Pinned Repositories
AMFM_decompy
Package containing the tools necessary for decomposing a speech signal into its modulated components (also known as AM-FM decomposition). Includes the algorithms of the QHM family and the YAAPT pitch tracker.
Android-Audio-Processing-Using-WebRTC
All in all WebRTC. A Complete Guide to enable Rich and High Quality of **Real-Time Voice Communication** on Android Platform. This repository involves a complete understanding, implementation and documentation related to WebRTC Audio Processing.
apkinson_server
crepe
CREPE: A Convolutional REpresentation for Pitch Estimation -- pre-trained model (ICASSP 2018)
DeepFormants
Formant Tracking & Estimation
DeepSpeech
A PaddlePaddle implementation of DeepSpeech2 architecture for ASR.
deepspeech.pytorch
Speech Recognition using DeepSpeech2.
espnet
End-to-End Speech Processing Toolkit
FastCqt
Fast constant-Q transform feature, c++ implement
FormantNet
Software and data supporting Lilley & Bunnell (2021) InterSpeech paper
wgsh3706's Repositories
wgsh3706/AMFM_decompy
Package containing the tools necessary for decomposing a speech signal into its modulated components (also known as AM-FM decomposition). Includes the algorithms of the QHM family and the YAAPT pitch tracker.
wgsh3706/Android-Audio-Processing-Using-WebRTC
All in all WebRTC. A Complete Guide to enable Rich and High Quality of **Real-Time Voice Communication** on Android Platform. This repository involves a complete understanding, implementation and documentation related to WebRTC Audio Processing.
wgsh3706/apkinson_server
wgsh3706/crepe
CREPE: A Convolutional REpresentation for Pitch Estimation -- pre-trained model (ICASSP 2018)
wgsh3706/DeepFormants
Formant Tracking & Estimation
wgsh3706/DeepSpeech
A PaddlePaddle implementation of DeepSpeech2 architecture for ASR.
wgsh3706/deepspeech.pytorch
Speech Recognition using DeepSpeech2.
wgsh3706/espnet
End-to-End Speech Processing Toolkit
wgsh3706/FastCqt
Fast constant-Q transform feature, c++ implement
wgsh3706/FormantNet
Software and data supporting Lilley & Bunnell (2021) InterSpeech paper
wgsh3706/GPT-SoVITS
1 min voice data can also be used to train a good TTS model! (few shot voice cloning)
wgsh3706/omnizart
Omniscient Mozart, being able to transcribe everything in the music, including vocal, drum, chord, beat, instruments, and more.
wgsh3706/Paddle
PArallel Distributed Deep LEarning （『飞桨』核心框架，高性能单机、分布式训练和跨平台部署）
wgsh3706/penn
Pitch Estimating Neural Networks (PENN)
wgsh3706/Parselmouth
Praat in Python, the Pythonic way
wgsh3706/pesto-full
Full models and training code for PESTO
wgsh3706/Real-Time-Convolutional-Neural-Network-Based-Speech-Source-Localization-on-Smartphone
speech source localization on phone/pad
wgsh3706/REAPER
wgsh3706/rnnoise
Recurrent neural network for audio noise reduction
wgsh3706/SMA2
Sorce code of Apkinson: android app to monitor the motor symptoms of Parkinson's patients
wgsh3706/Speaker-Diarization
speaker diarization by uis-rnn and speaker embedding by vgg-speaker-recognition
wgsh3706/speechbrain
A PyTorch-based Speech Toolkit
wgsh3706/spleeter
Deezer source separation library including pretrained models.
wgsh3706/VGG-Speaker-Recognition
Utterance-level Aggregation For Speaker Recognition In The Wild
wgsh3706/webrtc-sdk
wgsh3706/webrtc-sdk-rnnoise
Enhance real-time communication with our WebRTC SDK integrated with advanced RNNoise technology. Enjoy noise-free audio calls, powered by Chromium open-source code. Flexible API for seamless development, cross-platform compatibility, and improved user experience.