DocumentCode
310531
Title
Model adaptation based on HMM decomposition for reverberant speech recognition
Author
Takiguchi, Tetsuya ; Nakamura, Satoshi ; Qiang Hou ; Shikano, Kiyahiro
Author_Institution
Graduate Sch. of Inf., Nara Inst. of Sci. & Technol., Japan
Volume
2
fYear
1997
fDate
21-24 Apr 1997
Firstpage
827
Abstract
The performance of a speech recognizer is degraded drastically in reverberant environments. The authors propose a novel algorithm which can model an observation signal by composition of HMMs of clean speech, noise and an acoustic transfer function. However, estimating HMM parameters of the acoustic transfer function is still a serious problem. In their previous paper, they measured real impulse responses of training positions in an experiment room. It is inconvenient and unrealistic to measure impulse responses for every possible new experiment room. The paper presents a new method for estimating HMM parameters of the acoustic transfer function from some adaptation data by using an HMM decomposition algorithm which is an inverse process of the HMM composition. Its effectiveness is confirmed by a series of speaker dependent and independent word recognition experiments on simulated distant-talking speech data
Keywords
acoustic signal processing; hidden Markov models; parameter estimation; speech recognition; transfer functions; acoustic transfer function; experiment room; hidden Markov model decomposition; model adaptation; observation signal modelling; parameter estimation; real impulse response measurement; reverberant environments; reverberant speech recognition; simulated distant-talking speech data; speaker dependent word recognition experiments; speaker independent word recognition experiments; speech recognizer; training positions; Acoustic measurements; Acoustic noise; Adaptation model; Degradation; Hidden Markov models; Parameter estimation; Speech enhancement; Speech recognition; Transfer functions; Working environment noise;
fLanguage
English
Publisher
ieee
Conference_Titel
Acoustics, Speech, and Signal Processing, 1997. ICASSP-97., 1997 IEEE International Conference on
Conference_Location
Munich
ISSN
1520-6149
Print_ISBN
0-8186-7919-0
Type
conf
DOI
10.1109/ICASSP.1997.596060
Filename
596060
Link To Document