Time difference of arrival estimation of speech source in a noisy and reverberant environment
Signal Processing - Content-based image and video retrieval
Performance analysis of dual source transfer-function generalized sidelobe canceller
Speech Communication
Noise cancellation with static mixtures of a nonstationary signal and stationary noise
EURASIP Journal on Applied Signal Processing
An integrated real-time beamforming and postfiltering system for nonstationary noise environments
EURASIP Journal on Applied Signal Processing
Subspace methods for multimicrophone speech dereverberation
EURASIP Journal on Applied Signal Processing
Signal processing in high-end hearing aids: state of the art, challenges, and future trends
EURASIP Journal on Applied Signal Processing
An improved array steering vector estimation method and its application in speech enhancement
EURASIP Journal on Applied Signal Processing
Sector-based detection for hands-free speech enhancement in cars
EURASIP Journal on Applied Signal Processing
A robust statistical-based speaker's location detection algorithm in a vehicular environment
EURASIP Journal on Applied Signal Processing
A Speech Enhancement Method in Subband
ISNN '07 Proceedings of the 4th international symposium on Neural Networks: Advances in Neural Networks, Part III
IEEE Transactions on Audio, Speech, and Language Processing
Convolutive transfer function generalized sidelobe canceler
IEEE Transactions on Audio, Speech, and Language Processing
On optimal frequency-domain multichannel linear filtering for noise reduction
IEEE Transactions on Audio, Speech, and Language Processing
A novel psychoacoustically motivated multichannel speech enhancement system
COST 2102'07 Proceedings of the 2007 COST action 2102 international conference on Verbal and nonverbal communication behaviours
New insights into the MVDR beamformer in room acoustics
IEEE Transactions on Audio, Speech, and Language Processing
Noninvertible gabor transforms
IEEE Transactions on Signal Processing
International Journal of Computer Applications in Technology
Speech enhancement using Gaussian scale mixture models
IEEE Transactions on Audio, Speech, and Language Processing
A study of the LCMV and MVDR noise reduction filters
IEEE Transactions on Signal Processing
A subband adaptive learning algorithm for microphone array based speech enhancement
ISNN'05 Proceedings of the Second international conference on Advances in neural networks - Volume Part II
Robust Adaptive Microphone Array with Mainlobe and Response Ripple Control
Journal of Signal Processing Systems
Multi-channel noise reduction in noisy environments
ISCSLP'06 Proceedings of the 5th international conference on Chinese Spoken Language Processing
A stereophonic acoustic signal extraction scheme for noisy and reverberant environments
Computer Speech and Language
Performance Study of the MVDR Beamformer as a Function of the Source Incidence Angle
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Generalized Spherical Array Beamforming for Binaural Speech Reproduction
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Distributed Delay and Sum Beamformer for Speech Enhancement via Randomized Gossip
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Location Feature Integration for Clustering-Based Speech Separation in Distributed Microphone Arrays
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Hi-index | 35.69 |
We consider a sensor array located in an enclosure, where arbitrary transfer functions (TFs) relate the source signal and the sensors. The array is used for enhancing a signal contaminated by interference. Constrained minimum power adaptive beamforming, which has been suggested by Frost (1972) and, in particular, the generalized sidelobe canceler (GSC) version, which has been developed by Griffiths and Jim (1982), are the most widely used beamforming techniques. These methods rely on the assumption that the received signals are simple delayed versions of the source signal. The good interference suppression attained under this assumption is severely impaired in complicated acoustic environments, where arbitrary TFs may be encountered. In this paper, we consider the arbitrary TF case. We propose a GSC solution, which is adapted to the general TF case. We derive a suboptimal algorithm that can be implemented by estimating the TFs ratios, instead of estimating the TFs. The TF ratios are estimated by exploiting the nonstationarity characteristics of the desired signal. The algorithm is applied to the problem of speech enhancement in a reverberating room. The discussion is supported by an experimental study using speech and noise signals recorded in an actual room acoustics environment