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a-convolutional-recurrent-neural-network-for-real-time-speech-enhancement

Implement A Convolutional Recurrent Neural Network for Real-Time Speech Enhancement by PyTorch.

api_auxiva_mask_online

athena-signal

audio-bandwidth-extension

This research project aims at studying and finding a suitable method to implement audio bandwidth extension to bandlimited audio files.

autoeq

Automatic headphone equalization from frequency responses

awesome-deep-learning-music

List of articles related to deep learning applied to music

bandwidth_extension

beamformit

BeamformIt acoustic beamforming software

cnn-for-single-channel-speech-enhancement

Convolutional neural nets for single channel speech enhancement

deepdoa

Finding Direction of arrival (DOA) of small UAVs using Sparse Denoising Autoencoders and Deep Neural Networks.

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.

dnnmapper

Mapping features using Deep Neural Networks (DNNs) with application to Voice Conversion (VC). The implementations are on top of Theano Python library

dual_channel_beamformer_and_postfilter

This project gives an example of dual microphone speech enhancement based on GSC beamformer and multiple channel postfilter.

fftresample

A sample for resampling a signal with FFT in Octave or Matlab

gcc-nmf

Real-time GCC-NMF Blind Speech Separation and Enhancement

git-recipes

🥡 Git recipes in Chinese by Zhongyi Tong. 高质量的Git中文教程.

griffin_lim

Implementation of the Griffin and Lim algorithm to recover an audio signal from a magnitude-only spectrogram.

gsc_beamforming

Generalised Sidelobe Canceler beamformer for an array of microphones using matlab

hello-world

just test project

ilrma

Copyright 2018 Daichi Kitamura. These programs are distributed only for academic research at universities and research institutions. It is not allowed to use or modify these programs for commercial or industrial purpose without our permission. When you use or modify these programs and write research articles, cite the following references: D. Kitamura, N. Ono, H. Sawada, H. Kameoka, H. Saruwatari, "Determined blind source separation unifying independent vector analysis and nonnegative matrix factorization," IEEE/ACM Trans. ASLP, vol. 24, no. 9, pp. 1626-1641, September 2016. D. Kitamura, N. Ono, H. Sawada, H. Kameoka, H. Saruwatari, "Determined blind source separation with independent low-rank matrix analysis," Audio Source Separation. Signals and Communication Technology., S. Makino, Ed. Springer, Cham, pp. 125-155, March 2018.

linear-dma

Linear Differential Microphone Array Processing

mcse

Multi-channel speech enhancement system (MVDR beamformer + several postfilters)

mlsp

Examples of machine learning and signal processing algorithms.

music-source-separation

Separating singing voice from music based on deep neural net in Tensorflow

nn-gev

Neural network supported GEV beamformer

nnaec-neuralnetworkbasedacousticechocancellation

NNAEC-Neural Network based Acoustic Echo Cancellation

norbert

Painless Wiener filters for audio separation

open-unmix-pytorch

Open-Unmix - Music Source Separation for PyTorch

pyvocoder

a python library for different types of vocoders like LPC, MCEP, PSOLA, etc.

realtime-yukarin

An application for real-time voice conversion