(简体中文|English)
FunASR hopes to build a bridge between academic research and industrial applications on speech recognition. By supporting the training & finetuning of the industrial-grade speech recognition model released on ModelScope, researchers and developers can conduct research and production of speech recognition models more conveniently, and promote the development of speech recognition ecology. ASR for Fun!
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Challenge details ref to (CN/EN)
Speaker Recognition
Punctuation Restoration
Endpoint Detection
Timestamp Prediction
Install from pipshell pip3 install -U funasr # For the users in China, you could install with the command: # pip3 install -U funasr -i https://mirror.sjtu.edu.cn/pypi/web/simple
Or install from source code
git clone https://github.com/alibaba/FunASR.git && cd FunASR
pip3 install -e ./
# For the users in China, you could install with the command:
# pip3 install -e ./ -i https://mirror.sjtu.edu.cn/pypi/web/simple
If you want to use the pretrained models in ModelScope, you should install the modelscope:
pip3 install -U modelscope
# For the users in China, you could install with the command:
# pip3 install -U modelscope -f https://modelscope.oss-cn-beijing.aliyuncs.com/releases/repo.html -i https://mirror.sjtu.edu.cn/pypi/web/simple
For more details, please ref to installation
You can use FunASR in the following ways:
Supports real-time streaming speech recognition, uses non-streaming models for error correction, and outputs text with punctuation. Currently, only single client is supported. For multi-concurrency, please refer to the C++ version service deployment SDK below.
cd funasr/runtime/python/websocket
python funasr_wss_server.py --port 10095
python funasr_wss_client.py --host "127.0.0.1" --port 10095 --mode 2pass --chunk_size "5,10,5"
For more examples, please refer to docs.
Currently, offline file transcription service (CPU) is supported, and concurrent requests of hundreds of channels are supported.
You can use the following command to complete the deployment with one click:
curl -O https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/shell/funasr-runtime-deploy-offline-cpu-zh.sh
sudo bash funasr-runtime-deploy-offline-cpu-zh.sh install --workspace ./funasr-runtime-resources
python3 funasr_wss_client.py --host "127.0.0.1" --port 10095 --mode offline --audio_in "../audio/asr_example.wav"
For more examples, please refer to docs
If you want to use the pre-trained industrial models in ModelScope for inference or fine-tuning training, you can refer to the following command:
from modelscope.pipelines import pipeline
from modelscope.utils.constant import Tasks
inference_pipeline = pipeline(
task=Tasks.auto_speech_recognition,
model='damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch',
)
rec_result = inference_pipeline(audio_in='https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav')
print(rec_result)
# {'text': '欢迎大家来体验达摩院推出的语音识别模型'}
More examples could be found in docs
If you want to train from scratch, usually for academic models, you can start training and inference with the following command:
cd egs/aishell/paraformer
. ./run.sh --CUDA_VISIBLE_DEVICES="0,1" --gpu_num=2
More examples could be found in docs
If you have any questions about FunASR, please contact us by
| Dingding group | Wechat group |
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This project is licensed under the The MIT License. FunASR also contains various third-party components and some code modified from other repos under other open source licenses.
The use of pretraining model is subject to model licencs
@inproceedings{gao2023funasr,
author={Zhifu Gao and Zerui Li and Jiaming Wang and Haoneng Luo and Xian Shi and Mengzhe Chen and Yabin Li and Lingyun Zuo and Zhihao Du and Zhangyu Xiao and Shiliang Zhang},
title={FunASR: A Fundamental End-to-End Speech Recognition Toolkit},
year={2023},
booktitle={INTERSPEECH},
}
@inproceedings{gao22b_interspeech,
author={Zhifu Gao and ShiLiang Zhang and Ian McLoughlin and Zhijie Yan},
title={{Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition}},
year=2022,
booktitle={Proc. Interspeech 2022},
pages={2063--2067},
doi={10.21437/Interspeech.2022-9996}
}