribe_v2 was evaluated via the public /v1/speech-to-text API on the same full test sets (n=17,560); strict WER is 18.62% (read_aloud) and 31.38% (conversation). OpenAI gpt-4o-transcribe was evaluated via the public /v1/audio/transcriptions API on the same full test sets; strict WER is 26.34% (read_aloud) and 55.24% (conversation).

Inference speed

On a single NVIDIA RTX 3090, hviske-v5.3 reaches RTFx ≈ 425 — i.e. it transcribes audio about 425× faster than real time. 60 minutes of audio is processed in ≈ 8.5 seconds.

Installation

pip install "transformers==4.57.6" torch soundfile librosa huggingface_hub sentencepiece protobuf
pip install datasets  # only needed for the streaming examples below

Usage

Load the model with AutoModelForSpeechSeq2Seq and trust_remote_code=True. The model exposes both a high-level model.transcribe(...) helper and the standard model.generate(...) interface.

1. Quick start — single file

import torch, numpy as np, soundfile as sf
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq

processor = AutoProcessor.from_pretrained("syvai/hviske-v5.3", trust_remote_code=True)
model = AutoModelForSpeechSeq2Seq.from_pretrained(
    "syvai/hviske-v5.3", trust_remote_code=True, dtype=torch.bfloat16
).to("cuda").eval()

audio, sr = sf.read("your_audio.wav")
audio = np.asarray(audio, dtype=np.float32)

hyp = model.transcribe(
    processor=processor,
    language="da",
    audio_array