Title :
An analog VLSI architecture for auditory based feature extraction
Author :
Kumar, Nagendra ; Himmelbauer, Wolfgang ; Cauwenberghs, Gert ; Andreou, Andreas G.
Author_Institution :
Center for Language & Speech Process., Johns Hopkins Univ., Baltimore, MD, USA
Abstract :
We have developed a low power analog VLSI chip for real time signal processing motivated by the principles of the human auditory system. An analog cochlear filter bank (which is implemented on the chip) decomposes the input audio signal into several frequency bands that have almost equal bandwidth on a log scale. This step is thus similar to computing the wavelet transform. The chip then computes signal energies and zero crossing time intervals of frequency components in a cochlear filter bank. The chip is intended to work as a front-end of a speech recognition system. We include experimental results on a VLSI implementation of the auditory front-end. We present speech recognition results on the TI-DIGITS database obtained from computer simulations which model the functionality of the feature extraction VLSI hardware. We use hidden Markov models (HMM) in combination with linear discriminant analysis (LDA) for the recognizer design
Keywords :
MOS integrated circuits; VLSI; analogue integrated circuits; band-pass filters; ear; feature extraction; hearing; hidden Markov models; speech processing; speech recognition; MOSIS; TI-DIGITS database; analog VLSI architecture; analog cochlear filter bank; auditory based feature extraction; auditory front-end; bandwidth; computer simulations; experimental results; frequency bands; frequency components; hidden Markov models; human auditory system; input audio signal; linear discriminant analysis; low power analog VLSI chip; real time signal processing; signal energies; speech recognition system; wavelet transform; zero crossing time intervals; Feature extraction; Filter bank; Frequency; Hidden Markov models; Humans; Linear discriminant analysis; Real time systems; Signal processing; Speech recognition; Very large scale integration;
Conference_Titel :
Acoustics, Speech, and Signal Processing, 1997. ICASSP-97., 1997 IEEE International Conference on
Conference_Location :
Munich
Print_ISBN :
0-8186-7919-0
DOI :
10.1109/ICASSP.1997.604843