Model reference · open weights
ARK-ASR is an open-weight audio or speech model from Audio8. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.
Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.
What it is
| Released by | Audio8 |
|---|---|
| Type | Audio & music |
| Task | Speech→text |
| Parameters (lead) | 4.1B |
| Context | 32k tokens |
| Runs with | transformers |
| Released | 2026-06-22 |
| Popularity | 16k downloads / month |
| Licence | Open weights |
About
TL;DR ARK-ASR-3B is a multilingual automatic speech recognition model. It achieves current state-of-the-art results on the Hugging Face Open ASR Leaderboard English short-form benchmark, with an average WER of 5.04% and RTFx of 490.98 across AMI, Earnings22, GigaSpeech, LibriSpeech, SPGISpeech, and VoxPopuli. The accompanying training, inference, and evaluation code is available at AutoArk/open-audio-opd.
ARK-ASR-3B is a 3B-scale audio-capable autoregressive Transformers model for automatic speech recognition.
It combines a Whisper-style audio encoder, an MLP adapter, and a Qwen decoder with custom arkasr remote code.
ARK-ASR currently supports Chinese, English, German, Japanese, French, Korean, Spanish, Polish, Italian, Romanian, Hungarian, Czech, Dutch, Finnish, Croatian, Slovak, Slovene, Estonian, and Lithuanian ASR.
Chinese, English, German, Japanese, French, Korean, Spanish, Polish, Italian, Romanian, Hungarian, Czech, Dutch, Finnish, Croatian, Slovak, Slovene, Estonian, and Lithuanian.
arkasr remote codesafetensorsscripts/infer/ark_asr_transformers.pyscripts/vllm/ark_asr_vllmThe model should be loaded with trust_remote_code=True. The official inference script handles the processor, tokenizer, audio prompt format, generation cleanup, and ASR token filtering.
The following results are from the Hugging Face Open ASR Leaderboard. Lower WER is better. ARK-ASR-3B reaches the current state of the art on this English short-form benchmark.
| Model | AMI | Earnings22 | GigaSpeech | LS Clean | LS Other | SPGISpeech | VoxPopuli | Avg |
|---|---|---|---|---|---|---|---|---|
| ARK-ASR-3B | 8.79% | 8.23% | 6.98% | 1.03% | 2.35% | 2.46% | 5.47% | 5.04% |
| ARK-ASR-0.6B | 10.02% | 9.77% | 8.00% | 1.53% | 3.51% | 2.63% | 6.31% | 5.97% |
| Model | AISHELL-1 | WenetSpeech test meeting | WenetSpeech test-net |
|---|---|---|---|
| ARK-ASR-3B | 1.80% | 4.97% | 4.58% |
| ARK-ASR-0.6B | 2.02% | 5.92% | 4.96% |
Run ASR inference with Hugging Face Transformers:
import torch
from transformers import AutoModelForCausalLM, AutoProcessor, AutoTokenizer
model_path = "AutoArk-AI/ARK-ASR-3B"
audio_path = "assets/libai.wav"
device = "cuda" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.bfloat16 if device == "cuda" else torch.float32
processor = AutoProcessor.from_pretrained(model_path, trust_remote_code=True)
tokenizer = AutoTokenizer.from_pretrained(model_path, trust_remote_code=True)
model = AutoModelForCausalLM.from_pretrained(
model_path,
trust_remote_code=True,
torch_dtype=torch_dtype,
attn_implementation="sdpa",
).to(device)
model.eval()
def build_bad_words_ids(tokenizer):
eos_ids = tokenizer.eos_token_id
keep_ids = {eos_ids} if isinstance(eos_ids, int) else set(eos_ids or [])
bad_ids = set(tokenizer.all_special_ids) - keep_ids
bad_ids.update(
token_id
for token, token_id in tokenizer.get_added_vocab().items()
if token.startswith("") and token_id not in keep_ids
)
return [[token_id] for token_id in sorted(bad_ids)]
conversation = [
{
"role": "user",
"content": [
{"type": "audio", "path": audio_path},
{"type": "text", "text": "Please transcribe this audio."},
],
}
]
inputs = processor.apply_chat_template(
conversation,
add_generation_prompt=True,
return_tensors="pt",
sampling_rate=16000,
audio_padding="longest",
text_kwargs={"padding": "longest"},
audio_max_length=30 * 16000,
)
inputs = inputs.to(device)
if "audios" in inputs:
inputs["audios"] = inputs["audios"].to(dtype=torch_dtype)
bad_words_ids = build_bad_words_ids(tokenizer)
with torch.inference_mode():
outputs = model.generate(
**inputs,
do_sample=False,
max_new_tokens=256,
pad_token_id=tokenizer.pad_token_id,
eos_token_id=tokenizer.eos_token_id,
bad_words_ids=bad_words_ids,
)
decoded_outputs = tokenizer.batch_decode(
outputs[:, inputs.input_ids.shape[1] :],
skip_special_tokens=True,
)
print(decoded_outputs)
For batch JSONL inference, use the open-source inference code:
git clone https://github.com/AutoArk/open-audio-opd
cd open-audio-opd
pip install -e .
The input JSONL should contain one ASR sample per line:
{"audio":"/path/to/audio.wav","text":"","task":"asr","begin_time":-1,"end_time":-1}
python scripts/infer/ark_asr_transformers.py \
--input /path/to/input.jsonl \
--output runs/infer/predictions.jsonl \
--model_path AutoArk-AI/ARK-ASR-3B \
--processor_path AutoArk-AI/ARK-ASR-3B \
--batch_size 40 \
--dtype bfloat16 \
--attn_impl sdpa
The output JSONL preserves input metadata and adds:
pred_text: cleaned prediction text for downstream evaluationpred_text_raw: raw decoded generation before cleanupARK-ASR can also be deployed as a vLLM-backed online ASR service with the
adapter in
[scripts/vllm/ark_asr_vllm](https://github.com/AutoArk/open-audio-op
From the published model card. Full card on the HuggingFace links in the sidebar.
Using it via the API
Once AxForge deploys ark-asr for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (ark-asr below is illustrative; you get the exact model name on deployment.)
$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \ -H "Authorization: Bearer $AXFORGE_API_KEY" \ -F model="ark-asr" -F file=@audio.mp3
Create an account — your API key is available in the console. 3M free tokens every 30 days with every new account.