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newoneincntk's Projects

asteroid icon asteroid

The PyTorch-based audio source separation toolkit for researchers

attentionaugmentedconvlstm icon attentionaugmentedconvlstm

Implementation of TAAConvLSTM and SAAConvLSTM used in "Attention Augmented ConvLSTM for Environment Prediction"

audiogpt icon audiogpt

AudioGPT: Understanding and Generating Speech, Music, Sound, and Talking Head

awesome-speech-enhancement icon awesome-speech-enhancement

A tutorial for Speech Enhancement researchers and practitioners. The purpose of this repo is to organize the worldโ€™s resources for speech enhancement and make them universally accessible and useful.

bs-roformer icon bs-roformer

Implementation of Band Split Roformer, SOTA Attention network for music source separation out of ByteDance AI Labs

cdiffuse icon cdiffuse

Conditional Diffusion Probabilistic Model for Speech Enhancement

clarity icon clarity

Clarity Challenge toolkit - software for building Clarity Challenge systems

cleanunet icon cleanunet

Official PyTorch Implementation of CleanUNet (ICASSP 2022)

cmgan icon cmgan

Conformer-based Metric GAN for speech enhancement

cntk icon cntk

Computational Network Toolkit (CNTK)

cntkx icon cntkx

Deep learning library that builds on and extends Microsoft CNTK

complex_networks icon complex_networks

Contain some activations, dropout, BN, LN, Linear, Conv, ConvTranspose, LSTM(LN)

complexpytorch icon complexpytorch

A high-level toolbox for using complex valued neural networks in PyTorch

conformer icon conformer

PyTorch implementation of "Conformer: Convolution-augmented Transformer for Speech Recognition" (INTERSPEECH 2020)

cruse icon cruse

a lightweight network for monaural speech enhancement

denoiser icon denoiser

Real Time Speech Enhancement in the Waveform Domain (Interspeech 2020)We provide a PyTorch implementation of the paper Real Time Speech Enhancement in the Waveform Domain. In which, we present a causal speech enhancement model working on the raw waveform that runs in real-time on a laptop CPU. The proposed model is based on an encoder-decoder architecture with skip-connections. It is optimized on both time and frequency domains, using multiple loss functions. Empirical evidence shows that it is capable of removing various kinds of background noise including stationary and non-stationary noises, as well as room reverb. Additionally, we suggest a set of data augmentation techniques applied directly on the raw waveform which further improve model performance and its generalization abilities.

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