andong-li-speech Goto Github PK
Name: AndongLi
Type: User
Bio: Institute of Acoustics, Chinese Academy of Sciences (IACAS). Love is love : )
Location: Beijing, China
Name: AndongLi
Type: User
Bio: Institute of Acoustics, Chinese Academy of Sciences (IACAS). Love is love : )
Location: Beijing, China
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Multi-level Wavelet-CNN for Image Restoration
Different implementations of "Weighted Prediction Error" for speech dereverberation
Network Slimming (Pytorch) (ICCV 2017)
Neural network supported GEV beamformer
Audio processing by using pytorch 1D convolution network
An official implementation of "Learning with Privileged Information for Efficient Image Super-Resolution" (ECCV2020) in PyTorch.
Progressive Image Deraining Networks: A Better and Simpler Baseline (CVPR 2019)
simple and efficient python implemention of a series of adaptive filters. including time domain adaptive filters(lms、nlms、rls、ap、kalman)、nonlinear adaptive filters(volterra filter、functional link adaptive filters)、frequency domain adaptive filters(frequency domain adaptive filter、frequency domain kalman filter) for acoustic echo cancellation.
Pyroomacoustics is a package for audio signal processing for indoor applications. It was developed as a fast prototyping platform for beamforming algorithms in indoor scenarios.
i-RevNet Pytorch Code
An STFT/iSTFT for PyTorch.
A temporal module for PyTorch-ComplexTensor
Single Image Deraining Using Bilateral Recurrent Network (TIP 2020)
implementation of Monaural Speech Enhancement with Recursive Learning in the Time Domain
Supervised Spiking Network
Codebase for the paper "Sep-Stereo: Visually Guided Stereophonic Audio Generation by Associating Source Separation" (ECCV2020)
Official Implementation of SERIL in Pytorch
A Python toolbox for speech features extraction
Spherical Microphone array Impulse Response generator (SMIRgen)
target speaker extraction and verification for multi-talker speech
Use DNN to estimate the phase information of the speech in STFT filed to improve the speech quality and intelligibility.
transform-average-concatenate (TAC) method for end-to-end microphone permutation and number invariant ad-hoc beamforming.
This is the implementation of the manuscript "Learning General All-Neural Speech Enhancement based on Taylor's Approximation Theory", which is the extension of our previous work in IJCAI2022 and is submitted to TASLP.
The implementation of TaylorBeamformer, which is in submission to Interspeech2022
This is the demo website of TaylorBM, which is in submission to ICASSP 2023.
This is the implementation of the paper ''Taylor, Can You Hear Me Now? A Taylor-Unfolding Framework for Monaural Speech Enhancement'', which was accepted by IJCAI-ECAI2022 (Long oral)
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