Wearable microphone array as user interface
AUIC '04 Proceedings of the fifth conference on Australasian user interface - Volume 28
Development of a compact two-sensor directional audio-capturing device
Signal Processing
Signal processing for in-car communication systems
Signal Processing
An acoustic human-machine front-end for multimedia applications
EURASIP Journal on Applied Signal Processing
Sector-based detection for hands-free speech enhancement in cars
EURASIP Journal on Applied Signal Processing
EURASIP Journal on Applied Signal Processing
Speech enhancement by multichannel crosstalk resistant ANC and improved spectrum subtraction
EURASIP Journal on Applied Signal Processing
Detection and separation of speech events in meeting recordings using a microphone array
EURASIP Journal on Audio, Speech, and Music Processing
Performance evaluation of adaptive dual microphone systems
Speech Communication
Convolutive transfer function generalized sidelobe canceler
IEEE Transactions on Audio, Speech, and Language Processing
On spatial aliasing in microphone arrays
IEEE Transactions on Signal Processing
OFDM for cognitive radio: merits and challenges
IEEE Wireless Communications
Linear programming algorithms for sparse filter design
IEEE Transactions on Signal Processing
Conventional beamformer using post-filter for speech enhancement
ICHIT'06 Proceedings of the 1st international conference on Advances in hybrid information technology
Audio analysis for multimedia retrieval from a ubiquitous home
MMM'08 Proceedings of the 14th international conference on Advances in multimedia modeling
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
A review on speaker diarization systems and approaches
Speech Communication
State of the art of smart homes
Engineering Applications of Artificial Intelligence
A stereophonic acoustic signal extraction scheme for noisy and reverberant environments
Computer Speech and Language
Compressive speech enhancement
Speech Communication
Performance Study of the MVDR Beamformer as a Function of the Source Incidence Angle
IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP)
Hi-index | 35.69 |
This paper proposes a new robust adaptive beamformer applicable to microphone arrays. The proposed beamformer is a generalized sidelobe canceller (GSC) with a new adaptive blocking matrix using coefficient-constrained adaptive filters (CCAFs) and a multiple-input canceller with norm-constrained adaptive filters (NCAFs). The CCAFs minimize leakage of the target-signal into the interference path of the GSC. Each coefficient of the CCAFs is constrained to avoid mistracking. The input signal to all the CCAFs is the output of a fixed beamformer. In the multiple-input canceller, the NCAFs prevent undesirable target-signal cancellation when the target-signal minimization at the blocking matrix is incomplete. The proposed beamformer is shown to be robust to target-direction errors as large as 200 with almost no degradation in interference-reduction performance, and it can be implemented with several microphones. The maximum allowable target-direction error can be specified by the user. Simulated anechoic experiments demonstrate that the proposed beamformer cancels interference by over 30 dB. Simulation with real acoustic data captured in a room with 0.3-s reverberation time shows that the noise is suppressed by 19 dB. In subjective evaluation, the proposed beamformer obtains 3.8 on a five-point mean opinion score scale, which is 1.0 point higher than the conventional robust beamformer