Subband Beamforming Research Articles

Broadband beamforming of multiplet line arrays using subband optimal beamformers eliminating port/starboard ambiguity

Development of beamforming methods for towed sonar multiplet line arrays is stimulated by their increased use in undersea surveillance. Such methods employ various forms of narrowband optimal beamforming to resolve port/starboard ambiguity present when the array is conventionally beamformed. In this paper, we depart from the purely narrowband approach and discuss methods for port/starboard ambiguity rejection (PSAR) using a broadband formulation. The proposed methodology relies on subband beamforming for which we use non-adaptive variants of beamspace Minimum Variance Distortionless Response (MVDR) and the Linear Constraint Minimum Variance (LCMV) beamformers. We provide detailed description and comparison of both subband methods and establish a connection between them. The paper provides assessment of PSAR properties of the developed broadband beamformers using outputs of signal processing of both simulated and experimental sonar data. Experimental data for this work were obtained through the participation in the Littoral Continuous Active Sonar (LCAS) trials.

Superdirective non-linear beamforming with deep neural network

Beamforming has been one of important issues in acoustic signal processing, since it can achieve signal enhancement and sound source localization. In general, traditional beamformers are designed by an analytical approach or an adaptive approach. It is, however, difficult to properly optimize the beamformers under the complicated acoustical scene. An alternative non-linear beamforming can be substituted for the linear beamforming. In this study, a flexible framework for optimizing the beamformer is introduced based on a deep neural network. Capturing acoustic signals using a microphone array is regarded as spatial sampling, so that annoying grating lobes appear in beam-pattern when the relationship between the wavelength and the microphone spacing does not satisfy the sampling theorem. The proposed method achieves sub-band beamforming using the non-uniform microphone array with eight nesting microphones, which are carefully designed not to cause spatial aliasing. Feasibility of the proposed method has been confirmed by computer simulation. The proposed non-linear beamformer could successfully achieve superdirectivity compared with conventional beamformers.

Reception of wideband signals from geostationary collocated satellites with antenna arrays

Reception of satellites in Ka-band is complicated because of narrow beams of ground station antennas. Narrower beams bring about higher pointing losses and the impossibility to capture multiple geostationary collocated satellites with the same aperture. One way to solve both problems is by using arrays of small dishes. Besides, if rain diversity is used, several ground stations must be deployed in sites separated by several tens of kilometres. The signals received by these stations could be combined in a central site to gain additional margin in the link budget. This study shows the processing scheme necessary to combine the signals both at the remote sites and the central site. It is based on a subband beamformer with subband delay compensation in order to be able to cope with the delay difference between the satellites. In order to estimate the phase and delay offsets between far away ground stations, an algorithm that works with the subband signals is proposed. Theoretical and simulation results validate the proposed approach.

Speech enhancement by non-linear beamforming tolerant to misalignment of target source direction

Speech interface are put to practical use in our daily life. However, current speech interfaces are fragile against acoustical interferences. It is dispensable to install noise reduction function in order to capture the target signal accurately. Spatial filtering using multiple observed signals, which are captured by spatially distributed microphones, is one of the possible noise reduction methods. A spatial filter aims to enhance the target signal and reduce the acoustical interferences relying on the arrival direction of the sound source. In this paper, non-linear sensor arrangement is proposed to achieve efficient beamforming. In general, a spatial filter with a sharp beam-pattern is weak against the misalignment of the arrival direction of the target signal. To improve the robustness of the spatial filter, the sub-band non-linear beamformer is optimized in each frequency range, and is weighted and merged to achieve the smooth beam-pattern in the spatial-spectro domain. The performance of the proposed scheme is examined objectively and subjectively under the condition with the misalignment of the target signal direction. Signal-to noise ratio in energy is calculated as the objective evaluation, and the formal listening tests are carried out as the subjective evaluation. It is confirmed that the proposed non-linear beamformer has an advantage in both noise reduction performance and the robustness against the misalignment of the target signal direction over the conventional linear and non-linear delay-and-sum beamformers.

Robustness analysis of nearfield subband beamformers in the presence of microphone gain and phase errors

Subband beamforming has found many applications in microphone array processing field, due to its advantages over the fullband counterpart. In this paper, the performance of nearfield subband beamformers for arbitrary arrays in the presence of microphone gain and phase errors is studied from the perspective of statistical analysis. Through the bias and variance analysis of array response, some insightful properties on the robustness of nearfield subband beamformers have been derived. It is shown that the robustness of nearfield subband beamformers is dependent on the source-to-array distance, i.e., the robustness will deteriorate when the source is close to microphone arrays. Moreover, the variation in sound speed, i.e., the temperature in homogeneous environments, has little effect on the robustness performance of subband beamformers. The theoretical results are further verified by several numerical examples on nearfield subband beamformers.

Subband design of fixed wideband beamformers based on the least squares approach

Subband method is an effective way to reduce the computational complexity of a wideband system and in this paper we study the subband design problem for fixed wideband beamformers, with an emphasis on the design of frequency invariant beamformers (FIBs). We first express the equivalent fullband beam response as a function of the subband beamformer coefficients and then formulate the design problem based on the least squares approach. One direct least squares formulation is first proposed for the design of a general wideband beamformer, and then extended to the FIB design case, followed by three further variations.

Fast Convergence Blind Source Separation Using Frequency Subband Interpolation by Null Beamforming

We propose a new algorithm for the blind source separation (BSS) approach in which independent component analysis (ICA) and frequency subband beamforming interpolation are combined. The slow convergence of the optimization of the separation filters is a problem in ICA. Our approach to resolving this problem is based on the relationship between ICA and null beamforming (NBF). The proposed method consists of the following three parts: (I) a frequency subband selector part for learning ICA, (II) a frequency domain ICA part with direction-of-arrivals (DOA) estimation of sound sources, and (III) an interpolation part in which null beamforming constructed with the estimated DOA is used. The results of the signal separation experiments under a reverberant condition reveal that the convergence speed is superior to that of the conventional ICA-based BSS methods.

Blind Subband Beamforming With Time-Delay Constraints for Moving Source Speech Enhancement

A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.

Adaptive Wideband Beamforming with Combined Spatial/Temporal Subband Decomposition

An adaptive wideband beamforming structure with a lower computational com- plexity and higher convergence speed is proposed, which is achieved by combined spatial/temporal subband decomposition. First, the received array signals are processed by a transformation ma- trix, and then split into subbands by a series of analysis fllter banks. At each subband, an independent beamformer can be operated, where some of the lower subbands can be discarded without afiecting its performance due to the highpass flltering efiect of the transformation matrix. 1. INTRODUCTION Beamforming has found many applications in various areas ranging from sonar and radar to wireless communications (1). It is a signal processing technique to form beams in order to receive signals illuminating a sensor array from speciflc directions, whilst attenuating signals from other directions. In a statistically optimum beamformer (2), since the statistics of the array data are often not known or may change over time, adaptive algorithms may be used to determine the beamformer's coe-cients. A beamformer structure for wideband array signals is shown in Fig. 1, where each of the M received signals is processed by an FIR fllter with a length of J and the outputs of these FIR fllters are then sumed up to form the flnal output. To perform adaptive wideband beamforming with high interference rejection and angular resolu- tion, arrays with a large number of sensors and fllter coe-cients have to be employed. Reducing the resultant high computational complexity and increasing its convergence speed has motivated many difierent solutions, including partially adaptive beamforming (3), transform-domain or frequency- domain beamforming (4,5), and subband beamforming (6,7). Recently, a generalised sidelobe can- celler (GSC) with combined subband decomposition in both the temporal and spatial domains was proposed in (8), where the columns of the blocking matrix are judiciously designed to generate a highpass flltering efiect to the array signals. The bandlimited spectra of the resultant outputs are then exploited by subband decomposition and appropriately discarding the low-pass subbands, where there is no signal existing. In this paper, we generalize the idea in (8) and propose a subband-selective transformation matrix, which is applied to the received array signals as a preprocessing step. Each of the outputs

Subcanceller: adaptive generalised sidelobe canceller with optimum subband decomposition

This paper presents a subband beamformer with the generalised sidelobe canceller (GSC) as the underlying structure. This new beamformer, referred to as the subcanceller, determines the statistically optimum filter bank succeeding the blocking matrix of the GSC based on the minimum mean-square error criterion. As a consequence, the signals passing through the lower branch of the GSC are decomposed into more appropriate subband components by such filter banks according to the input data statistics to enhance the interference suppression capability. The determination of the filter bank coefficients, however, calls for computationally demanding nonlinear optimisation. To alleviate the computational overhead, an iterative procedure is also proposed, which solves the Rayleigh quotient in each iteration. Furthermore, a detailed performance analysis of the proposed beamformer is conducted, which includes the development expression of the output signal to interference-plus-noise ratio and its implications, and the analysis of the convergence characteristic. Furnished simulations show that the new scheme yields superior interference suppression performance with a faster convergence rate compared with previous studies.

Optimal FIR subband beamforming for speech enhancement in multipath environments

The letter provides an analysis of optimal finite-impulse response subband beamforming for speech enhancement in multipath environments. A modification of the direct-path standard Wiener formulation is shown to give nearly optimal performance when it comes to signal-to-noise-plus-interference ratio, by including coherent multipath propagation in the criterion.

Performance limits in subband beamforming

This paper analyzes subband beamforming schemes mainly aimed at speech enhancement and acoustic echo suppression applications such as hands-free telephony for both mobile and office environments, Internet telephony and video conferencing. Analytical descriptions of both causal finite-length and noncausal infinite-length subband microphone array structures are given. More specifically, this paper compares finite Wiener filter performance with the noncausal Wiener solution, giving a comprehensive theoretical suppression limit. It is shown that even short filters will yield a good approximation of the infinite solution, provided that the element spacing and temporal sampling is matched to the frequency band of interest. Typically, 10-20 FIR taps are sufficient in each subband.

Robust microphone arrays using subband adaptive filters

A new adaptive beamformer which combines the idea of subband processing and the generalised sidelobe canceller structure is presented. The proposed subband beamformer has a blocking matrix that uses coefficient-constrained subband adaptive filters to limit target cancellation within an allowable range of direction of arrival. Simulations comparing the fullband and subband adaptive beamformers show that the subband beamformer has better performance than the fullband beamformer when the target and/or interfering signals are coloured. In reverberant environments, the proposed subband beamformer also performs better than its fullband counterpart.

Multirate sub-band adaptive beamforming using different kernel QMF banks

An investigation into the performance of a multirate sub-band adaptive beamforming array is carried out. Quadrature mirror filter (QMF) banks are used here to partition the frequency band, rather than the conventional Fast Fourier transform (FFT) algorithm. The MSE performances for the sub-band adaptive beamformer with (i) different filter order of the QMF banks but fixed number of sub-bands and (ii) fixed length of the QMF bank but different number of sub-bands are analysed. Both cases show reduction in MSE as either the QMF order is increased or the number of bands is increased. The selection of the QMF bank and the number of bands are left to the user, depending upon the computational complexity and implementation cost. Two examples are chosen from the above analysis and further investigations are carried out on them. The behaviour of the different structures, in terms of eigenvalue performance, radiation characteristics and mean square errors (MSE) are studied and analysed. The filter orders used are 16 (for eight sub-bands) and 32 (for four sub-bands) in length. The MSE are almost identical for both cases.

Efficient multirate adaptive beamforming technique

An efficient beamforming technique based on the recently proposed quadrature mirror filter based multirate sub-band adaptive beamformer (QMF-MSAB), is presented. The improved technique uses only one adaptive processor for all sub-bands, resulting in a considerable reduction in hardware requirements. In addition, the new structure maintains the overall superiority of the multirate approach over full-band beamforming.