Superdirectional microphone arrays
Acoustic signal processing for telecommunication
Convolutive blind separation of speech mixtures using the natural gradient
Speech Communication - Special issue on speech processing for hearing aids
ICASSP '94 Proceedings of the Acoustics, Speech, and Signal Processing,1994. on IEEE International Conference - Volume 04
Subspace methods for multimicrophone speech dereverberation
EURASIP Journal on Applied Signal Processing
EURASIP Journal on Applied Signal Processing
Dereverberation by using time-variant nature of speech production system
EURASIP Journal on Advances in Signal Processing
Adaptive dereverberation of speech signals with speaker-position change detection
ICASSP '09 Proceedings of the 2009 IEEE International Conference on Acoustics, Speech and Signal Processing
Integrated speech enhancement method using noise suppression and dereverberation
IEEE Transactions on Audio, Speech, and Language Processing
EVAM: an eigenvector-based algorithm for multichannel blinddeconvolution of input colored signals
IEEE Transactions on Signal Processing
Prediction error method for second-order blind identification
IEEE Transactions on Signal Processing
IEEE Transactions on Audio, Speech, and Language Processing
IEEE Transactions on Audio, Speech, and Language Processing
Robust Speech Dereverberation Using Multichannel Blind Deconvolution With Spectral Subtraction
IEEE Transactions on Audio, Speech, and Language Processing
An Improved Method for Late-Reverberant Suppression Based on Statistical Model
Speech Communication
Online Speech Dereverberation Algorithm Based on Adaptive Multichannel Linear Prediction
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Hi-index | 0.00 |
This paper proposes a statistical model-based speech dereverberation approach that can cancel the late reverberation of a reverberant speech signal captured by distant microphones without prior knowledge of the room impulse responses. With this approach, the generative model of the captured signal is composed of a source process, which is assumed to be a Gaussian process with a time-varying variance, and an observation process modeled by a delayed linear prediction (DLP). The optimization objective for the dereverberation problem is derived to be the sum of the squared prediction errors normalized by the source variances; hence, this approach is referred to as variance-normalized delayed linear prediction (NDLP). Inheriting the characteristic of DLP, NDLP can robustly estimate an inverse system for late reverberation in the presence of noise without greatly distorting a direct speech signal. In addition, owing to the use of variance normalization, NDLP allows us to improve the dereverberation result especially with relatively short (of the order of a few seconds) observations. Furthermore, NDLP can be implemented in a computationally efficient manner in the time-frequency domain. Experimental results demonstrate the effectiveness and efficiency of the proposed approach in comparison with two existing approaches.