模型:
anton-l/wav2vec2-base-superb-sv
This is a ported version of S3PRL's Wav2Vec2 for the SUPERB Speaker Verification task .
The base model is wav2vec2-large-lv60 , which is pretrained on 16kHz sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
For more information refer to SUPERB: Speech processing Universal PERformance Benchmark
The model should not be used to intentionally create hostile or alienating environments for people.
Significant research has explored bias and fairness issues with language models (see, e.g., Sheng et al. (2021) and Bender et al. (2021) ). Predictions generated by the model may include disturbing and harmful stereotypes across protected classes; identity characteristics; and sensitive, social, and occupational groups.
Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations.
See the superb dataset card
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See the superb dataset card
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Carbon emissions can be estimated using the Machine Learning Impact calculator presented in Lacoste et al. (2019) .
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BibTeX:
@misc{https://doi.org/10.48550/arxiv.2006.11477, doi = {10.48550/ARXIV.2006.11477}, url = {https://arxiv.org/abs/2006.11477}, author = {Baevski, Alexei and Zhou, Henry and Mohamed, Abdelrahman and Auli, Michael}, keywords = {Computation and Language (cs.CL), Machine Learning (cs.LG), Sound (cs.SD), Audio and Speech Processing (eess.AS), FOS: Computer and information sciences, FOS: Computer and information sciences, FOS: Electrical engineering, electronic engineering, information engineering, FOS: Electrical engineering, electronic engineering, information engineering}, title = {wav2vec 2.0: A Framework for Self-Supervised Learning of Speech Representations}, publisher = {arXiv}, @misc{https://doi.org/10.48550/arxiv.2105.01051, doi = {10.48550/ARXIV.2105.01051}, url = {https://arxiv.org/abs/2105.01051}, author = {Yang, Shu-wen and Chi, Po-Han and Chuang, Yung-Sung and Lai, Cheng-I Jeff and Lakhotia, Kushal and Lin, Yist Y. and Liu, Andy T. and Shi, Jiatong and Chang, Xuankai and Lin, Guan-Ting and Huang, Tzu-Hsien and Tseng, Wei-Cheng and Lee, Ko-tik and Liu, Da-Rong and Huang, Zili and Dong, Shuyan and Li, Shang-Wen and Watanabe, Shinji and Mohamed, Abdelrahman and Lee, Hung-yi}, keywords = {Computation and Language (cs.CL), Sound (cs.SD), Audio and Speech Processing (eess.AS), FOS: Computer and information sciences, FOS: Computer and information sciences, FOS: Electrical engineering, electronic engineering, information engineering, FOS: Electrical engineering, electronic engineering, information engineering}, title = {SUPERB: Speech processing Universal PERformance Benchmark}, publisher = {arXiv}, year = {2021}, }
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Anton Lozhkov in collaboration with Ezi Ozoani and the Hugging Face team
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Use the code below to get started with the model.
Click to expandfrom transformers import AutoProcessor, AutoModelForAudioXVector processor = AutoProcessor.from_pretrained("anton-l/wav2vec2-base-superb-sv") model = AutoModelForAudioXVector.from_pretrained("anton-l/wav2vec2-base-superb-sv")