Abstract
Performance of automatic speech recognition systems trained on close-talking data suffers when used in a distant-talking environment due to the mismatch in training and testing conditions. Microphone array sound capture can reduce some mismatch by removing ambient noise and reverberation but offers insufficient improvement in performance. However, using array signal capture in conjunction with Hidden Markov Model (HMM) adaptation on the clean-speech models can result in improved recognition accuracy. This paper describes an experiment in which the output of an 8-element microphone array system using MFA processing is used for speech recognition with LT-MLLR adaptation. The recognition is done in two passes. In the first pass, an HMM trained on clean data is used to recognize the speech. Using the results of this pass, the HMM model is adapted to the environment using the LT-MLLR algorithm. This adapted model, a product of MFA and LT-MLLR, results in improved recognition performance.
Original language | English |
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Pages (from-to) | 777-780 |
Number of pages | 4 |
Journal | ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings |
Volume | 2 |
Publication status | Published - 1999 |
Externally published | Yes |
Event | Proceedings of the 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP-99) - Phoenix, AZ, USA Duration: 1999 Mar 15 → 1999 Mar 19 |
ASJC Scopus subject areas
- Software
- Signal Processing
- Electrical and Electronic Engineering