Official model provided by Wespeaker project, ResNet293 based r-vector (After large margin finetune)
The model is trained on VoxCeleb2 Dev dataset, containing 5994 speakers.
Model Sources
- Repository: https://github.com/wenet-e2e/wespeaker
- Paper: https://arxiv.org/pdf/2210.17016.pdf
- Demo: https://huggingface.co./spaces/wenet/wespeaker_demo
Results on VoxCeleb
Model | Params | Flops | LM | AS-Norm | vox1-O-clean | vox1-E-clean | vox1-H-clean |
---|---|---|---|---|---|---|---|
ResNet293-TSTP-emb256 | 28.62M | 28.10G | × | × | 0.595 | 0.756 | 1.433 |
× | √ | 0.537 | 0.701 | 1.276 | |||
√ | × | 0.532 | 0.707 | 1.311 | |||
√ | √ | 0.447 | 0.657 | 1.183 |
Install Wespeaker
pip install git+https://github.com/wenet-e2e/wespeaker.git
for development install:
git clone https://github.com/wenet-e2e/wespeaker.git
cd wespeaker
pip install -e .
Command line Usage
$ wespeaker -p resnet293_download_dir --task embedding --audio_file audio.wav --output_file embedding.txt
$ wespeaker -p resnet293_download_dir --task embedding_kaldi --wav_scp wav.scp --output_file /path/to/embedding
$ wespeaker -p resnet293_download_dir --task similarity --audio_file audio.wav --audio_file2 audio2.wav
$ wespeaker -p resnet293_download_dir --task diarization --audio_file audio.wav
Python Programming Usage
import wespeaker
model = wespeaker.load_model_local(resnet293_download_dir)
# set_gpu to enable the cuda inference, number < 0 means using CPU
model.set_gpu(0)
# embedding/embedding_kaldi/similarity/diarization
embedding = model.extract_embedding('audio.wav')
utt_names, embeddings = model.extract_embedding_list('wav.scp')
similarity = model.compute_similarity('audio1.wav', 'audio2.wav')
diar_result = model.diarize('audio.wav')
# register and recognize
model.register('spk1', 'spk1_audio1.wav')
model.register('spk2', 'spk2_audio1.wav')
model.register('spk3', 'spk3_audio1.wav')
result = model.recognize('spk1_audio2.wav')
Citation
@inproceedings{wang2023wespeaker,
title={Wespeaker: A research and production oriented speaker embedding learning toolkit},
author={Wang, Hongji and Liang, Chengdong and Wang, Shuai and Chen, Zhengyang and Zhang, Binbin and Xiang, Xu and Deng, Yanlei and Qian, Yanmin},
booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)},
pages={1--5},
year={2023},
organization={IEEE}
}
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