冲击噪声背景下基于最小均方归一化误差的波束形成算法

 本文提出一种适用于任意未知统计特性的代数拖尾冲击噪声环境下的MMSNE波束形成算法。算法利用输出信号和参考信号之间的“归一化误差”最小化来求解最优权向量。“归一化误差”定义为接收信号的瞬时自适应无穷范数归一化的形式。与基于最小分数低阶误差波束形成算法相比,MMSNE波束形成算法计算更为简单;不需要噪声特征指数的先验信息或估计;适用于更广的冲击噪声环境;具有更小的估计误差;具有更强的干扰抑制能力。     

一种冲击噪声环境下的最优波束形成算法

针对冲击噪声（包括所有对称α稳定分布噪声）环境下一般未知噪声统计特性,提出了一种最小几何功率无失真响应波束形成算法。该算法利用求取有限对数矩时无需噪声统计特性的特点,用对数矩代替二阶矩在约束期望信号增益为1的条件下使快拍信号的几何功率最小,最终使输出信号能达到更高的几何信干噪比。并通过一种新的迭代的方法得到最优权向量。仿真实验证明了该算法的有效性。     

冲击噪声背景下基于归一化的线性约束特征干扰相消器

针对冲击噪声背景下,常规波束形成算法性能下降的问题,本文提出一种适用于任意未知统计特性的代数拖尾冲击噪声环境下的基于归一化的线性约束特征干扰相消器（N-LCEC）算法。该算法在附加线性约束的条件下,以噪声功率最小化为目标函数;通过对输入信号进行无穷范数归一化,使变换信号的二阶统计量在代数拖尾的冲击噪声环境下存在且有界,然后将自适应权矢量约束于噪声子空间的方法,提高了波束形成器在冲击噪声背景下的性能。N-LCEC算法无需噪声特征指数的先验信息,适用冲击噪声环境更广;N-LCEC算法具有运算简单,干扰抑制能力强,同时保持静态方向图的副瓣特征等优点。仿真结果验证了该算法的有效性和优越性。      

一种适用于任意峰度非高斯信号的多目标盲波束形成方法

该文提出了一种新的适用于任意峰度非高斯信号的多目标盲波束形成算法。该方法在最大峰度算法(KMA)的基础上,引入信号的互相关约束定义了一种新的代价函数。通过对该代价函数的最大化,来盲估计波束形成器的权向量,从而完成对多目标的分离和定向。由于波束形成过程不需要正交化处理,避免了波束形成器锁定噪声的可能性。同时,采用一种复数编码的遗传算法来求解权向量,不但可避免陷入局部极值的可能性,而且还能有效提高运行速度。仿真实验验证了该算法的有效性和正确性。     

CDMA系统智能天线盲自适应波束形成

给出了基于最大化信号对干扰和噪声的功率比(MSINR)准则的盲自适应波束形成在线递推算法,旨在提高DS-CDMA系统中智能天线的性能.将MSINR准则转化为一种新的等价的无约束准则函数,通过最大化此准则函数来在线估计最优波束形成向量.算法直接利用了DS-CDMA系统解扩前和解扩后的观测信号,无须训练序列.通过在线迭代估计,降低了计算波束权值的复杂度.不同通信环境中的仿真结果表明,该算法具有较高的估计精度和很好的抗多址干扰能力.     

基于二次型约束的鲁棒自适应波束形成算法

针对期望信号方向向量存在偏差会导致自适应波束形成算法的性能急剧下降这一问题,该文提出了一种基于二次型约束的鲁棒自适应波束形成算法。通过对期望信号波达方向附近范围内的方向向量的误差模值进行约束,来提高算法的鲁棒性,并在此约束条件下对权重向量进行优化求解,且优化解中的参数能够准确求出。该算法可有效地控制波束主瓣区域内信号的畸变,并能够抑制方向向量偏差所带来的影响,提高了系统的鲁棒性,同时使干扰和噪声的功率输出最小,保证了对干扰信号的抑制能力,改善了阵列输出的信干噪比,使其更接近最优值。仿真结果验证了所提算法的有效性与优越性。     

基于半正定秩松弛方法的稳健波束形成

现有的向量加权稳健波束形成方法只有在指向误差较小的情况下才能有效估计目标的信号功率;矩阵加权波束形成方法在指向误差较大时,虽然可以估计目标的信号功率,但是它的系统实现复杂度与向量加权稳健波束形成方法相比较大。针对以上问题,该文提出基于半正定秩松弛(SDR)方法的稳健波束形成,该方法优化模型中的目标函数与Capon算法的目标函数相同,优化变量为加权向量的协方差矩阵,并约束方向图的主瓣幅度波动范围、旁瓣电平,协方差矩阵的秩为1。应用SDR方法求得加权向量的协方差矩阵,将该矩阵中的每一行(列)转化为加权向量,然后选择使得方向图主瓣与0 dB之间失真最大值最小的一个加权向量。该方法的系统实现复杂度与传统向量加权方法一致,对信号功率的估计性能与矩阵加权方法相当。仿真实验验证了该文方法可以得到理想的方向图形状,并且可以在大指向误差条件下有效估计信号功率。     

一种基于恒模算法的多用户盲波束形成新方法

提出了一种适用于多用户的盲波束形成新方法.该方法基于最小二乘恒模算法,当最小二乘恒模算法收敛于某一信号后,我们导出了算法的权向量与其它用户波达方向的关系,进而在不同用户的波达方向上形成初始增益,从而实现对所有用户的波束形成.文中给出了算法的并行和串行实现方案.     

智能天线中新的波束形成方法

提出了智能天线中三种利用估计得到的信号导向向量直接形成波束的新方法.根据CDMA信号模型和多径信道模型,用码滤波法估计得到用户最强信号(主径)的导向向量,利用这些导向向量形成恢复期望信号,抑制干扰的制约条件.根据制约条件可直接得到最小二乘(LS)波束形成方法;按照制约条件使输出能量最小得到广义最小方差无失真响应(GMVDR)波束形成方法;在GMVDR方法的基础上引入虚拟信号来进一步抑制旁瓣得到扩展最小方差无失真响应(EMVDR)波束形成方法.模拟表明这三种方法都能很好的恢复期望信号、抑制干扰和噪声,但通过比较看出EMVDR方法形成的波束最好.     

一种改进均方误差性能的新型鲁棒性波束形成算法

结合信干噪比最大化和均方误差最小化两个优化目标,提出一种新型的鲁棒性波束形成算法.该方法考虑信号估计误差,在传统的最小方差的代价函数中引入信号协方差矩阵的估计误差,并在波达角估计误差的约束下,将鲁棒性波束形成器转换成基于支持向量机形式的波束形成器,通过一种高效的新型支持向量机训练算法计算阵列权值;然后以均方误差最小化为目标来修正阵列权值.仿真结果表明:该方法降低了波束形成器对信号估计误差的敏感度,提高了其抑制非平稳干扰的能力,且具有更好的均方误差性能.     

Spatial blocking filter derivative constraints for the generalizedsidelobe canceller and MUSIC

The paper presents a new approach to spatial derivative constraints for the generalized sidelobe canceller (GSC). Spatial derivative constraints have been applied to linearly constrained minimum variance beamformers to reduce the sensitivity to steering error. Earlier approaches to this problem constrained derivatives of the beamformer power and phase response, leading to beamformer performance that depended on the coordinate reference location of the array. Current approaches constrain only the beamformer power response, eliminating the problem with phase dependence. However, nonlinear minimization is required in order to solve for the linear constraint equations. An alternative approach for the GSC, which is presented in the paper, is to use derivative constraints to flatten the null of the spatial blocking filter power response. Thus, for a small steering error, the desired signal is still blocked from the noise cancelling filter, and the GSC output is unaffected by the steering error. These derivative constraints can be used with wideband array calibration, leading to effective performance in the presence of array errors. This same approach to derivative constraints can be used in other applications involving spatial blocking filters, such as the constrained MUSIC direction finding algorithm to give robustness against direction error of the known signal subspace     

复广义高斯分布多通道最大似然联合去噪去混响波束形成器

提出了一种基于复超高斯分布的多通道联合去噪去混响波束形成器。本文采用复超高斯模型对语音信号建模,在最大似然准则下首次推导出联合去噪去混响波束形成器的解析表达式,并证明了该式是现有多种联合去噪去混响波束形成器的一般化形式。同时通过理论推导证明本文所提算法性能优于传统多通道预测误差算法级联最小功率无失真波束形成器。仿真实验与实际实验结果均表明,本文提出的算法在多个客观指标上明显优于现有联合去噪去混响算法。      

一种新的相干信号波束形成方法

存在相干信号情况下的常规波束形成器,由于相干信号的相对信号强度估计误差、天线阵导向矢量扰动误差和采样协方差矩阵估计误差的存在,会产生波束畸变及信号相消,输出信干噪比会明显下降。该文提出一种基于量子信号处理中内积成形算法的波束形成器,并分析了波束形成器的性能,通过仿真比较验证了新的波束形成器的有效性。分析和实验表明,新波束形成器作为一种鲁棒性波束形成器,在上述应用环境中,通过适当调整影响因子,可以得到比传统波束形成器更稳定的性能输出。     

An adaptive beamforming technique based on cyclostationary signalproperties

A new approach to adaptive beamforming is presented. The method is based on the property of cyclostationary signals to generate spectral lines when they pass through certain nonlinear transformations. The beamformer coefficients are selected according to a new optimization objective, which consists on minimizing the mean square error between the array output after the nonlinearity and a complex exponential. This approach optimally extracts any signal that generates a spectral line at the same frequency as the reference complex exponential. A gradient-based algorithm is derived to compute the optimum weights. Since the proposed cost function is a nonconvex function of the array coefficients, minima are analyzed for the three most common types of perturbations found in communications: Gaussian noise, multiple interferences, and multipath propagation. It is demonstrated via analysis and simulations that minima correspond to points where output noise power is minimized, interferences are canceled, and intersymbol interference is removed, i.e., the beamformer eliminates the distortion introduced by the radiocommunication channel     

旁瓣电平软约束下干扰功率搜索的波束形成

实际中,期望信号或干扰的先验方向信息往往不能精确已知(即存在阵列流形误差),当信噪比超过一定门限时,线性约束自适应波束形成器对阵列流形误差有高敏感性从而导致输出信干噪比下降。为解决上述问题,该文提出一种旁瓣电平软约束条件下搜索人工干扰功率值的波束形成方法。在粗估干扰来波方向的基础上,约束旁瓣最大峰值起伏在一定范围内,采用子空间投影和加宽干扰零陷区的技术提高波束形成器的鲁棒性,通过粗、精两种模式的搜索获得干扰零陷区的注入功率值估计,具有良好的干扰抑制效果和对阵列流形误差不敏感的特性。计算机仿真验证了方法的有效性和鲁棒性。     

Novel Transmit Beamforming Schemes for Time-Selective Fading Multiantenna Systems

Transmit beamforming has been widely adopted for wireless systems with multiple transmit antennas. For a block fading channel, the Grassmannian beamformer has been shown to provide very good performance for finite rate feedback. However, the original Grassmannian beamformer does not take the time domain correlation of the channel fading into consideration. In this paper, based on a first-order autoregressive (AR1) dynamic fading model, we develop two new classes of beamforming algorithms that exploit the interframe correlations in the channel fading. We first introduce an algorithm based on a standard predictive vector quantization (PVQ) approach, and the resulting PVQ beamformer accomplishes superior power delivery at the receiver. However, the error performance of the PVQ beamformer is not satisfactory at high signal-to-noise ratios, and it also has a high implementation complexity. To resolve these issues, we then develop a novel successive beamforming (SBF) algorithm. The new SBF scheme uses the knowledge of the previous fading blocks to aid the beamforming codebook design of the current fading block. The beamforming codebook is constructed based on the successive partition of the surface of a spherical cap. The new SBF scheme accomplishes nearly the same performance as that of the PVQ beamformer, and it has a much simpler implementation. Through numerical simulations, we demonstrate that the proposed beamformers outperform the other previously proposed beamformers at various fading scenarios     

A robust adaptive beamformer for microphone arrays with a blockingmatrix using constrained adaptive filters

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     

A New Post-filter Algorithm Combined with Two-step Adaptive Beamformer

The optimal microphone array, in the sense of minimum mean square errors (MMSE), includes two processing blocks: the minimum variance distortionless response (MVDR) beamformer and the single-channel Wiener filter, which acts as post-filter. In this paper, we propose a new post-filter algorithm based on assumptions that both the noise power attenuation factor (NPAF) and signal power attenuation factor (SPAF) are time invariant in the reverberant room. The algorithm recursively estimates both factors from available measurements and uses them in estimation of the post-filter parameters. Additionally, to overcome the problem of the poor performance of the MVDR beamformer in reverberant conditions, we propose the usage of the two-step (TS) MVDR algorithm. This algorithm improves the robustness of the beamformer and its ability to suppress the interferences using an estimate of the desired speaker transfer function. Although TS MVDR beamformer and proposed post-filter can work separately, or combined with other algorithms, the best performance is obtained when they work together. The performance of the proposed combination of new post-filter algorithm and TS MVDR beamformer is tested in a simulated reverberant room and compared with similar algorithms, which gave rather good results.