Robust automatic speech recognition with missing and unreliable acoustic data
Speech Communication
Blind Model Selection for Automatic Speech Recognition in Reverberant Environments
Journal of VLSI Signal Processing Systems
Discrete-time speech signal processing: principles and practice
Discrete-time speech signal processing: principles and practice
Harmonicity-Based Blind Dereverberation for Single-Channel Speech Signals
IEEE Transactions on Audio, Speech, and Language Processing
A two-stage algorithm for one-microphone reverberant speech enhancement
IEEE Transactions on Audio, Speech, and Language Processing
IEEE Transactions on Audio, Speech, and Language Processing - Special issue on processing reverberant speech: methodologies and applications
Model-based feature enhancement for reverberant speech recognition
IEEE Transactions on Audio, Speech, and Language Processing - Special issue on processing reverberant speech: methodologies and applications
Computer Speech and Language
Uncertainty-based learning of acoustic models from noisy data
Computer Speech and Language
An Improved Method for Late-Reverberant Suppression Based on Statistical Model
Speech Communication
Structural Bayesian Linear Regression for Hidden Markov Models
Journal of Signal Processing Systems
Hi-index | 0.00 |
The performance of automatic speech recognition is severely degraded in the presence of noise or reverberation. Much research has been undertaken on noise robustness. In contrast, the problem of the recognition of reverberant speech has received far less attention and remains very challenging. In this paper, we use a dereverberation method to reduce reverberation prior to recognition. Such a preprocessor may remove most reverberation effects. However, it often introduces distortion, causing a dynamic mismatch between speech features and the acoustic model used for recognition. Model adaptation could be used to reduce this mismatch. However, conventional model adaptation techniques assume a static mismatch and may therefore not cope well with a dynamic mismatch arising from dereverberation. This paper proposes a novel adaptation scheme that is capable of managing both static and dynamic mismatches. We introduce a parametric model for variance adaptation that includes static and dynamic components in order to realize an appropriate interconnection between dereverberation and a speech recognizer. The model parameters are optimized using adaptive training implemented with the Expectation Maximization algorithm. An experiment using the proposed method with reverberant speech for a reverberation time of 0.5 s revealed that it was possible to achieve an 80% reduction in the relative error rate compared with the recognition of dereverberated speech (word error rate of 31%), and the final error rate was 5.4%, which was obtained by combining the proposed variance compensation and MLLR adaptation.