DocumentCode
2791497
Title
Experimental studies on continuous speech recognition using neural architectures with “adaptive” hidden activation functions
Author
Siniscalchi, Sabato Marco ; Svendsen, Tørbjrn ; Sorbello, Filippo ; Lee, Chin-Hui
Author_Institution
Dept. of Electron. & Telecommun., NTNU, Trondheim, Norway
fYear
2010
fDate
14-19 March 2010
Firstpage
4882
Lastpage
4885
Abstract
The choice of hidden non-linearity in a feed-forward multi-layer perceptron (MLP) architecture is crucial to obtain good generalization capability and better performance. Nonetheless, little attention has been paid to this aspect in the ASR field. In this work, we present some initial, yet promising, studies toward improving ASR performance by adopting hidden activation functions that can be automatically learned from the data and change shape during training. This adaptive capability is achieved through the use of orthonormal Hermite polynomials. The “adaptive” MLP is used in two neural architectures that generate phone posterior estimates, namely, a standalone configuration and a hierarchical structure. The posteriors are input to a hybrid phone recognition system with good results on the TIMIT corpus. A scheme for optimizing the contributions of high-accuracy neural architectures is also investigated, resulting in a relative improvement of ~9.0% over a non-optimized combination. Finally, initial experiments on the WSJ Nov92 task show that the proposed technique scales well up to large vocabulary continuous speech recognition (LVCSR) tasks.
Keywords
maximum likelihood estimation; multilayer perceptrons; polynomials; speech recognition; transfer functions; vocabulary; MLP; adaptive hidden activation functions; feedforward multilayer perceptron; hybrid phone recognition system; neural architectures; orthonormal Hermite polynomials; phone posterior estimation; vocabulary continuous speech recognition; Automatic speech recognition; Computer architecture; Feedforward systems; Hidden Markov models; Neural networks; Neurons; Polynomials; Shape; Speech recognition; Training data; Neural networks; Speech recognition;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics Speech and Signal Processing (ICASSP), 2010 IEEE International Conference on
Conference_Location
Dallas, TX
ISSN
1520-6149
Print_ISBN
978-1-4244-4295-9
Electronic_ISBN
1520-6149
Type
conf
DOI
10.1109/ICASSP.2010.5495120
Filename
5495120
Link To Document