449 Repositories
Python speech-annotation Libraries
Cross-lingual Transfer for Speech Processing using Acoustic Language Similarity
Cross-lingual Transfer for Speech Processing using Acoustic Language Similarity Indic TTS Samples can be found at https://peter-yh-wu.github.io/cross-
Explore different way to mix speech model(wav2vec2, hubert) and nlp model(BART,T5,GPT) together
SpeechMix Explore different way to mix speech model(wav2vec2, hubert) and nlp model(BART,T5,GPT) together. Introduction For the same input: from datas
PyTorch implementation of "A Two-Stage End-to-End System for Speech-in-Noise Hearing Aid Processing"
Implementation of the Sheffield entry for the first Clarity enhancement challenge (CEC1) This repository contains the PyTorch implementation of "A Two
Implemented shortest-circuit disambiguation, maximum probability disambiguation, HMM-based lexical annotation and BiLSTM+CRF-based named entity recognition
Implemented shortest-circuit disambiguation, maximum probability disambiguation, HMM-based lexical annotation and BiLSTM+CRF-based named entity recognition
Installation, test and evaluation of Scribosermo speech-to-text engine
Scribosermo STT Setup Scribosermo is a LGPL licensed, open-source speech recognition engine to "Train fast Speech-to-Text networks in different langua
Simple web application, which has a single endpoint, dedicated to annotation parsing and convertion.
Simple web application, which has a single endpoint, dedicated to annotation parsing and conversion.
Text to speech converter with GUI made in Python.
Text-to-speech-with-GUI Text to speech converter with GUI made in Python. To run this download the zip file and run the main file or clone this repo.
Azure Text-to-speech service for Home Assistant
Azure Text-to-speech service for Home Assistant The Azure text-to-speech platform uses online Azure Text-to-Speech cognitive service to read a text wi
Terminal-based audio-to-text converter
att Terminal-based audio-to-text converter Project description A terminal-based audio-to-text converter written in python, enabling you to convert .wa
Simple, hackable offline speech to text - using the VOSK-API.
Simple, hackable offline speech to text - using the VOSK-API.
Text layer for bio-image annotation.
napari-text-layer Napari text layer for bio-image annotation. Installation You can install using pip: pip install napari-text-layer Keybindings and m
Simple Speech to Text, Text to Speech
Simple Speech to Text, Text to Speech 1. Download Repository Opsi 1 Download repository ini, extract di lokasi yang diinginkan Opsi 2 Jika sudah famil
The PyTorch based implementation of continuous integrate-and-fire (CIF) module.
CIF-PyTorch This is a PyTorch based implementation of continuous integrate-and-fire (CIF) module for end-to-end (E2E) automatic speech recognition (AS
Cobra is a highly-accurate and lightweight voice activity detection (VAD) engine.
On-device voice activity detection (VAD) powered by deep learning.
SpeechBrain is an open-source and all-in-one speech toolkit based on PyTorch.
The SpeechBrain Toolkit SpeechBrain is an open-source and all-in-one speech toolkit based on PyTorch. The goal is to create a single, flexible, and us
🤗 Transformers: State-of-the-art Natural Language Processing for Pytorch, TensorFlow, and JAX.
English | 简体中文 | 繁體中文 | 한국어 State-of-the-art Natural Language Processing for Jax, PyTorch and TensorFlow 🤗 Transformers provides thousands of pretrai
PyTorch Implementation of ByteDance's Cross-speaker Emotion Transfer Based on Speaker Condition Layer Normalization and Semi-Supervised Training in Text-To-Speech
Cross-Speaker-Emotion-Transfer - PyTorch Implementation PyTorch Implementation of ByteDance's Cross-speaker Emotion Transfer Based on Speaker Conditio
A flask application to predict the speech emotion of any .wav file.
This is a speech emotion recognition app. It will allow you to train a modular MLP model with the RAVDESS dataset, and then use that model with a flask application to predict the speech emotion of any .wav file.
Official implementation of the paper: "LDNet: Unified Listener Dependent Modeling in MOS Prediction for Synthetic Speech"
LDNet Author: Wen-Chin Huang (Nagoya University) Email: [email protected] This is the official implementation of the paper "LDNet
A simple Blog Using Django Framework and Used IBM Cloud Services for Text Analysis and Text to Speech
ElhamBlog Cloud Computing Course first assignment. A simple Blog Using Django Framework and Used IBM Cloud Services for Text Analysis and Text to Spee
Neural Lexicon Reader: Reduce Pronunciation Errors in End-to-end TTS by Leveraging External Textual Knowledge
Neural Lexicon Reader: Reduce Pronunciation Errors in End-to-end TTS by Leveraging External Textual Knowledge This is an implementation of the paper,
Minimal GUI for accessing the Watson Text to Speech service.
Description Minimal graphical application for accessing the Watson Text to Speech service. Requirements Python 3 plus all dependencies listed in requi
Recognition of 38 speech commands in russian. Based on Yandex Cup 2021 ML Challenge: ASR
Speech_38_ru_commands Recognition of 38 speech commands in russian. Based on Yandex Cup 2021 ML Challenge: ASR Программа умеет распознавать 38 ключевы
A Python/Pytorch app for easily synthesising human voices
Voice Cloning App A Python/Pytorch app for easily synthesising human voices Documentation Discord Server Video guide Voice Sharing Hub FAQ's System Re
The end-to-end platform for building voice products at scale
Picovoice Made in Vancouver, Canada by Picovoice Picovoice is the end-to-end platform for building voice products on your terms. Unlike Alexa and Goog
On-device speech-to-intent engine powered by deep learning
Rhino Made in Vancouver, Canada by Picovoice Rhino is Picovoice's Speech-to-Intent engine. It directly infers intent from spoken commands within a giv
KITTI-360 Annotation Tool is a framework that developed based on python(cherrypy + jinja2 + sqlite3) as the server end and javascript + WebGL as the front end.
KITTI-360 Annotation Tool is a framework that developed based on python(cherrypy + jinja2 + sqlite3) as the server end and javascript + WebGL as the front end.
Connectionist Temporal Classification (CTC) decoding algorithms: best path, beam search, lexicon search, prefix search, and token passing. Implemented in Python.
CTC Decoding Algorithms Update 2021: installable Python package Python implementation of some common Connectionist Temporal Classification (CTC) decod
TagLab: an image segmentation tool oriented to marine data analysis
TagLab: an image segmentation tool oriented to marine data analysis TagLab was created to support the activity of annotation and extraction of statist
Production First and Production Ready End-to-End Speech Recognition Toolkit
WeNet 中文版 Discussions | Docs | Papers | Runtime (x86) | Runtime (android) | Pretrained Models We share neural Net together. The main motivation of WeN
glow-speak is a fast, local, neural text to speech system that uses eSpeak-ng as a text/phoneme front-end.
Glow-Speak glow-speak is a fast, local, neural text to speech system that uses eSpeak-ng as a text/phoneme front-end. Installation git clone https://g
Identify the emotion of multiple speakers in an Audio Segment
MevonAI - Speech Emotion Recognition Identify the emotion of multiple speakers in a Audio Segment Report Bug · Request Feature Try the Demo Here Table
Neural Lexicon Reader: Reduce Pronunciation Errors in End-to-end TTS by Leveraging External Textual Knowledge
Neural Lexicon Reader: Reduce Pronunciation Errors in End-to-end TTS by Leveraging External Textual Knowledge This is an implementation of the paper,
Official implementation of the paper: "LDNet: Unified Listener Dependent Modeling in MOS Prediction for Synthetic Speech"
LDNet Author: Wen-Chin Huang (Nagoya University) Email: [email protected] This is the official implementation of the paper "LDNet
Replication Package for AequeVox:Automated Fariness Testing for Speech Recognition Systems
AequeVox Replication Package for AequeVox:Automated Fariness Testing for Speech Recognition Systems README under development. Python Packages Required
End-to-End Speech Processing Toolkit
ESPnet: end-to-end speech processing toolkit system/pytorch ver. 1.3.1 1.4.0 1.5.1 1.6.0 1.7.1 1.8.1 1.9.0 ubuntu20/python3.9/pip ubuntu20/python3.8/p
CPC-big and k-means clustering for zero-resource speech processing
The CPC-big model and k-means checkpoints used in Analyzing Speaker Information in Self-Supervised Models to Improve Zero-Resource Speech Processing.
Light-SERNet: A lightweight fully convolutional neural network for speech emotion recognition
Light-SERNet This is the Tensorflow 2.x implementation of our paper "Light-SERNet: A lightweight fully convolutional neural network for speech emotion
Maix Speech AI lib, including ASR, chat, TTS etc.
Maix-Speech 中文 | English Brief Now only support Chinese, See 中文 Build Clone code by: git clone https://github.com/sipeed/Maix-Speech Compile x86x64 c
translate using your voice
speech-to-text-translator Usage translate using your voice description this project makes translating a word easy, all you have to do is speak and...
To lazy to read your homework ? Get it done with LOL
LOL To lazy to read your homework ? Get it done with LOL Needs python 3.x L:::::::::L OO:::::::::OO L:::::::::L L:::::::
PocketSphinx is a lightweight speech recognition engine, specifically tuned for handheld and mobile devices, though it works equally well on the desktop
PocketSphinx 5prealpha This is PocketSphinx, one of Carnegie Mellon University's open source large vocabulary, speaker-independent continuous speech r
An Agnostic Computer Vision Framework - Pluggable to any Training Library: Fastai, Pytorch-Lightning with more to come
IceVision is the first agnostic computer vision framework to offer a curated collection with hundreds of high-quality pre-trained models from torchvision, MMLabs, and soon Pytorch Image Models. It orchestrates the end-to-end deep learning workflow allowing to train networks with easy-to-use robust high-performance libraries such as Pytorch-Lightning and Fastai
MaD GUI is a basis for graphical annotation and computational analysis of time series data.
MaD GUI Machine Learning and Data Analytics Graphical User Interface MaD GUI is a basis for graphical annotation and computational analysis of time se
LeBenchmark: a reproducible framework for assessing SSL from speech
LeBenchmark: a reproducible framework for assessing SSL from speech
AutoSub is a CLI application to generate subtitle files (.srt, .vtt, and .txt transcript) for any video file using Mozilla DeepSpeech.
AutoSub About Motivation Installation Docker How-to example How it works TO-DO Contributing References About AutoSub is a CLI application to generate
Self-Supervised Speech Pre-training and Representation Learning Toolkit.
What's New Sep 2021: We host a challenge in AAAI workshop: The 2nd Self-supervised Learning for Audio and Speech Processing! See SUPERB official site
UniSpeech - Large Scale Self-Supervised Learning for Speech
UniSpeech The family of UniSpeech: UniSpeech (ICML 2021): Unified Pre-training for Self-Supervised Learning and Supervised Learning for ASR UniSpeech-
FG-transformer-TTS Fine-grained style control in transformer-based text-to-speech synthesis
LST-TTS Official implementation for the paper Fine-grained style control in transformer-based text-to-speech synthesis. Submitted to ICASSP 2022. Audi
This is the implementation of "SELF SUPERVISED REPRESENTATION LEARNING WITH DEEP CLUSTERING FOR ACOUSTIC UNIT DISCOVERY FROM RAW SPEECH" submitted to ICASSP 2022
CPC_DeepCluster This is the implementation of "SELF SUPERVISED REPRESENTATION LEARNING WITH DEEP CLUSTERING FOR ACOUSTIC UNIT DISCOVERY FROM RAW SPEEC
Leon is an open-source personal assistant who can live on your server.
Leon Your open-source personal assistant. Website :: Documentation :: Roadmap :: Contributing :: Story 👋 Introduction Leon is an open-source personal
Python SDK for working with Voicegain Speech-to-Text
Voicegain Speech-to-Text Python SDK Python SDK for the Voicegain Speech-to-Text API. This API allows for large vocabulary speech-to-text transcription
A PyTorch implementation of paper "Learning Shared Semantic Space for Speech-to-Text Translation", ACL (Findings) 2021
Chimera: Learning Shared Semantic Space for Speech-to-Text Translation This is a Pytorch implementation for the "Chimera" paper Learning Shared Semant
African language Speech Recognition - Speech-to-Text
Swahili-Speech-To-Text Table of Contents Swahili-Speech-To-Text Overview Scenario Approach Project Structure data: models: notebooks: scripts tests: l
A Low Complexity Speech Enhancement Framework for Full-Band Audio (48kHz) based on Deep Filtering.
DeepFilterNet A Low Complexity Speech Enhancement Framework for Full-Band Audio (48kHz) based on Deep Filtering. libDF contains Rust code used for dat
FAST-RIR: FAST NEURAL DIFFUSE ROOM IMPULSE RESPONSE GENERATOR
This is the official implementation of our neural-network-based fast diffuse room impulse response generator (FAST-RIR) for generating room impulse responses (RIRs) for a given acoustic environment.
Azure Neural Speech Service TTS
Written in Python using the Azure Speech SDK. App.py provides an easy way to create an Text-To-Speech request to Azure Speech and download the wav file.
Any-to-any voice conversion using synthetic specific-speaker speeches as intermedium features
MediumVC MediumVC is an utterance-level method towards any-to-any VC. Before that, we propose SingleVC to perform A2O tasks(Xi → Ŷi) , Xi means utter
Official implementation of EdiTTS: Score-based Editing for Controllable Text-to-Speech
EdiTTS: Score-based Editing for Controllable Text-to-Speech Official implementation of EdiTTS: Score-based Editing for Controllable Text-to-Speech. Au
StrengthNet: Deep Learning-based Emotion Strength Assessment for Emotional Speech Synthesis
StrengthNet Implementation of "StrengthNet: Deep Learning-based Emotion Strength Assessment for Emotional Speech Synthesis" https://arxiv.org/abs/2110
A 10000+ hours dataset for Chinese speech recognition
WenetSpeech Official website | Paper A 10000+ Hours Multi-domain Chinese Corpus for Speech Recognition Download Please visit the official website, rea
Speech Separation Using an Asynchronous Fully Recurrent Convolutional Neural Network
Speech Separation Using an Asynchronous Fully Recurrent Convolutional Neural Network This repository is the official implementation of Speech Separati
PyTorch Implementation of PortaSpeech: Portable and High-Quality Generative Text-to-Speech
PortaSpeech - PyTorch Implementation PyTorch Implementation of PortaSpeech: Portable and High-Quality Generative Text-to-Speech. Model Size Module Nor
Persian Kaldi profile for Rhasspy built from open speech data
Persian Kaldi Profile A Rhasspy profile for Persian (fa). Installation Get started by first installing Vosk: # Create virtual environment python3 -m v
Every Google, Azure & IBM text to speech voice for free
TTS-Grabber Quick thing i made about a year ago to download any text with any tts voice, over 630 voices to choose from currently. It will split the i
text to speech toolkit. 好用的中文语音合成工具箱,包含语音编码器、语音合成器、声码器和可视化模块。
ttskit Text To Speech Toolkit: 语音合成工具箱。 安装 pip install -U ttskit 注意 可能需另外安装的依赖包:torch,版本要求torch=1.6.0,=1.7.1,根据自己的实际环境安装合适cuda或cpu版本的torch。 ttskit的
Implementation of "StrengthNet: Deep Learning-based Emotion Strength Assessment for Emotional Speech Synthesis"
StrengthNet Implementation of "StrengthNet: Deep Learning-based Emotion Strength Assessment for Emotional Speech Synthesis" https://arxiv.org/abs/2110
EdiTTS: Score-based Editing for Controllable Text-to-Speech
Official implementation of EdiTTS: Score-based Editing for Controllable Text-to-Speech
PortaSpeech - PyTorch Implementation
PortaSpeech - PyTorch Implementation PyTorch Implementation of PortaSpeech: Portable and High-Quality Generative Text-to-Speech. Model Size Module Nor
IJON is an annotation mechanism that analysts can use to guide fuzzers such as AFL.
IJON SPACE EXPLORER IJON is an annotation mechanism that analysts can use to guide fuzzers such as AFL. Using only a small (usually one line) annotati
PyTorch implementation of Microsoft's text-to-speech system FastSpeech 2: Fast and High-Quality End-to-End Text to Speech.
An implementation of Microsoft's "FastSpeech 2: Fast and High-Quality End-to-End Text to Speech"
Refactored version of FastSpeech2
Refactored version of FastSpeech2. An implementation of Microsoft's "FastSpeech 2: Fast and High-Quality End-to-End Text to Speech"
An official reimplementation of the method described in the INTERSPEECH 2021 paper - Speech Resynthesis from Discrete Disentangled Self-Supervised Representations.
Speech Resynthesis from Discrete Disentangled Self-Supervised Representations Implementation of the method described in the Speech Resynthesis from Di
VoiceFixer VoiceFixer is a framework for general speech restoration.
VoiceFixer VoiceFixer is a framework for general speech restoration. We aim at the restoration of severly degraded speech and historical speech. Paper
A 10000+ hours dataset for Chinese speech recognition
A 10000+ hours dataset for Chinese speech recognition
Voicefixer aims at the restoration of human speech regardless how serious its degraded.
Voicefixer aims at the restoration of human speech regardless how serious its degraded.
Unofficial Implementation of Zero-Shot Text-to-Speech for Text-Based Insertion in Audio Narration
Zero-Shot Text-to-Speech for Text-Based Insertion in Audio Narration This repo contains only model Implementation of Zero-Shot Text-to-Speech for Text
A Non-Autoregressive Transformer based TTS, supporting a family of SOTA transformers with supervised and unsupervised duration modelings. This project grows with the research community, aiming to achieve the ultimate TTS.
A Non-Autoregressive Transformer based TTS, supporting a family of SOTA transformers with supervised and unsupervised duration modelings. This project grows with the research community, aiming to achieve the ultimate TTS.
Lingvo is a framework for building neural networks in Tensorflow, particularly sequence models.
Lingvo is a framework for building neural networks in Tensorflow, particularly sequence models.
Live Speech Portraits: Real-Time Photorealistic Talking-Head Animation (SIGGRAPH Asia 2021)
Live Speech Portraits: Real-Time Photorealistic Talking-Head Animation This repository contains the implementation of the following paper: Live Speech
Unofficial PyTorch implementation of Google AI's VoiceFilter system
VoiceFilter Note from Seung-won (2020.10.25) Hi everyone! It's Seung-won from MINDs Lab, Inc. It's been a long time since I've released this open-sour
A PyTorch implementation of the WaveGlow: A Flow-based Generative Network for Speech Synthesis
WaveGlow A PyTorch implementation of the WaveGlow: A Flow-based Generative Network for Speech Synthesis Quick Start: Install requirements: pip install
🤗 Transformers: State-of-the-art Natural Language Processing for Pytorch, TensorFlow, and JAX.
English | 简体中文 | 繁體中文 State-of-the-art Natural Language Processing for Jax, PyTorch and TensorFlow 🤗 Transformers provides thousands of pretrained mo
PyTorch implementation of convolutional neural networks-based text-to-speech synthesis models
Deepvoice3_pytorch PyTorch implementation of convolutional networks-based text-to-speech synthesis models: arXiv:1710.07654: Deep Voice 3: Scaling Tex
PyTorch implementation of Tacotron speech synthesis model.
tacotron_pytorch PyTorch implementation of Tacotron speech synthesis model. Inspired from keithito/tacotron. Currently not as much good speech quality
Silero Models: pre-trained speech-to-text, text-to-speech models and benchmarks made embarrassingly simple
Silero Models: pre-trained speech-to-text, text-to-speech models and benchmarks made embarrassingly simple
Labelling platform for text using distant supervision
With DataQA, you can label unstructured text documents using rule-based distant supervision.
Speech Recognition using DeepSpeech2.
deepspeech.pytorch Implementation of DeepSpeech2 for PyTorch using PyTorch Lightning. The repo supports training/testing and inference using the DeepS
Accurately generate all possible forms of an English word e.g "election" -- "elect", "electoral", "electorate" etc.
Accurately generate all possible forms of an English word Word forms can accurately generate all possible forms of an English word. It can conjugate v
Neural network models for joint POS tagging and dependency parsing (CoNLL 2017-2018)
Neural Network Models for Joint POS Tagging and Dependency Parsing Implementations of joint models for POS tagging and dependency parsing, as describe
This project converts your human voice input to its text transcript and to an automated voice too.
Human Voice to Automated Voice & Text Introduction: In this project, whenever you'll speak, it will turn your voice into a robot voice and furthermore
Codename generator using WordNet parts of speech database
codenames Codename generator using WordNet parts of speech database References: https://possiblywrong.wordpress.com/2021/09/13/code-name-generator/ ht
AnnIE - Annotation Platform, tool for open information extraction annotations using text files.
AnnIE - Annotation Platform, tool for open information extraction annotations using text files.
Code of the paper "Performance-Efficiency Trade-offs in Unsupervised Pre-training for Speech Recognition"
SEW (Squeezed and Efficient Wav2vec) The repo contains the code of the paper "Performance-Efficiency Trade-offs in Unsupervised Pre-training for Speec
Efficient Conformer: Progressive Downsampling and Grouped Attention for Automatic Speech Recognition
Efficient Conformer: Progressive Downsampling and Grouped Attention for Automatic Speech Recognition Official implementation of the Efficient Conforme
Control the classic General Instrument SP0256-AL2 speech chip and AY-3-8910 sound generator with a Raspberry Pi and this Python library.
GI-Pi Control the classic General Instrument SP0256-AL2 speech chip and AY-3-8910 sound generator with a Raspberry Pi and this Python library. The SP0
✨Rubrix is a production-ready Python framework for exploring, annotating, and managing data in NLP projects.
✨A Python framework to explore, label, and monitor data for NLP projects
Implementation of "A Deep Learning Loss Function based on Auditory Power Compression for Speech Enhancement" by pytorch
This repository is used to suspend the results of our paper "A Deep Learning Loss Function based on Auditory Power Compression for Speech Enhancement"
The purpose of this code base is to add a specified signal-to-noise ratio noise from MUSAN dataset to a pure speech signal and to generate far-field speech data using room impulse response data from BUT Speech@FIT Reverb Database.
Add_noise_and_rir_to_speech The purpose of this code base is to add a specified signal-to-noise ratio noise from MUSAN dataset to a pure speech signal
OpenCVのGrabCut()を利用したセマンティックセグメンテーション向けアノテーションツール(Annotation tool using GrabCut() of OpenCV. It can be used to create datasets for semantic segmentation.)
[Japanese/English] GrabCut-Annotation-Tool GrabCut-Annotation-Tool.mp4 OpenCVのGrabCut()を利用したアノテーションツールです。 セマンティックセグメンテーション向けのデータセット作成にご使用いただけます。 ※Grab