Search Results for "microphone array"

BTK contains C++ and Python libraries that implement speech processing and microphone array techniques such as speech feature extraction, speech enhancement, speaker tracking, beamforming, dereverberation and echo cancellation algorithms. The Millennium ASR provides C++ and python libraries for automatic speech recognition. The Millennium ASR implements a weighted finite state transducer (WFST) decoder, training and adaptation methods.

ManyEars implements real-time microphone array processing to perform sound source localisation, tracking and separation. It was designed for mobile robot audition in dynamic environments. NOTE: Development will continue on github : https://github.com/introlab/manyears

Nonlinearly change frequencies and echos for live music by CPU. I found a way to normalize 1d wavefunction amplitude so this hack and its random heat vibrations are still unitary, even while microphone vibrating adds energy to part of 1d string of position and speed scalar arrays. The sparse part is, while the arrays are perfectly dense and linear, time is sparse when some springs vibrate with a larger multiplier of position subtracted from speed. In other words, this hack is a quanta level discontinuous field but in theory may be continuous as change in natural resonance frequency (what part of the "guitar string" would vibrate as if nothing acted on it) These few kilobytes of java code, many versions of CochleaSim.java and JSoundCard0.5 as an easily replaced dependency that reads microphone and writes to speakers as numbers ranging -1 to 1 44100 times per second, which is where SparseDoppler hooks in its array, microphone at one end and speakers hear the sum of the whole string.

When generating audio, why use sine waves and standard effects? Directly tell the sound-card how much electricity should be in the speaker/microphone wires many times per second as numbers from -1 to 1. Automatically finds good sound-card options and balances between linear interpolation speeds of consuming microphone buffer (and then any Java transform function per sample) and producing into speakers buffer and to keep delay between them small. Its really simple you only have to write 1 function that takes an array parameter. ...