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aec

aec-challenge

AEC Challenge

amphion

Amphion (/æmˈfaɪən/) is a toolkit for Audio, Music, and Speech Generation. Its purpose is to support reproducible research and help junior researchers and engineers get started in the field of audio, music, and speech generation research and development.

aps

A workspace for single/multi-channel speech enhancement & separation & recognition.

aqua-tk

AQUA-Tk = Audio QUality Assessment-Toolkit. (In development)

audioldm2

Text-to-Audio/Music Generation

birdsoundsdenoising

bssd

Blind Source Separation and Dereverberation

causal-u-net

unofficial PyTorch implementation of 《A Causal U-net based Neural Beamforming Network for Real-Time Multi-Channel Speech Enhancement》

cdr-dereverb

Coherence-based Dereverberation for Speech Enhancement

child-asr-paper

A list of papers for child ASR

complex-mtassnet

Multi-Task Audio Source Separation, Two-Stage Model, Complex Domain.

conferencingspeech2021

Conferencing Speech Challenge

cyclegan-music-style-transfer

Symbolic Music Genre Transfer with CycleGAN

cyclegan-music-style-transfer-refactorization

Symbolic Music Genre Transfer with CycleGAN - Refactorization

deepfilternet

Noise supression using deep filtering

demucs

Code for the paper Music Source Separation in the Waveform Domain

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.

desktopsharing

桌面共享, 支持RTSP转发, RTSP推流, RTMP推流。

dfsmn-based-lightweight-speech-enhancement

Deep Feedforward sequential memory networks(FSMN)

directional-sparse-filtering-tf

Python Implementation for Directional Sparse Filtering with Tensorflow/Keras

distant_speech_recognition

spatial signal processing toolkit a.k.a beamforming toolkit 2.0 (BTK2.0)

dns-challenge

This repo contains the scripts, models and required files for the Interspeech 2020 Deep Noise Suppression (DNS) Challenge. We are open sourcing clean speech and noise files as well. Participants of this challenge will use the scripts from this repo to create data to train their noise suppressors. They will compare their method with our baseline noise suppressor and report the results.

dsp-digital-audio-processors-in-matlab

Digital audio processors such as a compressor/limiter, expander/gate, phase vocoder, multi-tap delay, flanger, reverb, dereverb, and others.

dual-microphone-noise-reduction-by-pld-technique

Working on a dual-microphone noise reduction for mobile phone in noisy environment by Power Level Different Technique (PLD).

eusipco2019

The code used for EUSIPCO 2019. The latest version is available in SoundSourceSeparation repository.

filterbanks_fastpythonimplementation

Filter Banks, Fast Python Implementation

freevc

FreeVC: Towards High-Quality Text-Free One-Shot Voice Conversion

graph_pit

groove2groove

Code for "Groove2Groove: One-Shot Music Style Transfer with Supervision from Synthetic Data"