Performance evaluation of digital beamforming strategies for satellite communications

Smart antennas are becoming one of the promising technologies to meet the rapidly increasing demands for more capacity of satellite communication systems. A main component in a smart antenna system is beamforming. Because of the limitations of analog beamforming, digital beamforming will be employed in future satellite communication systems. We evaluate the performance of various digital beamforming strategies proposed in the literature for satellite communications: 1) single fixed beam/single user, 2) single fixed beam/multiple users, 3) single adaptive beam/single user, and 4) single Chebyshev dynamic beam/multiple users. Multiple criteria including coverage, system capacity, signal-to-interference-plus-noise ratio (SINR), and computation complexity are used to evaluate these satellite communication beamforming strategies. In particular, a Ka-band satellite communication system is used to address the various issues of these beamforming strategies. For the adaptive beamforming approach, subarray structure is used to obtain the weights of a large 2D antenna array, and a globally convergent recurrent neural network (RNN) is proposed to realize the adaptive beamforming algorithm in parallel. The new subarray-based neural beamforming algorithm can reduce the computation complexity greatly, and is more effective than the conventional least mean square (LMS) beamforming approach. It is shown that the single adaptive beam/single user approach has the highest system capacity.     

小波域的自适应波束形成算法

在分析传统自适应波束形成的基础上,首次提出了一种小波域的自适应波束形成算法。与通常的自适应波束形成算法相比,该算法利用小波变换对小波空间进行了分解,信号经小波变换自相关性会下降,收敛速度提高,同时在此分解过程中,根据信号与白噪声在不同尺度上的小波变换模极大值表现完全不同的特性进行信号的消噪。理论分析和仿真结果表明了该算法收敛速度较快,且计算量增加较少,易于实时实现,而且具有较好性能。同时仿真实验表明该算法收敛速度与小波基和尺度的选择有关,尺度越大收敛速度越快;对于同一小波基系列,小波基的正则性越好收敛速度越快。     

利用压缩网格方法提高旋转声源定位计算效率

利用两种压缩网格方法:基于传统波束形成的CG2压缩网格方法和基于小样本数据的CG4压缩网格方法,对经典时域旋转声源波束形成(ROSI)算法进行优化加速。实验结果表明:两种压缩网格方法均不影响ROSI算法旋转声源定位效果,基于传统波束压缩形成的CG2网格方法能够提高ROSI算法的旋转声源定位计算效率1～2倍,基于小样本数据的CG4压缩网格方法能够提高ROSI算法的旋转声源定位计算效率13～18倍。除此之外,基于小样本数据的CG4压缩网格方法在相控麦克风阵列平面与旋转声源平面垂直条件下,仍能准确进行旋转声源定位。      

基于阵列天线的一种GPS多径抑制方法

为了抑制多径效应对GPS的严重影响,在分析了多径效应对伪码跟踪环路影响的基础上,提出了一种基于空间平滑的改进稳健波束形成算法来抑制GPS多径效应。以直线阵为基础,采用前后向空间平滑算法对阵列接收信号进行预处理,针对各种失配,采用改进的稳健波束形成算法进行处理。该方法既可以有效地抑制GPS多径效应,又可以有效地改善伪码跟踪环路的跟踪精度。仿真实验结果验证了所提方法的正确性和有效性,具有较高的工程实践价值。     

飞机起落架噪声源定位的压缩感知算法

目前针对飞机起落架噪声源定位的研究方法主要是将麦克风阵列与波束成形算法相结合。常规波束成形（CBF）算法在计算时存在主瓣宽度过宽、结果易受旁瓣影响的问题。高级波束成形算法在计算时效率较差,有时会有违背物理现象的假声源出现。提出了一种将正交匹配追踪（OMP）算法与奇异值分解（SVD）相结合的起落架噪声源定位的OMP-SVD压缩感知算法。在消声实验室内进行飞机起落架噪声源定位试验,将OMP-SVD算法、CBF算法和OMP算法在不同频率下获得的结果进行对比。试验结果表明：①与OMP算法相比,OMP-SVD算法在不同频率下均能准确定位出起落架主声源;②与CBF算法相比,OMP-SVD算法显著提高了分辨率。     

Digital Beamforming on Receive: Techniques and Optimization Strategies for High-Resolution Wide-Swath SAR Imaging

Synthetic Aperture Radar (SAR) is a well-proven imaging technique for remote sensing of the Earth. However, conventional SAR systems are not capable of fulfilling the increasing demands for improved spatial resolution and wider swath coverage. To overcome these inherent limitations, several innovative techniques have been suggested which employ multiple receive-apertures to gather additional information along the synthetic aperture. These digital beamforming (DBF) on receive techniques are reviewed with particular emphasis on the multi-aperture signal processing in azimuth and a multi-aperture reconstruction algorithm is presented that allows for the unambiguous recovery of the Doppler spectrum. The impact of Doppler aliasing is investigated and an analytic expression for the residual azimuth ambiguities is derived. Further, the influence of the processing on the signal-to-noise ratio (SNR) is analyzed, resulting in a pulse repetition frequency (PRF) dependent factor describing the SNR scaling of the multi-aperture beamforming network. The focus is then turned to a complete high-resolution wide-swath SAR system design example which demonstrates the intricate connection between multi-aperture azimuth processing and the system architecture. In this regard, alternative processing approaches are compared with the multi-aperture reconstruction algorithm. In a next step, optimization strategies are discussed as pattern tapering, prebeamshaping-on-receive, and modified processing algorithms. In this context, the analytic expressions for both the residual ambiguities and the SNR scaling factor are generalized to cascaded beamforming networks. The suggested techniques can moreover be extended in many ways. Examples discussed are a combination with ScanSAR burst mode operation and the transfer to multistatic sparse array configurations.     

基于矩阵填充的二维自适应波束形成算法

针对基于矩阵填充的二维自适应波束形成问题,提出一种基于奇异值门限(SVT)的特征分解线性约束最小方差(SVT-ELCMV)算法。首先建立二维自适应波束形成矩阵填充模型,其次验证接收信号矩阵满足零空间性质(NSP),并分析最小可恢复阵元数,最后以SVT算法将稀疏阵列信号恢复为完整信号,并通过修正的特征分解线性约束最小方差(LCMV)形成有效波束。算法解决了稀疏阵列平均副瓣大幅度上升的缺陷,且在平面阵列部分阵元无法正常工作时依然有效。计算机仿真表明:SVT-ELCMV算法可使稀疏阵列具有与完整阵列相同的二维波束形成能力,并可有效抑制干扰信号,验证了算法的有效性和优越性。     

MIMO雷达DOD和DOA联合估计算法: RTR-ESPRIT

 针对双基地多输入多输出(MIMO)雷达多目标波离角(DOD)和波达角(DOA)的联合估计问题,提出一种接收-发射-接收(RTR)-ESPRIT算法。该算法首先利用一维接收ESPRIT(R-ESPRIT)预估计目标DOA,随后分别利用一维发射ESPRIT(T-ESPRIT)和一维接收ESPRIT得到目标的高精度DOD和DOA估计,在每两次ESPRIT算法之间分别构造正交投影算子对接收信号进行接收波束形成和发射波束形成。与传统ESPRIT算法相比,该算法大大降低了数据协方差矩阵维数和计算复杂度,无需额外的配对算法,且理论证明了该方法还可以用于相干目标和单快拍情况下DOD和DOA的联合估计。仿真结果表明了该算法的正确性及良好的估计性能。     

Fast Algorithms for Time Domain Broadband Adaptive Array Processing

Recursive algorithms are presented for time domain, broadband, adaptive beamforming. The algorithms are rapidly converging and can be computationally efficient for a certain range of array processor parameters. The algorithms are presented for two forms of array processor. One form is a Frost-type structure in which explicit constraints are required for defining the array-look direction and also to control the sensitivity of the array processor to implementation errors. The other form is a partitioned array processor in which constraints are built into the processor and the adaptive weight control algorithm is therefore unconstrained. The two processors presented are both element-space processors but the algorithms can be applied also to beam-space processors.     

Improving ESPRIT via beamforming

Beamforming techniques are presented that improve the performance of the ESPRIT (estimation of signal parameters via rotational invariance techniques) algorithm for solving the narrowband multiple source location problem with uniform linear arrays. Using ESPRIT with these beamforming approaches maintains the computational advantage of the original ESPRIT while further exploiting the special structure of uniform linear arrays. Simulation results show that the use of ESPRIT with beamforming yields significantly improved performance compared with the original ESPRIT     

四元数最小均方误差算法及其在波束形成中的应用

 作为多维信号处理的一个重要工具,四元数代数已在各个领域有所应用。对四元数最小均方误差(QMMSE)算法进行了研究,首先推导了四元数实数形式的最小均方误差(QRMMSE)算法,进一步推导了四元数复数形式的最小均方误差(QCMMSE)算法,并且分析了两种算法的区别和计算量。最后将QMMSE算法应用到机载简化矢量传感器阵列的波束形成中,与复数长矢量最小均方误差(LVMMSE)算法相比较,QCMMSE算法的性能有所提高,计算量有所减少。计算机仿真结果验证了所提算法的有效性。     

Direction-of-arrival estimation using signal subspace modeling

A computationally viable algorithm for estimating the direction-of-arrival (DOA) of multiple wavefields that are incident on an array of sensors is developed. The geometry of the array is unrestricted and the incident wavefields may be generated by mixtures of incoherent and coherent emitting sources. In the approach taken, the sensor signals are modeled as a noise-contaminated linear combination of steering vectors which are functionally dependent on a set of DOA parameters. These parameters are to be chosen so that this sensor signal model is most compatible with empirically measured data (i.e., snapshot data). An iterative procedure is developed for selecting the most data compatible set of DOA parameters in both the snapshot domain and the array covariance matrix domain. Critical to the success of such iterative solution procedures is the generation of quality DOA parameters to initialize the algorithm. A sequential beamforming method for this initialization is presented. Numerical examples to illustrate the effectiveness of the proposed algorithmic approach are given     

Exact least squares adaptive beamforming using an orthogonalizationnetwork

The pros and cons of various classical and state-of-the-art methods in adaptive array processing are discussed, and the relevant concepts and historical developments are pointed out. A set of easy-to-understand equations for facilitating derivation of any least-squares-based algorithm is derived. Using this set of equations and incorporating all of the useful properties associated with various techniques, an efficient solution to the real-time adaptive beamforming problem is developed     

A flexible processor for a digital adaptive array radar

A digital beamforming processor for an adaptive array radar is described. The functionality and the architecture of the processor are strongly driven by a goal of achieving adaptive null depths in the 60-dB to 70-dB range, which necessitates substantial preprocessing of each channel. In particular, conversion to baseband quadrature channels is accomplished digitally using a single A/D converter per channel, and FIR (finite impulse response) equalizing filters are employed in each channel to match channel transfer functions. The processor is highly modular, and this not only distributes the total processing load, but also the I/O (input/output) bandwidth requirement. This is accomplished by distributing the adaptive beamforming algorithm systolically across a linear array of processing nodes. The processor is expandable to a different number of channels and sufficiently flexible to be applied to other problems of an array signal processing nature. Experimental data presented demonstrate that the processor is capable of supporting channel-to-channel cancellation of interfering signals to the level of -65 dB     

基于状态观测器的波束形成算法及其航空应用

随着数字信号处理技术的发展,基于声传感器阵列的经典波束形成算法广泛地应用于航空气动噪声检测领域中.但是,经典算法计算量大,并且它的算法结构决定了其无法用作实时算法.本文给出了一种基于状态观测器理论的新算法,介绍了观测器算法的理论背景、算法稳定性的证明、麦克风阵列的设计准则以及基于阵列进行的仿真结果.这种新算法在形式上类...     

An intelligent algorithm for coherent sound source localization based on a strong tracking filter

Background noise is inevitable when sensor arrays are used for aeroacoustic measurements in wind tunnels. The direct removal of background noise, however, would affect the measurement accuracy. In particular, the existing array signal processing algorithms are either invalid or inefficient for removing the noise that is coherent with the signal of interest. In this paper, an intelligent algorithm is developed to localize the coherent sound sources and background noise in real time by iteratively checking the collected data from the array. The proposed method can automatically adjust the suboptimal fading factor to extract useful information as much as possible in residual sequence. This algorithm is tested in simulations and then demonstrated in an experiment.Compared to the two existing methods, the results indicate that the new method has a good phase-shift tracking ability and rapid estimation-error convergence speed, and can achieve an acceptable performance even for low-cost acoustic sensors. Overall, the proposed method should assist array beamforming and hence benefit aeroacoustic measurement.     

Tracking of conventional beamforming with hydrophone array ofvarying geometry

In conventional beamformers of a large towed hydrophone array of fixed linear array geometry the sensors may be displaced from their nominal positions due to various kinds of force on the array. The displacements are usually time varying and can severely degrade the performance of the beamformer. This problem is approached by restoring the sensor output signals before beamforming by adaptively tracking and compensating for the displacements. This approach does not require modifications of the conventional shipboard beamformer and hence can be used cost-effectively to upgrade existing conventional systems. Two adaptive algorithms are used for two different cases: known sensor moving trajectories and unknown trajectories with relatively small displacements compared with the intersensor spacing. Tracking performance of the algorithms is analyzed. Simulation results show that for known sensor trajectories the algorithm significantly improves the performance of the conventional shipboard beamformer whereas only limited improvement can be achieved when the trajectories are unknown     

Array shape estimation tracking using active sonar reverberation

This paper concerns the problem of array shape estimation and tracking for towed active sonar arrays, using received reverberation returns from a single transmitted CW pulse. Uniform linear arrays (ULAs) deviate from their nominal geometry while being towed due to ship maneuvers as well as ocean currents. In such scenarios, conventional beamforming performed under the assumption of a ULA can sometimes lead to unacceptably high spatial sidelobes. The reverberation leaking through the sidelobes can potentially mask weak targets in Doppler, especially when the target Doppler is close to that of the mainlobe reverberation and the reverberation-to-target ratio (RTR) is very high. Although heading sensors located along the array can be used to provide shape estimates, they may not be sufficiently available or accurate to provide the required sidelobe levels. We propose an array shape calibration algorithm using multipath reverberation returns from each ping as a distributed source of opportunity. More specifically, a maximum likelihood (ML) array shape calibration algorithm is developed, which exploits a deterministic relationship between the reverberation spatial and Doppler frequencies causing it to be low rank in the space-time vector space formed across a single coherent processing interval (CPI). In this application, a sequence of overlapped CPI length snapshots of duration less than the CW pulse is used. The ML estimates obtained for each snapshot are tracked using a Kalman filter with a state equation corresponding to the water pulley model for array dynamics. Simulations performed using real heading sensor data in conjunction with simulated reverberation suggest that 8-10 dB improvement in sidelobe level may be possible using the proposed array shape tracking algorithm versus an algorithm that uses only the available heading information.     

Digital beamforming calibration for FMCW radar

Three sources of beamforming deterioration are identified for an FMCW radar. These all relate to time-of-arrival and cable delays. There are small range-dependent phase errors at the receiver outputs. The delays cause frequency shifts, which produce phase errors at the discrete Fourier transform (DFT) outputs after ranging. These frequency shifts cause amplitude errors when the signals are resolved into the same range bin. Two methods of compensating for phase and amplitude errors are proposed. The first corrects for phase errors prior to beamforming. Amplitude errors are ignored but the residual error is usually small. The second method aligns the receiver frequencies digitally before ranging. This eliminates the phase errors and the amplitude errors at the DFT range estimates