Fun-ASR is a family of end-to-end speech recognition models from Tongyi Lab. Checkpoint capabilities are distinct: Fun-ASR-Nano-2512 is trained on tens of millions of hours of speech and supports Chinese, English, Japanese, and Chinese dialects and accents; Fun-ASR-MLT-Nano-2512 is an 800M multilingual checkpoint trained on hundreds of thousands of hours and supports 31 languages. Both checkpoints integrate with FunASR for inference and serving.
Model repositories: Fun-ASR-Nano (ModelScope, Hugging Face) · Fun-ASR-MLT-Nano (ModelScope, Hugging Face)
Online Experience: ModelScope Community Space, huggingface space
Runnable examples cover quickstart inference, direct inference, speaker diarization, vLLM batch inference, and the streaming SDK.
| Model Name | Task Details | Training Data | Parameters |
|---|---|---|---|
| Fun-ASR-Nano (⭐ 🤗) |
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 (⭐ 🤗) |
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, and Swedish: 31 languages in total. | Hundreds of thousands of hours | 800M |
- 2026/06: Fun-ASR-Nano on llama.cpp / GGUF — run it on CPU/edge as a single self-contained binary (whisper.cpp-style), built-in VAD, no Python at runtime. Quantized models down to ~484 MB. runtime/llama.cpp/ · Releases · Nano GGUF · FSMN-VAD GGUF
- 2026/05: vLLM Inference Engine — native high-throughput batch (3-5x faster) + WebSocket real-time streaming service. See vLLM Guide.
- 2026/05: The FunASR pipeline can combine Fun-ASR-Nano with separate FSMN-VAD, CAM++, and punctuation models to produce per-sentence speaker labels. Diarization is not a native output of the Nano checkpoint. Requires installing FunASR from source:
pip install git+https://github.com/modelscope/FunASR.git - 2025/12: Fun-ASR-Nano-2512 was released for Chinese, English, Japanese, and Chinese dialects and accents. For 31-language recognition, use the separate Fun-ASR-MLT-Nano-2512 checkpoint.
- 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.
Fun-ASR focuses on high-precision speech recognition, checkpoint-specific multilingual 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
- Checkpoint-specific language coverage: Fun-ASR-Nano supports Chinese, English, Japanese, and Chinese dialects and accents. Fun-ASR-MLT-Nano supports 31 languages, with emphasis on East and Southeast Asian languages.
- Music Background Lyric Recognition: Enhanced speech recognition performance under music background interference, supporting accurate recognition of lyric content in songs.
git clone https://github.com/FunAudioLLM/Fun-ASR.git
cd Fun-ASR
pip install -r requirements.txt- Reliable checkpoint-native timestamps
The released Fun-ASR-Nano
model.ptcheckpoint does not include trainedctc_decoder.*/ctc.*weights. Any timestamp output is therefore not reliable. For accurate character-level timestamps, use Paraformer, for exampleAutoModel(model="paraformer-zh", vad_model="fsmn-vad", ...). See issue #106. - Checkpoint-native speaker diarization
Fun-ASR-Nano and Fun-ASR-MLT-Nano do not emit speaker labels by themselves. Compose them in FunASR with the separate
fsmn-vadandcam++models, as shown below. - Model training
Run Fun-ASR-Nano as a single self-contained binary — like whisper.cpp but for FunASR, with strong Chinese accuracy. Built-in FSMN-VAD, no Python at runtime.
bash runtime/llama.cpp/download-funasr-model.sh nano ./gguf
llama-funasr-cli --enc ./gguf/funasr-encoder-f16.gguf -m ./gguf/qwen3-0.6b-q8_0.gguf -a audio.wav --vad ./gguf/fsmn-vad.gguffsmn-vad.gguf is hosted in the shared FunAudioLLM/fsmn-vad-GGUF repo, not inside the Nano GGUF repo. The nano downloader above fetches it automatically; to fetch only VAD from the Hugging Face UI/CLI, use:
hf download FunAudioLLM/fsmn-vad-GGUF --include "*.gguf" --local-dir ./ggufPrebuilt binaries: Releases · Download & quickstart: funasr.com/llama-cpp · GGUF: Nano encoder/LLM · FSMN-VAD · Docs & benchmarks: runtime/llama.cpp/
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",
# hub:download models from ms (for ModelScope) or hf (for Hugging Face).
hub="hf"
)
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()When transcribing long audio or many files on the funasr (PyTorch) path, pass
batch_size_s to batch the VAD segments through the LLM decoder together. This
greatly improves GPU utilization:
res = model.generate(
input=[wav_path],
cache={},
language="中文",
itn=True,
batch_size_s=120, # batch VAD segments up to ~120s of audio per LLM call
)On Fun-ASR-Nano-2512 (184 Chinese files / 11,539 s, single H100) this is about 1.6x faster than the default per-segment decoding (RTFx 19.8 -> 31.8) with no loss in accuracy. For the highest throughput, use the vLLM path below.
This example is a composed FunASR pipeline: FSMN-VAD segments the audio,
Fun-ASR-Nano transcribes it, CAM++ assigns speaker labels, and CT-Punc restores
punctuation. The start and end values are VAD segment boundaries, not
reliable checkpoint-native character timestamps.
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",
vad_model="fsmn-vad",
vad_kwargs={"max_single_segment_time": 30000},
spk_model="cam++",
punc_model="ct-punc",
device="cuda:0",
hub="hf",
)
wav_path = f"{model.model_path}/example/zh.mp3"
res = model.generate(input=[wav_path], cache={}, batch_size=1, language="中文")
# Per-sentence results with speaker labels
for sent in res[0]["sentence_info"]:
print(f"Speaker {sent['spk']}: [{sent['start']}ms - {sent['end']}ms] {sent['sentence']}")
if __name__ == "__main__":
main()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()Parameter Description (click to expand)
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.pyin the current directory), supporting both absolute and relative paths.device: Specify the device to use, such as "cuda:0" or "cpu".
Fun-ASR natively integrates the vLLM engine for high-throughput batch inference and production-grade real-time streaming service.
Full guide: docs/vllm_guide.md | API docs: modelscope.github.io/FunASR/vllm.html
| Mode | Use Case | Entry |
|---|---|---|
| Offline Batch | Large-scale transcription | AutoModelVLLM |
| Streaming SDK | Real-time subtitles | FunASRNanoStreamingVLLM |
| WebSocket Service | Production deployment | serve_realtime_ws.py |
from funasr.auto.auto_model_vllm import AutoModelVLLM
model = AutoModelVLLM(
model="FunAudioLLM/Fun-ASR-Nano-2512",
tensor_parallel_size=2, # Multi-GPU
gpu_memory_utilization=0.8,
)
results = model.generate(
["audio1.wav", "audio2.wav", "audio3.wav"],
language="中文",
hotwords=["张三", "北京"],
)
for r in results:
print(f"[{r['key']}] {r['text']}")Long audio:
AutoModelVLLMdecodes each input in a single pass, so a long recording (e.g. a multi-minute meeting) can be truncated — pre-segment it with VAD and pass the segments, or use the high-levelAutoModel(model=..., vad_model="fsmn-vad"), which segments long audio automatically.
# Start server (with dynamic VAD + speaker diarization)
python serve_realtime_ws.py --port 10095 --language 中文 --tensor-parallel-size 2
# Browser client
open client_mic.html
# Python client
python client_python.py --server ws://localhost:10095 --micWebSocket Protocol:
Client: "START" → Server: {"event":"started"}
Client: [audio bytes] → Server: {"sentences":[...], "partial":"..."}
Client: "STOP" → Server: {"sentences":[...], "is_final":true}
from funasr.models.fun_asr_nano.inference_vllm_streaming import FunASRNanoStreamingVLLM
engine = FunASRNanoStreamingVLLM.from_pretrained(
model="FunAudioLLM/Fun-ASR-Nano-2512", chunk_ms=720
)
for result in engine.streaming_generate("audio.wav", language="中文"):
print(f"[{result['audio_duration_ms']:.0f}ms] {result['fixed_text']}")| Method | Time (184 files, 11,541s) | RTFx | CER |
|---|---|---|---|
| PyTorch native | 550s | 21x | 8.06% |
| vLLM (ours) | 34s | 340x | 8.20% |
16x faster than PyTorch with nearly identical accuracy (CER diff < 0.2%)
pip install funasr>=1.3.3 vllm>=0.12.0Please refer to docs/finetune.md
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.
| 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.
| 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 |
- Fun-ASR-vllm (@yuekaizhang) — a community vLLM implementation of Fun-ASR (~50% speedup over PyTorch), with batch inference and an NVIDIA Triton Inference Server integration for high-concurrency production deployment. See #34.
Native vLLM support is also built in — see vLLM High-Throughput Inference 🚀 above for the
AutoModelVLLMbatch engine, the streaming SDK, and the WebSocket service.
Fun-ASR-Nano is part of the FunAudioLLM family:
| Project | Description | Stars |
|---|---|---|
| FunASR | Industrial speech recognition toolkit — VAD, ASR, punctuation, diarization | |
| SenseVoice | Multilingual speech understanding — ASR + emotion + audio events | |
| CosyVoice | Natural speech generation — multi-language, zero-shot cloning | |
| FunClip | AI-powered video clipping with speech recognition |
- Source code in this repository is licensed under the Apache License 2.0.
- Model weights are distributed separately and follow the license metadata on their model cards. The official Fun-ASR-Nano and Fun-ASR-MLT-Nano cards currently list Apache-2.0; review the card for the specific artifact you download.
@misc{an2025funasrtechnicalreport,
title={Fun-ASR Technical Report},
author={Keyu An and Yanni Chen and Zhigao Chen and Chong Deng and Zhihao Du and Changfeng Gao and Zhifu Gao and Bo Gong and Xiangang Li and Yabin Li and Ying Liu and Xiang Lv and Yunjie Ji and Yiheng Jiang and Bin Ma and Haoneng Luo and Chongjia Ni and Zexu Pan and Yiping Peng and Zhendong Peng and Peiyao Wang and Hao Wang and Haoxu Wang and Wen Wang and Wupeng Wang and Yuzhong Wu and Biao Tian and Zhentao Tan and Nan Yang and Bin Yuan and Jieping Ye and Jixing Yu and Qinglin Zhang and Kun Zou and Han Zhao and Shengkui Zhao and Jingren Zhou and Yanqiao Zhu},
year={2025},
eprint={2509.12508},
archivePrefix={arXiv},
primaryClass={cs.CL},
url={https://arxiv.org/abs/2509.12508},
}
