Open Source ChromeOS Speech Software

Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players.

FreeTTS is a speech synthesis engine written entirely in the Java(tm) programming language. FreeTTS was written by the Sun Microsystems Laboratories Speech Team and is based on CMU's Flite engine. FreeTTS also includes a partial JSAPI 1.0

SPTK is 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.

mp3 library, advanced ID3V1 and ID3V2 tagger, player. Organize a large mp3 library, over 40,000 songs. Speech synthesis and tag backup utilities. Scripts to maintain and organize song files.

PersonaPlex is an open-source real-time conversational speech AI model that goes beyond traditional text chat by providing full-duplex speech-to-speech interaction, meaning it can listen and talk at the same time instead of waiting for you to finish speaking before responding. This architectural approach eliminates awkward pauses and makes conversations feel much more human-like, with natural behaviors such as overlapping speech, interruptions, and fluent turn-taking, traits that traditional AI assistants typically lack. PersonaPlex also supports persona and voice control, allowing developers to define the role and speaking style of the agent using text prompts and voice conditioning, making it suitable for applications like customized voice assistants, interactive character agents, or domain-specific conversational tools. Internally, it processes continuous audio streams in a hybrid input format so that speech understanding and generation occur jointly.

A speech synthesis and recognition library that is cross-platform, accessible from Java and C++, and has a very small API. Uses CMU Sphinx4 and FreeTTS internally.

Source code from the Research Institute for Signals, Systems and Computational Intelligence http://fich.unl.edu.ar/sinc

Translating subtitles in format SubRip from one natural language to other. It is based on Google Translate without API and therefore without payment. Translator have automatic and manual spell checkers.

odt2daisy is an OpenOffice.org Writer extension, enabling to export in DAISY XML, Full DAISY (xml+audio) and Audiobook format. DAISY is an NISO Z39.86 standard for blind, visual impaired, print-disabled, and learning-disabled people.

A modular, extensible Hebrew text-to-speech engine tuned for Standard Israeli Hebrew, and associated tools.

'Text to Voice' or 'Text to Speech' is 1 of the coolest Firefox add-ons. It gives ur brwsr the pwr of speech. Select txt, clck the bttn on the bttm rite & this add-on spks the selectd txt 4 u. Isn't it brllant? Moreovr odio file cn b dnloaded as

MARF is a general cross-platform framework with a collection of algorithms for audio (voice, speech, and sound) and natural language text analysis and recognition along with sample applications (identification, NLP, etc.) of its use, implemented in Java.

Development of the Italian Version of FESTIVAL Text to Speech synthesis system

Text-to-Speech conversor for Basque and Spanish. It includes linguistic processing and built voices for the languages aforementioned. Its acoustic engine is based on hts_engine and it uses a high quality vocoder called AhoCoder. Developed by Aholab Signal Processing Laboratory: https://aholab.ehu.es/aholab/ http://aholab.ehu.es/ahocoder/

Performs actions on detected volume threshold Examples : - Launch music on clap - Launch speech recording when you start speaking - Launch guard webcam when a significant sound is detected - Increase or decrease headphones volume when ambient noise pass

Simple program to assist transcriptions from recorded interviews. Resembles somewhat the early F4. <br/>Keywords: transcription transcribe interview

ASR-Builder provides an easy-to-use interface to the HTK toolkit, that allows users to build ASR systems. ASR-Builder provides a platform that performs house-keeping tasks when using HTK and also provides default training/testing/recognition scripts.

AarTon is an automated text-to-speech application. It allows user to enter text in a web-based front-end and render these texts via a multi-channel sound card.

Text-to-Speech conversor for Basque, Spanish, Catalan, Galician and English. It includes linguistic processing and built voices for all the languages aforementioned. Its acoustic engine is based on hts_engine and it uses a high quality vocoder called AhoCoder. Developed by Aholab Signal Processing Laboratory: https://aholab.ehu.es/aholab/ http://aholab.ehu.es/ahocoder/

audacity-extra now provides a sleek dark themed version of the Audacity open source sound editor. The project experiments with Audacity variations. There's a vowel-sound target-practice display for language learners and an analog waveform data logger for embedded systems.

Auvai is a Java API and Java Swing based application for Text to Speech conversion of Unicode Tamil. Future direction of this API and application is to support Text to Speech conversion for all "Indic" languages.

BladeWareVXML is a portable VoiceXML 2.1 interpreter that is an enhanced version (performance, usability and integration) of OpenVXI. A commercial version, with documentation, sample code, and support options, is available from the Commetrex Website.

CJ7 is an open-source speech recognition engine.

Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.

Cowpie is an application for synchronizing character actions with sound files and animation tools like Blender and 3DS Max. Such actions include the phonemes of speech and facial expressions. Cowpie will also coordinate speech for multiple characters.