Adaptive beamformer

About: Adaptive beamformer is a research topic. Over the lifetime, 4934 publications have been published within this topic receiving 93100 citations.

Robust near-field adaptive beamforming with distance discrimination

Abstract: This paper proposes a robust near-field adaptive beamformer for microphone array applications in small rooms. Robustness against location errors is crucial for near-field adaptive beamforming due to the difficulty in estimating near-field signal locations especially the radial distances. A near-field regionally constrained adaptive beamformer is proposed to design a set of linear constraints by filtering on a low rank subspace of the near-field signal over a spatial region and frequency band such that the beamformer response over the designed spatial-temporal region can be accurately controlled by a small number of linear constraint vectors. The proposed constraint design method is a systematic approach which guarantees real arithmetic implementation and direct time domain algorithms for broadband beamforming. It improves the robustness against large errors in distance and directions of arrival, and achieves good distance discrimination simultaneously. We show with a nine-element uniform linear array that the proposed near-field adaptive beamformer is robust against distance errors as large as /spl plusmn/32% of the presumed radial distance and angle errors up to /spl plusmn/20/spl deg/. It can suppress a far field interfering signal with the same angle of incidence as a near-field target by more than 20 dB with no loss of the array gain at the near-field target. The significant distance discrimination of the proposed near-field beamformer also helps to improve the dereverberation gain and reduce the desired signal cancellation in reverberant environments.

$M$ -DCSK-Based Chaotic Communications in MIMO Multipath Channels With No Channel State Information

Abstract: We consider chaotic digital communications in multiple-input-multiple-output (MIMO) wireless multipath fading channels. In particular, we focus on systems that employ M -ary differential chaos shift keying (M-DCSK). We consider two transceiver schemes, both of which require no channel state information at either the transmitter or the receiver. The first one employs a distinct chaotic sequence at each transmit antenna to spread the same data symbol and transmits omnidirectionally. At each receive antenna, the corresponding differential detection statistic is formed, and these statistics are then combined with equal gain for symbol detection. The second scheme employs a single chaotic spreading sequence and makes use of adaptive transmit and receive beamforming. The beamformers are updated by using a simple stochastic gradient method that is based on the received signal power and a finite-rate feedback strategy. Simulation results show that both schemes can effectively exploit the spatial diversity of the underlying MIMO system, and the adaptive beamforming scheme significantly outperforms the omnidirectional transmission.

Adaptive Wideband Beamforming With Frequency Invariance Constraints

Abstract: A response variation (RV) element is introduced to control the consistency of an adaptive wideband beamformer's response over the frequency range of interest. By incorporating the RV element into the linearly constrained minimum variance (LCMV) beamformer, we develop a novel linearly constrained beamformer with an improved output signal-to-interference-plus-noise ratio (SINR), compared to both the traditional formulation and the eigenvector based formulation, due to an increased number of degrees of freedom for interference suppression. In addition, two novel wideband beamformers robust against look direction estimation errors are also proposed as a further application of the RV element. One is designed by imposing a constraint on the RV element and simultaneously limiting the magnitude response of the beamformer within a pre-defined angle range at a reference frequency; the other one is obtained by combining the RV element and the worst-case performance optimization method. Both of them are reformulated in a convex form as the second-order cone (SOC) programming problem and solved efficiently using interior point method. Compared with the original robust methods, a more efficient and effective control over the beamformer's response at the look direction region is achieved with an improved overall performance.

Widely-linear recursive least-squares algorithm for adaptive beamforming

Abstract: Adaptive beamforming algorithms typically rely on a complex linear model between the sensor measurements and the desired signal output that does not enable the best performance from the data in some situations. In this paper, we present an extension of the well-known recursive least-squares algorithm for adaptive filters to widely-linear complex-valued signal and system modeling. The widely-linear RLS algorithm exploits a structured covariance matrix update that maintains information about the non-circularity of the input data to solve the widely-linear least-squares task at each snapshot. In addition, the WL-RLS algorithm can easily be switched between conventional and widely-linear complex modeling as needed. Application of the method to adaptive beamforming of mixed BPSK and QPSK signal transmissions shows that the system can extract all of the transmitted signal outputs in certain overloaded scenarios, and it performs up to 3dB better than the conventional RLS beamformer when the array is not overloaded.

Multimicrophone adaptive beamforming for interference reduction in hearing aids.

Abstract: To reduce interference in monaural hearing aids from sound sources that are spatially separated from a target source, we are investigating methods for combining information from multiple microphones. In this paper, we describe an adaptive beamforming method that functions to preserve target signals arriving from straight-ahead of a microphone array while minimizing output power from off-axis interference sources. In a preliminary evaluation of a two-microphone system, sentence intelligibility tests were administered to normal-hearing subjects using processed and unprocessed materials from simulated environments in which the target was on-axis, the interference (speech babble) was 45 degrees off-axis, and the reverberation mimicked that of a living room, a conference room, and anechoic space. Compared to listening through a single microphone, the two-microphone beamformer reduced the target-to-interference ratio required to achieve 50 percent keyword intelligibility by 30, 14, and 0 dB in the anechoic, living-room, and conference-room conditions, respectively. The corresponding improvements over binaural listening (one microphone to each ear) were 24, 9, and 0 dB. Further tests in the living-room environment using the same beam-forming system but with filter impulse responses shortened by a factor of four (which would decrease the adaptation time by a factor of four) decreased the improvement by 5 dB. These results are sufficiently encouraging to warrant further tests involving more realistic reverberant conditions, multiple sources of interference, and time-varying acoustic environments.