GraySoft
Projects Models Compare Cloud benchmarks FAQ Download guIDE →
Model Intelligence Sheet

memoravox/Fun-ASR-Nano-2512-gguf overview

Fun ASR Nano 2512: transcribe.cpp GGUF GGUF conversions of FunAudioLLM/Fun ASR Nano 2512 https://huggingface.co/FunAudioLLM/Fun ASR Nano 2512 for use with tran…

transcribe.cppggufasrspeech-to-textfun-asr-nanosense-voice-encoderqwen3audio-llmautomatic-speech-recognitionzhenjaarxiv:2509.12508base_model:FunAudioLLM/Fun-ASR-Nano-2512base_model:quantized:FunAudioLLM/Fun-ASR-Nano-2512license:otherregion:usconversational

Runs locally from ~850.0 MB disk (4 GB VRAM class GPUs with llama.cpp / guIDE).

Downloads
0
Likes
0
Pipeline
automatic-speech-recognition
Author

Repository Files & Downloads

1 GGUF files detected
Direct downloads for local inference
FileTypeQuantizationSizeLink
Fun-ASR-Nano-2512-Q8_0.ggufGGUFQ8_0850.0 MBDownload

Model Details

Model IDmemoravox/Fun-ASR-Nano-2512-gguf
Authormemoravox
Pipelineautomatic-speech-recognition
Licenseother
Base modelFunAudioLLM/Fun-ASR-Nano-2512
Last modified2026-07-14T00:35:12.000Z

Model README

---

license: other

base_model: FunAudioLLM/Fun-ASR-Nano-2512

base_model_relation: quantized

library_name: transcribe.cpp

pipeline_tag: automatic-speech-recognition

language:

- zh

- en

- ja

tags:

- gguf

- transcribe.cpp

- asr

- speech-to-text

- fun-asr-nano

- sense-voice-encoder

- qwen3

- audio-llm

transcribe_cpp:

wer_librispeech_test_clean:

bf16: 1.78

f16: 1.79

q8_0: 1.79

q6_k: 1.78

q5_k_m: 1.82

q4_k_m: 1.92

rtf_m4_max:

metal: 77.5

cpu: 27

rtf_ryzen_4750u:

vulkan: 10.5

cpu: 6.5

streaming: false

translate: false

lang_detect: false

timestamps: none

---

Fun-ASR-Nano-2512: transcribe.cpp GGUF

GGUF conversions of FunAudioLLM/Fun-ASR-Nano-2512 for use

with transcribe.cpp.

Ported from upstream commit

a7088d620f755dcdca575b63db184c3ad55b2865,

pinned 2026-05-06.

Validated against the FunASR reference at transcribe.cpp commit

f094d28

on 2026-05-06.

Offline speech-to-text in Chinese, English, and Japanese, plus 7 Chinese

dialects (Wu, Cantonese, Min, Hakka, Gan, Xiang, Jin) and 26 regional

Mandarin accents. ~800M trainable parameters wrapping a frozen

SenseVoiceEncoderSmall (50 SAN-M main blocks + 20 transformer blocks),

a 2-layer audio adaptor (512 → 1024), and a bundled Qwen3-0.6B LLM

(28 layers, 16/8 GQA, BF16) that produces the transcript autoregressively.

Takes a 16 kHz mono WAV and emits text. Not a streaming model, no

translation, no built-in long-form chunking, no timestamps. ITN

(inverse text normalization) is supported by the model and exposed

via the --itn CLI flag and transcribe_funasr_nano_params { use_itn }

in the library API.

Downloads

| Quantization | Download | Size | WER (LibriSpeech test-clean) |

| --- | --- | ---: | ---: |

| BF16 | Fun-ASR-Nano-2512-BF16.gguf | 1590 MB | 1.78% |

| F16 | Fun-ASR-Nano-2512-F16.gguf | 1590 MB | 1.79% |

| Q8_0 | Fun-ASR-Nano-2512-Q8_0.gguf | 850 MB | 1.79% |

| Q6_K | Fun-ASR-Nano-2512-Q6_K.gguf | 659 MB | 1.78% |

| Q5_K_M | Fun-ASR-Nano-2512-Q5_K_M.gguf | 602 MB | 1.82% |

| Q4_K_M | Fun-ASR-Nano-2512-Q4_K_M.gguf | 531 MB | 1.92% |

WER measured on the full LibriSpeech test-clean split (2620 utterances)

with greedy LLM decoding via the bundled Qwen3-0.6B head. Publisher

reports 1.76% on this split (model card "Open-Source Dataset

Performance" table). Our FunASR 1.3.1 reference run scores 1.79%

(95% CI [1.63%, 1.95%]), within bootstrap noise of the publisher's

number. transcribe.cpp's BF16 port matches that baseline within

-0.01 percentage-points. LibriSpeech is an English-only benchmark;

Chinese (AISHELL-1, WenetSpeech) and Japanese (CommonVoice JA) are

the recommended complementary checks.

Usage

Build transcribe.cpp from source:

git clone git@github.com:handy-computer/transcribe.cpp.git
cd transcribe.cpp
cmake -B build && cmake --build build

Run on a 16 kHz mono WAV:

build/bin/transcribe-cli \
  -m Fun-ASR-Nano-2512-Q8_0.gguf \
  input.wav

If your audio isn't already 16 kHz mono WAV, convert it first:

ffmpeg -i input.mp3 -ar 16000 -ac 1 output.wav

See the transcribe.cpp model page for performance

numbers, numerical validation, and reproduction steps.

License

Inherited from the base model: FunASR Model Open Source License Agreement v1.1. See the

upstream model card for full terms.

---

Original Model Card

> The section below is reproduced from

> FunAudioLLM/Fun-ASR-Nano-2512 at commit

> a7088d620f755dcdca575b63db184c3ad55b2865 for offline reference. The upstream card is the

> authoritative source.

Fun-ASR

简体中文」|「English」

Fun-ASR is an end-to-end speech recognition large model launched by Tongyi Lab. It is trained on tens of millions of hours of real speech data, possessing powerful contextual understanding capabilities and industry adaptability. It supports low-latency real-time transcription and covers 31 languages. It excels in vertical domains such as education and finance, accurately recognizing professional terminology and industry expressions, effectively addressing challenges like "hallucination" generation and language confusion, achieving "clear hearing, understanding meaning, and accurate writing."

<div align="center">

<img src="images/funasr-v2.png">

</div>

<div align="center">

<h4>

<a href="https://funaudiollm.github.io/funasr"> Homepage </a>

|<a href="#core-features"> Core Features </a>

|<a href="#performance-evaluation"> Performance Evaluation </a>

|<a href="#environment-setup"> Environment Setup </a>

|<a href="#usage-tutorial"> Usage Tutorial </a>

</h4>

Model Repository: modelscope, huggingface

Online Experience:

ModelScope Community Space, huggingface space

</div>

| Model Name | Task Details | Training Data | Parameters |

| :-------------------------------------------------------------------------------------------------------------------------------------------------------------: | :----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------: | :----------------------------: | :--------: |

| Fun-ASR-Nano <br> ( 🤗) | Speech recognition supports Chinese, English, and Japanese. Chinese includes support for 7 dialects (Wu, Cantonese, Min, Hakka, Gan, Xiang, Jin) and 26 regional accents (Henan, Shanxi, Hubei, Sichuan, Chongqing, Yunnan, Guizhou, Guangdong, Guangxi and more than 20 other regions). English and Japanese cover multiple regional accents. Additional features include lyric recognition and rap speech recognition. | Tens of millions of hours | 800M |

| Fun-ASR-MLT-Nano <br> ( 🤗) | Speech recognition supports Chinese, English, Cantonese, Japanese, Korean, Vietnamese, Indonesian, Thai, Malay, Filipino, Arabic, Hindi, Bulgarian, Croatian, Czech, Danish, Dutch, Estonian, Finnish, Greek, Hungarian, Irish, Latvian, Lithuanian, Maltese, Polish, Portuguese, Romanian, Slovak, Slovenian, Swedish, and 31 languages in total. | Hundreds of thousands of hours | 800M |

<a name="What's News"></a>

What's New 🔥

  • 2025/12: Fun-ASR-Nano-2512 is an end-to-end speech recognition large model trained on tens of millions of hours real speech data. It supports low-latency real-time transcription and covers 31 languages.
  • 2024/7: FunASR is a fundamental speech recognition toolkit that offers a variety of features, including speech recognition (ASR), Voice Activity Detection (VAD), Punctuation Restoration, Language Models, Speaker Verification, Speaker Diarization and multi-talker ASR.

Core Features 🎯

Fun-ASR focuses on high-precision speech recognition, multi-language support, and industry customization capabilities

  • Far-field High-noise Recognition: Deeply optimized for far-distance sound pickup and high-noise scenarios (such as conference rooms, in-vehicle environments, industrial sites, etc.), improving recognition accuracy to 93%.
  • Chinese Dialects and Regional Accents:

- Supports 7 major dialects: Wu, Cantonese, Min, Hakka, Gan, Xiang, Jin

- Covers 26 regional accents: including Henan, Shaanxi, Hubei, Sichuan, Chongqing, Yunnan, Guizhou, Guangdong, Guangxi and more than 20 other regions

  • Multi-language Free Speech: Supports recognition of 31 languages, with focused optimization on East and Southeast Asian languages, supporting free language switching and mixed recognition.
  • Music Background Lyric Recognition: Enhanced speech recognition performance under music background interference, supporting accurate recognition of lyric content in songs.

Environment Setup 🐍

git clone https://github.com/FunAudioLLM/Fun-ASR.git
cd Fun-ASR
pip install -r requirements.txt

<a name="usage-tutorial"></a>

TODO

  • [ ] Support returning timestamps
  • [ ] Support speaker diarization
  • [ ] Support model training

Usage 🛠️

Inference

Using funasr for inference

from funasr import AutoModel


def main():
    model_dir = "FunAudioLLM/Fun-ASR-Nano-2512"
    model = AutoModel(
        model=model_dir,
        trust_remote_code=True,
        remote_code="./model.py",
        device="cuda:0",
    )

    wav_path = f"{model.model_path}/example/zh.mp3"
    res = model.generate(
        input=[wav_path],
        cache={},
        batch_size=1,
        hotwords=["开放时间"],
        # 中文、英文、日文 for Fun-ASR-Nano-2512
        # 中文、英文、粤语、日文、韩文、越南语、印尼语、泰语、马来语、菲律宾语、阿拉伯语、
        # 印地语、保加利亚语、克罗地亚语、捷克语、丹麦语、荷兰语、爱沙尼亚语、芬兰语、希腊语、
        # 匈牙利语、爱尔兰语、拉脱维亚语、立陶宛语、马耳他语、波兰语、葡萄牙语、罗马尼亚语、
        # 斯洛伐克语、斯洛文尼亚语、瑞典语 for Fun-ASR-MLT-Nano-2512
        language="中文",
        itn=True, # or False
    )
    text = res[0]["text"]
    print(text)

    model = AutoModel(
        model=model_dir,
        trust_remote_code=True,
        vad_model="fsmn-vad",
        vad_kwargs={"max_single_segment_time": 30000},
        remote_code="./model.py",
        device="cuda:0",
    )
    res = model.generate(input=[wav_path], cache={}, batch_size=1)
    text = res[0]["text"]
    print(text)


if __name__ == "__main__":
    main()

Direct Inference

from model import FunASRNano


def main():
    model_dir = "FunAudioLLM/Fun-ASR-Nano-2512"
    m, kwargs = FunASRNano.from_pretrained(model=model_dir, device="cuda:0")
    m.eval()

    wav_path = f"{kwargs['model_path']}/example/zh.mp3"
    res = m.inference(data_in=[wav_path], **kwargs)
    text = res[0][0]["text"]
    print(text)


if __name__ == "__main__":
    main()

<details><summary> Parameter Description (click to expand) </summary>

  • model_dir: Model name or local disk model path.
  • trust_remote_code: Whether to trust remote code for loading custom model implementations.
  • remote_code: Specify the location of specific model code (e.g., model.py in the current directory), supporting both absolute and relative paths.
  • device: Specify the device to use, such as "cuda:0" or "cpu".

</details>

Performance 📝

We evaluated Fun-ASR against other state-of-the-art models on open-source benchmarks, Chinese dialect datasets, and industry-specific test sets. The results demonstrate that Fun-ASR achieves superior performance across various scenarios.

1. Open-Source Dataset Performance (WER %)

| Test set | GLM-ASR-nano | GLM-ASR-nano\ | Whisper-large-v3 | Seed-ASR | Seed-ASR\ | Kimi-Audio | Step-Audio2 | FireRed-ASR | Fun-ASR-nano | Fun-ASR |

| :------------------ | :----------: | :------------: | :--------------: | :------: | :--------: | :--------: | :---------: | :---------: | :----------: | :-----: |

| Model Size | 1.5B | 1.5B | 1.6B | - | - | - | - | 1.1B | 0.8B | 7.7B |

| OpenSource | ✅ | ✅ | ✅ | ❌ | ❌ | ✅ | ✅ | ✅ | ✅ | ❌ |

| AIShell1 | 1.81 | 2.17 | 4.72 | 0.68 | 1.63 | 0.71 | 0.63 | 0.54 | 1.80 | 1.22 |

| AIShell2 | - | 3.47 | 4.68 | 2.27 | 2.76 | 2.86 | 2.10 | 2.58 | 2.75 | 2.39 |

| Fleurs-zh | - | 3.65 | 5.18 | 3.43 | 3.23 | 3.11 | 2.68 | 4.81 | 2.56 | 2.53 |

| Fleurs-en | 5.78 | 6.95 | 6.23 | 9.39 | 9.39 | 6.99 | 3.03 | 10.79 | 5.96 | 4.74 |

| Librispeech-clean | 2.00 | 2.17 | 1.86 | 1.58 | 2.8 | 1.32 | 1.17 | 1.84 | 1.76 | 1.51 |

| Librispeech-other | 4.19 | 4.43 | 3.43 | 2.84 | 5.69 | 2.63 | 2.42 | 4.52 | 4.33 | 3.03 |

| WenetSpeech Meeting | 6.73 | 8.21 | 18.39 | 5.69 | 7.07 | 6.24 | 4.75 | 4.95 | 6.60 | 6.17 |

| WenetSpeech Net | - | 6.33 | 11.89 | 4.66 | 4.84 | 6.45 | 4.67 | 4.94 | 6.01 | 5.46 |

> _Note: Seed-ASR\ results are evaluated using the official API on volcengine; GLM-ASR-nano\ results are evaluated using the open-source checkpoint._

2. Industry Dataset Performance (WER %)

| Test set | GLM-ASR-Nano | Whisper-large-v3 | Seed-ASR | FireRed-ASR | Kimi-Audio | Paraformer v2 | Fun-ASR-nano | Fun-ASR |

| :----------------- | :----------: | :--------------: | :-------: | :---------: | :--------: | :-----------: | :----------: | :-------: |

| Model Size | 1.5B | 1.6B | - | 1.1B | 8B | 0.2B | 0.8B | 7.7B |

| OpenSource | ✅ | ✅ | ❌ | ✅ | ✅ | ✅ | ✅ | ❌ |

| Nearfield | 16.95 | 16.58 | 7.20 | 10.10 | 9.02 | 8.11 | 7.79 | 6.31 |

| Farfield | 9.44 | 22.21 | 4.59 | 7.49 | 10.95 | 9.55 | 5.79 | 4.34 |

| Complex Background | 23.79 | 32.57 | 12.90 | 15.56 | 15.56 | 15.19 | 14.59 | 11.45 |

| English General | 16.47 | 18.56 | 15.65 | 21.62 | 18.12 | 19.48 | 15.28 | 13.73 |

| Opensource | 4.67 | 7.05 | 3.83 | 5.31 | 3.79 | 6.23 | 4.22 | 3.38 |

| Dialect | 54.21 | 66.14 | 29.45 | 52.82 | 71.94 | 41.16 | 28.18 | 15.21 |

| Accent | 19.78 | 36.03 | 10.23 | 14.05 | 27.20 | 17.80 | 12.90 | 10.31 |

| Lyrics | 46.56 | 54.82 | 30.26 | 42.87 | 65.18 | 50.14 | 30.85 | 21.00 |

| Hiphop | 43.32 | 46.56 | 29.46 | 33.88 | 57.25 | 43.79 | 30.87 | 28.58 |

| Average | 26.13 | 33.39 | 15.95 | 22.63 | 31.00 | 23.49 | 16.72 | 12.70 |

<div align="center">

<img src="images/compare_en.png" width="800" />

</div>

Citations

@article{an2025fun,
  title={Fun-ASR Technical Report},
  author={An, Keyu and Chen, Yanni and Deng, Chong and Gao, Changfeng and Gao, Zhifu and Gong, Bo and Li, Xiangang and Li, Yabin and Lv, Xiang and Ji, Yunjie and others},
  journal={arXiv preprint arXiv:2509.12508},
  year={2025}
}

Run memoravox/Fun-ASR-Nano-2512-gguf with guIDE

Download guIDE — the AI-native code editor with local LLM inference and 69 built-in tools.

Download guIDE → · Browse 524k+ models · Compare models

Source: Hugging Face · Compare models