Robust adaptive beamforming for constant modulus signal of interest

It is required in the diagonally loaded robust adaptive beamforming the automatic determination of the loading level which is practically a challenging problem.A constant modulus restoral method is herein presented to choose the diagonal loading level adaptively for the extraction of a desired signal with constant modulus(a common feature of the phase modulation signals).By introducing the temporal smoothing technique,the proposed constant modulus restoral diagonally loaded robust adaptive beamformer provides increased capability compared with some existing robust adaptive beamformers in rejecting interferences and noise while protecting the signal-of-interest.Simulation results are included to illustrate the performance of the proposed beamformer.

Fractional Processing Based Adaptive Beamforming Algorithm

Fractional order algorithms have shown promising results in various signal processing applications due to their ability to improve performance without significantly increasing complexity.The goal of this work is to inves-tigate the use of fractional order algorithm in the field of adaptive beam-forming,with a focus on improving performance while keeping complexity lower.The effectiveness of the algorithm will be studied and evaluated in this context.In this paper,a fractional order least mean square(FLMS)algorithm is proposed for adaptive beamforming in wireless applications for effective utilization of resources.This algorithm aims to improve upon existing beam-forming algorithms,which are inefficient in performance,by offering faster convergence,better accuracy,and comparable computational complexity.The FLMS algorithm uses fractional order gradient in addition to the standard ordered gradient in weight adaptation.The derivation of the algorithm is provided and supported by mathematical convergence analysis.Performance is evaluated through simulations using mean square error(MSE)minimization as a metric and compared with the standard LMS algorithm for various parameters.The results,obtained through Matlab simulations,show that the FLMS algorithm outperforms the standard LMS in terms of convergence speed,beampattern accuracy and scatter plots.FLMS outperforms LMS in terms of convergence speed by 34%.From this,it can be concluded that FLMS is a better candidate for adaptive beamforming and other signal processing applications.

Novel Adaptive Beamforming Algorithm Based on Wavelet Packet Transform

An analysis of the received signal of array antennas shows that the received signal has multi-resolution characteristics, and hence the wavelet packet theory can be used to detect the signal. By emplying wavelet packet theory to adaptive beamforming, a wavelet packet transform-based adaptive beamforming algorithm (WP-ABF) is proposed . This WP-ABF algorithm uses wavelet packet transform as the preprocessing, and the wavelet packet transformed signal uses least mean square algorithm to implement the ~adaptive beamforming. White noise can be wiped off under wavelet packet transform according to the different characteristics of signal and white under the wavelet packet transform. Theoretical analysis and simulations demonstrate that the proposed WP-ABF algorithm converges faster than the conventional adaptive beamforming algorithm and the wavelet transform-based beamforming algorithm. Simulation results also reveal that the convergence of the algorithm relates closely to the wavelet base and series; that is, the algorithm convergence gets better with the increasing of series, and for the same series of wavelet base the convergence gets better with the increasing of regularity.

Adaptive broadband beamformer for nonuniform linear array based on second order cone programming

Adaptive broadband beamforraing is a key issue in array applications. The adaptive broadband beamformer with tapped delay line （TDL） structure for nonuniform linear array （NLA） is designed according to the rule of minimizing the beamformer＇s output power while keeping the distortionless response （DR） in the direction of desired signal and keeping the constant beamwidth （CB） with the prescribed sidelobe level over the whole operating band. This kind of beamforming problem can be solved with the interior-point method after being converted to the form of standard second order cone programming （SOCP）. The computer simulations are presented which illustrate the effectiveness of our beamformer.

Power Estimation of Acoustic Sources by Sensor Array Processing

A constant problem is to localize a number of acoustic sources, to separate their individual signals and to estimate their strengths in a propagation medium. An acoustic receiving array with signal processing algorithms is then used. The most widely used algorithm is the conventional beamforming algorithm but it has a very low resolution and high sidelobes that may cause a signal leakage problem. Several new signal processors for arrays of sensors are derived to evaluate the strengths of acoustic signals arriving at an array of sensors. In particular, we present the covariance vector estimator and the pseudoinverse of the array manifold matrix estimator. The covariance vector estimator uses only the correlations between sensors and the pseudoinverse of the array manifold matrix estimator operates with the minimum eigenvalues of the covariance matrix. Numerical and experimental results are presented.

DOA Estimation Algorithm Based on Adaptive Filtering in Spatial Domain

In this paper, a novel DOA estimation methodology based upon the technology of adaptive nulling antenna is proposed. Initially, the nulling antenna obtains the weight vector by LMS algorithm and power inversion criterion.Afterwards, reciprocal of the antenna pattern is defined as the spatial spectrum and the extracted peak values are corresponded to the estimated DOA. Through observation of the spectrum and data analysis of variable steps and SNRs, the simulation results demonstrate that the proposed method can estimate DOA above board. Furthermore, the estimation error of the proposed technique is directly proportional to step size and is inversely proportional to SNR. Unlike the existing MUSIC algorithm, the proposed algorithm has less computational complexity as it eliminates the need of estimating the number of signals and the eigenvalue decomposition of covariance matrix. Also it outperforms MUSIC algorithm, the recently proposed MUSIC-Like algorithm and classical methods by achieving better resolution with narrow width of peaks.

Minimum geometric power distortionless response beamforming against heavy-tailed noise of unknown statistics

A minimum geometric power distortionless response beamforming approach against impulsive noise （including all α- stable noise） of unknown statistics is proposed. Due to that definite logarithmic moments require no priori knowledge of impulsive noise, this new beamformer substitutes the logarithmic moments for the second-order moments and iteratively minimizes the ＂ge- ometric power＂ of the beamformer.s output snapshots, subjected to a linear constraint. Therefore, the proposed beamformer can provide significantly higher output geometric signal-to-noise-andinterference ratio. Moreover, the optimum weight vector is obtained by using a new iteration process. The simulation results prove that the new method is effective.

Focused Widely Linear Beamforming

The problem addressed in this paper concerns the extension of widely linear beamforming to the wideband case,developing a wide-focused linear beamformer for the extraction of a wideband second-order(SO)noncircular signal-of-interest(SOI)contaminated by uncorrelated interferences and noise.In the proposed beamformer,the beamforming array observation is first focused to adopt a standard linear minimum variance distortionless response(MVDR)framework.The augmented SOI steering vector then is obtained by estimating the SOI noncircularity parameter with the newly proposed oblique projection with an augmented sparse representation scheme.The covariance matrix of the virtual interference,true interference and noise is further reconstructed using the newly presented complementary spatial spectrum technique.The wideband widely linear spatial filtering is finally realized via MVDR like beamforming.The performance of the proposed beamformer is verified by simulation.

Feasibility of a novel beamforming algorithm via retrieving spatial harmonics

This paper introduces an algorithm for beamforming systems by the aid of multidimensional harmonic retrieval(MHR).This algorithm resolves problems,removes limitations of sampling and provides a more robust beamformer.A new sample space is created that can be used for estimating weights of a new beamforming called spatial-harmonics retrieval beamformer(SHRB).Simulation results show that SHRB has a better performance,accuracy,and applicability and more powerful eigenvalues than conventional beamformers.A simple mathematical proof is provided.By changing the number of harmonics,as a degree of freedom that is missing in conventional beamformers,SHRB can achieve more optimal outputs without increasing the number of spatial or temporal samples.We will demonstrate that SHRB offers an improvement of 4 dB in signal to noise ratio(SNR) in bit error rate(BER) of 10~(-4) over conventional beamformers.In the case of direction of arrival(DOA) estimation,SHRB can estimate the DOA of the desired signal with an SNR of-25 dB,when conventional methods cannot have acceptable response.

Multi-channel differencing adaptive noise cancellation with multi-kernel method

Although a various of existing techniques are able to improve the performance of detection of the weak interesting sig- nal, how to adaptively and efficiently attenuate the intricate noises especially in the case of no available reference noise signal is still the bottleneck to be overcome. According to the characteristics of sonar arrays, a multi-channel differencing method is presented to provide the prerequisite reference noise. However, the ingre- dient of obtained reference noise is too complicated to be used to effectively reduce the interference noise only using the clas- sical linear cancellation methods. Hence, a novel adaptive noise cancellation method based on the multi-kernel normalized least- mean-square algorithm consisting of weighted linear and Gaussian kernel functions is proposed, which allows to simultaneously con- sider the cancellation of linear and nonlinear components in the reference noise. The simulation results demonstrate that the out- put signal-to-noise ratio （SNR） of the novel multi-kernel adaptive filtering method outperforms the conventional linear normalized least-mean-square method and the mono-kernel normalized least- mean-square method using the realistic noise data measured in the lake experiment.

Adaptive Radio Frequency Interference Mitigation for HF Surface Wave Radar

The paper analyses the characteristics of radio frequency interference （RFI） in HF surface wave radar （HFSWR） which adopts the linear frequency modulated interrupted continuous wave （FMICW）. RFI will influence all the range cells including all the positive frequency and negative frequency, and the negative frequency range cells contain only the interference information. Based on the above characteristics, we introduce and analyze a new adaptive interference mitigation beamforming algorithm using the negative frequency range cells samples to estimate the interference covariance matrix. Experimental results confirm that this general and robust algorithm can achieve effective RFI suppression using the data recorded by the HFSWR, located near Zhoushan in Zhejiang China.

Space discriminative function for microphone array robust speech recognition

Based on W-disjoint orthogonality of speech mixtures, a space d,scnmlnative tunetlon was proposer1 to enumerate and localize competing speakers in the surrounding environments. Then, a Wiener-like postfiherer was developed to adaptively suppress interferences. Experimental results with a hands-free speech recognizer under various SNR and competing speakers settings show that nearly 69 % error reduction can be obtained with a two-channel small aperture microphone array against the conventional single microphone baseline system. Comparisons were made against traditional delay-and-sum and Griffiths-Jim adaptive beamforming techniques to further assess the effectiveness of this method.

基于卷积神经网络的自适应波束形成

提出一种基于卷积神经网络的自适应波束形成方法,旨在通过自适应调整接收波束方向以提高通道中信号的信噪比。采用同时多波束接收信号的概念,对阵列天线接收信号以不同的相位加权并相加,即依据自适应多波束形成多通道接收。利用神经网络的非线性处理能力,在阵列接收不同来波方向信号时可以对其实现自适应波束形成。使用波束形成方式设计标签,通过比较输出信号的信噪比相对于输入信号信噪比的增益以评价该方法的有效性。仿真有效提高了信号的信噪比,接收阵列可以同时接收多个波束信号并按各输入信号的入射角度不同在不同的通道输出对应信号。

发射波束域MIMO-STAP雷达发射接收联合设计方法

研究了信号相关杂波背景下机载多输入多输出(multiple-input multiple-output,MIMO)雷达发射波束形成和接收滤波器的联合设计问题,建立了机载MIMO雷达发射波束域空时自适应处理(space-time adaptive processing,STAP)信号模型。为了提升杂波环境下目标的检测性能,通过最大化输出信干噪比构建了发射波束形成和接收滤波器的联合设计问题;然后,设计了一种新的基于优化最小化(majorization-minimization,MM)框架的迭代优化算法来解决联合设计问题。该算法通过合理寻找目标函数的下界可以有效提升算法的收敛速度并降低算法的运行时间。此外,与传统的相控阵雷达和MIMO雷达相比,优化后的发射波束形成和接收滤波器可以显著提升输出信干噪比,仿真实验验证了所提方法的有效性。

阻塞矩阵方法对消主瓣干扰

当存在主瓣干扰且有期望信号混入情况下,用常规自适应波束形成方法进行自适应干扰对消,不但会引起主瓣变形,而且期望信号也会被抑制,从而影响对消性能。提出一种新的阻塞矩阵方法,对接收数据进行预处理,消除它们对计算数据协方差阵的影响,再用其它方法确定自适应权值。理论分析和计算机仿真表明,用阻塞矩阵方法可以获得较大的性能提升。

有限次快拍下自适应波束形成

从协方差矩阵估计角度出发 ,解释了有限次快拍下造成自适应波束形成性能下降和自适应波束畸变的原因 .有限次快拍等效增加了干扰之间、干扰与噪声之间以及噪声之间的相关性 ,造成自适应波束形成性能下降 .同时 ,有限次快拍下噪声等效为色噪声 ,文中分析了色噪声环境下自适应波束畸变机理 。

迭代对角加载采样矩阵求逆鲁棒自适应波束形成

为有效克服导向矢量大失配误差对自适应波束形成器的影响,该文提出了一种迭代对角加载采样矩阵求逆鲁棒自适应波束形成算法。该算法对传统对角加载算法进行了迭代运算,基于Capon波束形成器的最优权矢量与假定导向矢量的基本关系,将每一步得到的权矢量,对应反解出一个比导向矢量假定值更为准确的导向矢量,并替代假定值,最终逼近真实的期望信号导向矢量。提出的方法在迭代过程中只需一步递推,无需对导向矢量建立不确定集,避免了在每步迭代中运用拉格朗日数值法或凸优化法,且明显提高了波束形成器的输出信干噪比。仿真结果验证了算法的正确性和有效性。

雷达数字波束形成器的研究与实现

介绍了一个自适应数字波束形成器的原理、组成和实现方法 ,给出了实验结果 ,验证了一些关键技术 ,表明数字波束形成 (DBF)技术可提高雷达天线性能。

一种虚拟波束形成自适应加权空间平滑算法

该文提出了一种用于自适应阵列的自适应加权空间平滑算法。通过构造虚拟自适应波束形成问题求取用于子阵协方差矩阵加权的加权向量,然后进行加权空间平滑自适应波束形成。理论分析与仿真结果表明,新算法能更有效地降低期望信号和干扰之间的相关性,使得用于空间平滑的子阵数减少,以降低阵列孔径损失。

基于波束域导向矢量估计的稳健自适应波束形成方法

针对导向矢量失配和样本协方差矩阵估计误差导致自适应波束形成性能下降的问题,该文提出一种基于波束域导向矢量估计的稳健方法。该方法首先利用期望信号角度失配区域的补集构造波束域转换矩阵,克服了当样本协方差矩阵中存在期望信号分量时现有算法出现有用信号对消的缺点。其次推导出波束域导向矢量估计方法,并将其转化为包含恒模约束的非凸二次约束二次规划问题,然后采用半正定松弛规划估计实际的波束域导向矢量。计算机仿真验证了该文所提方法的有效性。