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.

Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate for a either a sound input(microphone) or sound output(speakers). Can also be used to gate noise from your own mic or play your microphone through your speakers. REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower.

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.

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.

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

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

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.

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.

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.

Speech based User Interface Components Library for Java is a project to create Java controls and applications that can be used not only by literate people but also by non-literates. Speech and visual element with minimal text is used to create components

'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

Scalable Language API (SLAPI) The most comprehensive architecture for conversational natural-language applications including speech recognition/synthesis, semantics, & machine translation. Used on Android & other mobile app platforms.

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/

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

ctypes Python module for eSpeak http://espeak.sf.net speech synthesis

Project c2h - cetacean to human - building Seadragon, a tool for the scientific research of the acoustic communication of cetaceans, supporting the creation, emission, and recognition of underwater whistles. The blog: http://leafyseadragon.blogspot.com/

This project provide hidden Markov model speech recognition system by using Matlab/Simulink/Stateflow.

InproTK is an Incremental Spoken Dialogue Processing Toolkit, that is, a toolkit to help you build dialogue systems that listen and talk incrementally, allowing for advanced interactional behaviour. Please see our Wiki for more information: http://sourceforge.net/p/inprotk/wiki/

A program to record and sequentially play audio spots in e.g. supermarkets. This project is now on github: http://github.com/pryds/spotmachine

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.

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/

GibPhone is a highly extensible VoIP/IM client for the .NET framework that uses a powerful plugin engine to allow for UI extensions and any call control stack / media payload / transport protocol combination.

An initiative to create something similar to the windows program Roger Wilco, Teamspeak, BattleCom and Speak Freely, allowing users from different platforms talk with each other in real time with minimal CPU and bandwidth usage. Voice chat.....

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.

This project is being developed to be a Java based speech recognition (SR) program. In addition to the SR program itself, it includes a program which allows a user to view the sound being received by the computer. The user can manipulate this data.