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worstchan/EAT-large_epoch20_pretrain
EAT-large_epoch20_pretrain is a feature extraction model from worstchan. Use it when you need embeddings to search or compare text. It is set up for transformers. The card lists the license as mit.
This is the pre-trained EAT-large model at epoch 20, trained on the AS-2M dataset using the EAT framework for audio self-supervised learning. It offers efficient feature extraction and can also serve as a strong initi…
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From the Hugging Face model README
This is the pre-trained EAT-large model at epoch 20, trained on the AS-2M dataset using the EAT framework for audio self-supervised learning. It offers efficient feature extraction and can also serve as a strong initialization for fine-tuning on a wide range of downstream audio understanding tasks such as classification and captioning.
For more details on the EAT framework, please refer to the GitHub repository and our paper EAT: Self-Supervised Pre-Training with Efficient Audio Transformer.
You can load and use the model for feature extraction directly via Hugging Face Transformers:
import torchaudio
import torch
import soundfile as sf
import numpy as np
from transformers import AutoModel
model_id = "worstchan/EAT-large_epoch20_pretrain"
model = AutoModel.from_pretrained(model_id, trust_remote_code=True).eval().cuda()
source_file = "/path/to/input.wav"
target_file = "/path/to/output.npy"
target_length = 1024 # Recommended: 1024 for 10s audio
norm_mean = -4.268
norm_std = 4.569
# Load and resample audio
wav, sr = sf.read(source_file)
waveform = torch.tensor(wav).float().cuda()
if sr != 16000:
waveform = torchaudio.functional.resample(waveform, sr, 16000)
# Normalize and convert to mel-spectrogram
waveform = waveform - waveform.mean()
mel = torchaudio.compliance.kaldi.fbank(
waveform.unsqueeze(0),
htk_compat=True,
sample_frequency=16000,
use_energy=False,
window_type='hanning',
num_mel_bins=128,
dither=0.0,
frame_shift=10
).unsqueeze(0)
# Pad or truncate
n_frames = mel.shape[1]
if n_frames < target_length:
mel = torch.nn.ZeroPad2d((0, 0, 0, target_length - n_frames))(mel)
else:
mel = mel[:, :target_length, :]
# Normalize
mel = (mel - norm_mean) / (norm_std * 2)
mel = mel.unsqueeze(0).cuda() # shape: [1, 1, T, F]
# Extract features
with torch.no_grad():
feat = model.extract_features(mel)
feat = feat.squeeze(0).cpu().numpy()
np.save(target_file, feat)
print(f"Feature shape: {feat.shape}")
print(f"Saved to: {target_file}")
The model supports both frame-level (~50Hz) and utterance-level (CLS token) representations. See the feature extraction guide for more instructions.
If you find this model useful, please consider citing our paper:
@article{chen2024eat,
title={EAT: Self-supervised pre-training with efficient audio transformer},
author={Chen, Wenxi and Liang, Yuzhe and Ma, Ziyang and Zheng, Zhisheng and Chen, Xie},
journal={arXiv preprint arXiv:2401.03497},
year={2024}
}