Speech
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General
See also Automation etc.
Recognition
- https://en.wikipedia.org/wiki/Speech_recognition - an interdisciplinary subfield of computational linguistics that develops methodologies and technologies that enables the recognition and translation of spoken language into text by computers. It is also known as automatic speech recognition (ASR), computer speech recognition or speech to text (STT). It incorporates knowledge and research in the linguistics, computer science, and electrical engineering fields.
- https://github.com/Picovoice/speech-to-text-benchmark - a minimalist and extensible framework for benchmarking different speech-to-text engines. It has been developed and tested on Ubuntu using Python3. [1]
CMUSphinx
- CMUSphinx Open Source Speech Recognition
- https://github.com/cmusphinx/pocketsphinx - a lightweight speech recognition engine, specifically tuned for handheld and mobile devices, though it works equally well on the desktop
- https://github.com/syl22-00/pocketsphinx.js - Speech recognition in JavaScript and WebAssembly
pocketVox
- https://github.com/benoitfragit/pocketVox - a both an application and a library written in C by Benoit Franquet. It uses Pocketsphinx to do voice recognition. The voice recognition is done offline and doesn't need an Internet connexion. It's aim is to provide an eficient way to integrate voice recognition on the Linux desktop. More particularly, its main goal is to give to visual impaired (as I am) a powerfull tool to control their desktop.
Audiogrep
- Audiogrep - transcribes audio files and then creates "audio supercuts" based on search phrases. It uses CMU Pocketsphinx for speech-to-text and pydub to stitch things together. [2]
Speech Signal Processing Toolkit
- Speech Signal Processing Toolkit (SPTK) - a suite of speech signal processing tools for UNIX environments, e.g., LPC analysis, PARCOR analysis, LSP analysis, PARCOR synthesis filter, LSP synthesis filter, vector quantization techniques, and other extended versions of them. This software is released under the Modified BSD license. SPTK was developed and has been used in the research group of Prof. Satoshi Imai (he has retired) and Prof. Takao Kobayashi (currently he is with Interdisciplinary Graduate School of Science and Engineering, Tokyo Institute of Technology) at P&I laboratory, Tokyo Institute of Technology. A sub-set of tools was chosen and arranged for distribution by Prof. Keiichi Tokuda (currently he is with Department of Computer Science and Engineering, Nagoya Institute of Technology) as a coordinator in cooperation and other collaborates (see "Acknowledgments" and "Who we are" in README).The original source codes have been written by many people who took part in activities of the research group. The most original source codes of this distribution were written by Takao Kobayashi (graph, data processing, FFT, sampling rate conversion, etc.), Keiichi Tokuda (speech analysis, speech synthesis, etc.), and Kazuhito Koishida (LSP, vector quantization, etc.).
julius
- https://github.com/julius-speech/julius - Open-Source Large Vocabulary Continuous Speech Recognition Engine
Services
WaveNet
to sort
- Sequence Modeling with CTC - A visual guide to Connectionist Temporal Classification, an algorithm used to train deep neural networks in speech recognition, handwriting recognition and other sequence problems. [4]
- YouTube: Emily Shea - "Perl Out Loud"
Kaldi
- Kaldi - a toolkit for speech recognition, intended for use by speech recognition researchers and professionals. [6]
espnet
- https://github.com/espnet/espnet - an end-to-end speech processing toolkit, mainly focuses on end-to-end speech recognition and end-to-end text-to-speech. ESPnet uses chainer and pytorch as a main deep learning engine, and also follows Kaldi style data processing, feature extraction/format, and recipes to provide a complete setup for speech recognition and other speech processing experiments.
voice2json
- voice2json - a collection of command-line tools for offline speech/intent recognition on Linux. It is free, open source, and supports 16 languages.
sak
- https://github.com/fredvs/sak - Speecher Assistive Kit. With sak, your application becomes assistive directly, without changing anything in your code. sak uses the PortAudio and eSpeak Open Source Libraries.
Respeaker
- https://github.com/respeaker - To build voice interface objects
- https://github.com/respeaker/get_started_with_respeaker - This is the wiki of ReSpeaker Core V2, ReSpeaker Core and ReSpeaker Mic Array.
- https://github.com/respeaker/respeakerd - server application for the microphone array solutions of SEEED, based on librespeaker which combines the audio front-end processing algorithms.
- https://github.com/respeaker/respeaker_python_library - To build voice enabled objects/applications with Python and ReSpeaker
- https://github.com/respeaker/seeed-voicecard - 2 Mic Hat, 4 Mic Array, 6-Mic Circular Array Kit, and 4-Mic Linear Array Kit for Raspberry Pi
DeepSpeech
- https://github.com/mozilla/DeepSpeech - an open source Speech-To-Text engine, using a model trained by machine learning techniques based on Baidu's Deep Speech research paper. Project DeepSpeech uses Google's TensorFlow to make the implementation easier.
Voice control
Simon
- Simon - an open source speech recognition program that can replace your mouse and keyboard. The system is designed to be as flexible as possible and will work with any language or dialect.
Blather
- Blather is a speech recognizer that will run commands when a user speaks preset sentences.
Xvoice
- Xvoice - enables continuous speech dictation and speech control of most X applications. To convert users' speech into text it uses the IBM ViaVoice speech recognition engine, which is distributed separately (see below).When in dictation mode Xvoice passes this text directly to the currently focused X application. When in command mode, Xvoice matches the speech with predefined, user-modifieable, key sequences or commands. For instance "list" would match "ls -l" when commanding the console, so that when the user says "list" "ls -l" will be sent to the console as if the user had typed it.
CVoiceControl
- CVoiceControl - a speech recognition system that enables a user to connect spoken commands to unix commands. It automagically detects speech input from a microphone, performs recognition on this input and - in case of successful recognition - executes the associated unix command.
Perlbox Voice
- Perlbox Voice - an voice enabled application to bring your desktop under your command. With a single word, you can start your web browser, your favorite editor or whatever you want. Perlbox Voice now has desktop plugins, which will allow you to control desktop functions. You can switch virtual desktops, invoke the desktop menu, switch wallpaper or lock the screen. This is command, and it's right on the tip of your tongue! Start with the tutorial, which has a lot of screenshots.
Voximp
- https://github.com/arrdem/Voicely - A voice command interface for Linux, and a fork of the Voximp project
Analysis
ESPS
- ESPS - Entropic Signal Processing System, is a package of UNIX-like commands and programming libraries for speech signal processing. As a commercial product of Entropic Research Laboratory, Inc, it became extremely widely used in phonetics and speech technology research laboratories in the 1990's, in view of the wide range of functions it offered, such as get_f0 (for fundamental frequency estimation), formant (for formant frequency measurement), the xwaves graphical user interface, and many other commands and utilities. Following the acquisition of Entropic by Microsoft in 1999, Microsoft and AT&T licensed ESPS to the Centre for Speech Technology at KTH, Sweden, so that a final legacy version of the ESPS source code could continue to be made available to speech researchers. At KTH, code from the ESPS library (such as get_f0) was incorporated by Kåre Sjölander and Jonas Beskow into the Wavesurfer speech analysis tool. This is a very good alternative way to use many ESPS functions if you want a graphical user interface rather than scripting.
- https://github.com/jeremysalwen/ESPS - This archive contains source files from the ESPS toolkit.
NICO toolkit
- NICO toolkit - mainly intended for, and originally developed for speech recognition applications, a general purpose toolkit for constructing artificial neural networks and training with the back-propagation learning algorithm. The network topology is very flexible. Units are organized in groups and the group is a hierarchical structure, so groups can have sub-groups or other objects as members. This makes it easy to specify multi-layer networks with arbitrary connection structure and to build modular networks.
Speech Research Tools
- https://sourceforge.net/projects/speechresearch - Software for speech research. It includes programs and libraries for signal processing, along with general purpose scientific libraries. Most of the code is in Python, with C/C++ supporting code. Also, contains code releases corresponding to publishe
HTK Speech Recognition Toolkit
- HTK Speech Recognition Toolkit - a portable toolkit for building and manipulating hidden Markov models. HTK is primarily used for speech recognition research although it has been used for numerous other applications including research into speech synthesis, character recognition and DNA sequencing. HTK is in use at hundreds of sites worldwide.HTK consists of a set of library modules and tools available in C source form. The tools provide sophisticated facilities for speech analysis, HMM training, testing and results analysis. The software supports HMMs using both continuous density mixture Gaussians and discrete distributions and can be used to build complex HMM systems. The HTK release contains extensive documentation and examples.
speechrate
- speechrate - software for the analysis of speech. Below you will find a script that automatically detects syllable nuclei in order to measure speech rate without the need of a transcription. Peaks in intensity (dB) that are preceded and followed by dips in intensity are considered as potential syllable nuclei. The script subsequently discards peaks that are not voiced. On this page you find an example of how the script works.
HAT
- Higgins Annotation Tool - can be used to transcribe and annotate speech with one or more audio tracks (such as dialogue). Windows.
Synthesis
to sort/categorise
- https://archive.org/details/flexibleformants00lalw - Flexible formant synthesizer : a tool for improving speech production quality
- https://archive.org/details/flexiblehighqual00hsie - A flexible and high quality articulatory speech synthesizer
SAM
- https://en.wikipedia.org/wiki/Software_Automatic_Mouth - or SAM, is a speech synthesis program developed and sold by Don’t Ask Software. The program was released for the Apple II, Lisa, Atari 8-bit family, and Commodore 64.
rsynth
- rsynth - Text-to-Speech.
Festival
- Festival - or The Festival Speech Synthesis System, offers a general framework for building speech synthesis systems as well as including examples of various modules. As a whole it offers full text to speech through a number APIs: from shell level, though a Scheme command interpreter, as a C++ library, from Java, and an Emacs interface. Festival is multi-lingual (currently English (British and American), and Spanish) though English is the most advanced. Other groups release new languages for the system.
- Festvox - aims to make the building of new synthetic voices more systemic and better documented, making it possible for anyone to build a new voice.
Rocaloid
- Rocaloid - a free, open-source singing voice synthesis system. Its ultimate goal is to fast synthesize natural, flexible and multi-lingual vocal parts. Like other vocal synthesizing software, after installing the vocal database, inputting lyrics and pitch, you can synthesize attractive vocal parts. What’s more, Rocaloid highlights on providing you more controllable parameters which enabling to take control of exquisite dimensions of the synthesized voice and export with better quality. By using a fully constructed Rocaloid Database, you can synthesize singing voice in any phonetic-based languages.
- https://github.com/Rocaloid - dead?
Festvox
- Festvox - aims to make the building of new synthetic voices more systemic and better documented, making it possible for anyone to build a new voice. Specifically we offer: Documentation, including scripts explaining the background and specifics for building new voices for speech synthesis in new and supported languages. Example speech databases to help building new voices. Links, demos and a repository for new voices. This work is firmly grounded within Edinburgh University's Festival Speech Synthesis System and Carnegie Mellon University's small footprint Flite synthesis engine.
MaryTTS
- MaryTTS is an open-source, multilingual Text-to-Speech Synthesis platform written in Java. It was originally developed as a collaborative project of DFKI’s Language Technology Lab and the Institute of Phonetics at Saarland University. It is now maintained by the Multimodal Speech Processing Group in the Cluster of Excellence MMCI and DFKI.
eSpeak
- eSpeak - a compact open source software speech synthesizer for English and other languages, for Linux and Windows. eSpeak uses a "formant synthesis" method. This allows many languages to be provided in a small size. The speech is clear, and can be used at high speeds, but is not as natural or smooth as larger synthesizers which are based on human speech recordings.
- https://github.com/divVerent/ecantorix - a singing synthesis frontend for espeak. It works by using espeak to generate raw speech samples, then adjusting their pitch and length and finally creating a LMMS project file referencing the samples in sync to the input file.
OpenSource SpeechSynth
MBROLA
- http://tcts.fpms.ac.be/synthesis/ - The MBROLA Project
Assistive Context-Aware Toolkit
- Assistive Context-Aware Toolkit (ACAT) - an open source platform developed at Intel Labs to enable people with motor neuron diseases and other disabilities to have full access to the capabilities and applications of their computers through very constrained interfaces suitable for their condition. More specifically, ACAT enables users to easily communicate with others through keyboard simulation, word prediction and speech synthesis. Users can perform a range of tasks such as editing, managing documents, navigating the Web and accessing emails. ACAT was originally developed by researchers at Intel Labs for Professor Stephen Hawking, through a very iterative design process over the course of three years.
Praat
- Praat - doing phonetics by computer
Gnuspeech
- gnuspeech - makes it easy to produce high quality computer speech output, design new language databases, and create controlled speech stimuli for psychophysical experiments. gnuspeechsa is a cross-platform module of gnuspeech that allows command line, or application-based speech output. The software has been released as two tarballs that are available in the project Downloads area of http://savannah.gnu.org/projects/gnuspeech. [8]
Project Merlin
- Project Merlin - A truly free virtual singer, no matter how you want to send all kinds of ideas. [9]
- https://github.com/ProjectMeilin - not fully open yet?
- http://www.cstr.ed.ac.uk/projects/merlin/
- http://ml.cs.yamanashi.ac.jp/world/english
- YouTube: UTAU】Honeyworks ママ ver.acoustic 【徵音梅林cover】
- YouTube: 【徴音梅林】Umbrella カバー
UTAU
- https://en.wikipedia.org/wiki/Utau - a Japanese singing synthesizer application created by Ameya/Ayame. This program is similar to the Vocaloid software, with the difference that it is shareware instead of being released under third party licensing
- https://github.com/stakira/OpenUtau - Open source UTAU editing environment.
Sinsy
- Sinsy - HMM-based Singing Voice Synthesis System
- Sinsy - Singing Voice Synthesizer - how to
- https://github.com/hyperzlib/Sinsy-Remix - The HMM-Based Singing Voice Syntheis System Remix "Sinsy-r"
Mozilla TTS
- https://github.com/mozilla/TTS - Deep learning for Text to Speech
CMU Flite
- CMU Flite - a small, fast run-time open source text to speech synthesis engine developed at CMU and primarily designed for small embedded machines and/or large servers. Flite is designed as an alternative text to speech synthesis engine to Festival for voices built using the FestVox suite of voice building tools.
mesing
Adobe VoCo
- https://en.wikipedia.org/wiki/Adobe_Voco
- https://arstechnica.com/information-technology/2016/11/adobe-voco-photoshop-for-audio-speech-editing/ [10]
VST Speek
char2wav
loop
IPOX
- IPOX - an experimental, all-prosodic speech synthesizer, developed many years ago by Arthur Dirksen and John Coleman. It is still available for downloading, and was designed to run on a 486 PC running Windows 3.1 or higher, with a 16-bit Windows-compatible sound card, such as the Soundblaster 16. It still seems to run on e.g. XP, but I haven't tried it on Vista.
NPSS
- https://github.com/seaniezhao/torch_npss - pytorch implementation of Neural Parametric Singing Synthesizer 歌声合成
Pink Trombone
- Pink Trombone - Bare-handed procedural speech synthesis, version 1.1, March 2017, by Neil Thapen
Klatter
- https://github.com/fundamental/klatter - a bare bones formant synthesizer based upon the description given in the 1979 paper "Software For a Cascade/Parallel Formant Synthesizer" by Dennis Klatt. This program was not designed for interactive use, though there is code for some minimal midi control. In it's current state, it is enough of a curiosity that it will be preserved, though it may not see much if any use.
Tacotron 2
- https://github.com/Rayhane-mamah/Tacotron-2 - DeepMind's Tacotron-2 Tensorflow implementation
- https://github.com/NVIDIA/tacotron2 - Tacotron 2 - PyTorch implementation with faster-than-realtime inference
Real-Time-Voice-Cloning
- https://github.com/CorentinJ/Real-Time-Voice-Cloning - Clone a voice in 5 seconds to generate arbitrary speech in real-time
leesampler
- https://github.com/GloomyGhost-MosquitoSeal/lessampler - a Singing Voice Synthesizer
Neural Parametric Singing Synthesizer
VoiceOfFaust
- https://github.com/magnetophon/VoiceOfFaust - Turn your voice into a synthesizer!
tomomibot
- https://github.com/adzialocha/tomomibot - Artificial intelligence bot for live voice improvisation.