球谐域混叠消除最优波束形成方法EI北大核心CSCD

为解决现有的球谐域混叠消除波束形成方法设计目标单一、难以很好地兼容多变的应用场景的问题,提出了球谐域混叠消除最优波束形成方法,并建立了其二阶锥规划形式的多重约束凸优化数学模型。得益于该数学模型,提出了旁瓣约束的混叠消除最大指向性波束形成器、鲁棒的混叠消除最大指向性波束形成器、混叠消除最小方差无失真响应波束形成器、零点约束的混叠消除最大指向性波束形成器。提出方法继承了现有球谐域混叠消除波束形成方法的混叠消除功能,且能够在指向性指数、白噪声增益、旁瓣水平等多个冲突的性能指标之间寻求适宜折中,便于用户根据实际应用需求灵活地调整约束参数得到特定场景下综合性能最优的波束形成器。     

同心圆环阵列模态波束形成器设计

针对贝塞尔函数零点造成的模态波束形成器性能损失的问题,提出了基于均匀同心圆环阵列的模态波束形成器设计方法。从阵元域出发,建立了阵元域和圆谐波域之间的转换关系,推导了圆谐波域阵列信号处理表达式,其中重点推导了延时求和、最小方差无失真响应和多约束波束形成器的设计方法。理论证明了在平面各向同性噪声场中的基于同心环阵的圆谐波域最小方差无失真响应波束形成器等价于相位模式波束形成器。在该结论的基础上,综合考虑相位模式波束形成器在一定频率范围内的稳健性和指向性,提出了一种同心圆环阵列的内环半径优化方法。使用16元双环阵列对本文提出的波束形成算法进行仿真和实验,结果表明:同心圆环阵列能够较好地解决贝塞尔函数零点问题;其中多约束波束形成器能够在多个关联的性能指标之间取得合理的折衷,实现优于相位模式和延时求和波束形成器的声源定位性能。对内环半径优化方法进行仿真和分析结果表明,该方法得到了相较于传统方法更优的阵列结构。      

多项式结构宽带波束形成器的性能分析及其改进

首先从理论上分析了多项式结构宽带波束形成器的鲁棒性能。分别从传声器失配误差影响下阵列波束图扰动的统计分析以及白噪声增益分析两个方面揭示了影响多项式结构宽带波束形成器鲁棒性能的内在规律。发现当波束主瓣指向处于可调范围边缘时,与指向中心相比,多项式结构波束形成器的鲁棒性能变差。然后分析了影响多项式结构宽带波束形成器阵列指向性指数的主要因素,指出了多项式结构的截断误差会导致宽带波束形成器的主瓣指向出现偏差,进而造成阵列指向性指数性能下降。为了克服现有设计结构存在的缺陷,提出了一种具有波束主瓣指向自校正能力的改进设计结构。与现有设计结构不同,所提出的设计结构中的调向功能模块能够充分反映出波束期望指向和实际指向之间存在的非线性关系,避免了现有设计方法因采用线性调向功能模块而造成较大波束指向偏差的问题,有效改善了多项式结构宽带波束形成器的性能。      

刚性圆柱体上圆阵波束形成性能分析

以环绕在刚性圆柱体上的均匀圆阵为阵列模型,分析了刚性圆柱体对圆阵响应的影响,在声场分解所得到的相位模式空间,推导了相位模式波束形成阵列和延迟求和波束形成阵列的白噪声增益和指向性指数的定量表达式,分析了它们在不同频率段的变化规律。在存在和不存在刚性圆柱体两种情况下,对两种阵列性能进行了对比分析。研究了圆阵与圆柱体之间存在间隙时对阵列接收信号及阵列性能的影响。仿真结果表明,在两种情况下相位模式波束形成处理都具有频率不变的指向性,但在低频段牺牲了阵列的鲁棒性;延迟求和波束形成处理在整个频率段具有较好的鲁棒性,但随频率的降低其波束指向性下降,直至变成各向同性阵列。存在刚性圆柱体时,两种阵列鲁棒性均有了一定的提高,前者的指向性不变,而后者的指向性得到了提高。圆阵与圆柱体之间的间隙会降低阵列在高频段的鲁棒性。      

融合插值点优化的多项式结构宽带波束形成器设计方法*

多项式结构设计方法是主瓣指向可调宽带波束形成器设计的一类重要方法。多项式结构的阶数是有限的,导致主瓣实际指向与期望指向之间存在偏差,因而影响了波束形成器的指向性指数。针对这一问题,该文提出了一种基于插值点优化的多项式结构宽带波束形成器设计方法。首先,引入多项式结构插值点处阵列响应的空间导数约束,以减小主瓣指向偏差;进而利用粒子群优化算法对多项式结构中的插值点进行优化,以充分利用插值点位置提供的自由度进一步提升多项式结构宽带波束形成器的性能。优化设计结果表明,与现有设计方法相比,该文提出的方法不仅降低了主瓣的指向偏差,同时也提高了指向性指数,有效改善了多项式结构宽带波束形成器的性能。     

协方差矩阵重构的稳健自适应波束形成算法

针对协方差矩阵含有期望信号成分以及波束指向角失配时,传统自适应波束形成器性能严重下降的问题,提出了协方差矩阵重构的稳健自适应波束形成算法。该算法将全空域划分成若干互不重叠的区域,分别对应干扰区域与信号区域,先利用Capon波束形成器对干扰区域积分,由此构造出干扰协方差矩阵。然后,利用标准Capon波束形成器的波束域MUSIC谱估计法对信号区域积分,重构出信号协方差矩阵,以其主特征向量作为期望信号导引向量估计。由于算法重构了干扰加噪声协方差矩阵并对导引向量进行了修正,保证了自适应波束形成器的性能。理论分析和仿真实验结果表明,算法在训练数据含有期望信号成分和波束指向角度失配情况下具有良好的性能。      

球体表面圆环阵模态域稳健高增益波束形成方法研究

通过对考虑障板影响下噪声互谱矩阵的精确求解,给出了一种计算刚性球体表面圆环阵阵增益的模型,并在此基础上提出了一种模态域二阶锥规划稳健高增益波束形成方法。该方法根据相位模态波束形成理论,将阵元域稳健加权向量转换为模态域的稳健模态系数,从而设计出不同模态阶数下的稳健高增益波束。由于采用了白噪声增益约束以及低频段较低的模态阶数,该方法提高了超增益波束形成器的稳健性。仿真结果表明该方法能够提供更多的稳健波束形成的方案,在多个关联的波束性能指标之间获得比常规方法和阵元域稳健性方法更合理的折衷。      

基于小波变换和非负支撑域递归逆滤波算法的盲目图像复原

提出了一种将自适应正则化方法与非负支撑域递归逆滤波(NAS-RIF)算法相结合用于小波域的盲图像复原算法.该算法先对降质图像进行小波分解,得到了图像在不同子频段的信息.在各个子频段采用NAS-RIF算法进行复原.针对各个子频段内图像的频率和方向特性,分别引入了不同的正则化约束项.在各个子频段估计出噪声方差,提出了根据噪声方差和图像局部方差来选取正则化参数.分别对两幅模糊图像进行了仿真实验,复原结果取得的信噪比分别为19.66 dB和23.86 dB.实验结果表明,复原效果相对于空间自适应正则化方法有一定的提高.     

柱体表面圆环阵稳健高增益波束形成的模态域直接优化方法研究

以无限长刚性柱体表面圆环阵为研究对象,提出了一种直接模态域稳健高增益波束形成方法,并将其应用到更实际的有限长刚性柱体表面圆环阵。该方法运用相位模态理论将波束响应表示成特征波束与模态系数的乘积后求和的形式,并结合散射理论求解了其模态域的噪声互谱矩阵。与已有的模态域稳健高增益方法需利用阵元域权值间接求取近似模态权系数不同的是,该方法利用二阶锥规划,在满足相关约束条件下直接给出了最优的模态权系数。仿真结果表明,直接模态域方法不仅能通过白噪声增益约束提高稳健性,在低频段还能通过改变模态阶数更灵活地在阵增益和稳健性之间折中,同时对旁瓣级等其它性能指标也能很好地控制,可用以有效设计实用的稳健高增益波束形成器。      

从阵元域到模态域阵列信号处理

阐述了模态域阵列信号处理技术.以球面阵为例,从阵元域信号处理基础出发,推导了对应的模态域阵列信号处理表达式,其中重点推导了模态域波束形成器与时域宽带波束形成器设计及其实现方法,设计实例验证了方法的有效性.结果表明模态域处理与经典的阵元域处理具有相似的形式,因此可以将比较成熟的阵元域算法推广到模态域.相比于阵元域处理,模...     

Robust supergain beamforming for circular array via second-order cone programming

The phenomenon of supergain for a circular array and its robust beamforming are presented. The coplanar superdirective array gain of the circular array, although it is not so extreme as an endfire line array, outperforms a lot over that of a conventional delay-and-sum beamformer in isotropic noise fields when the inter-element spacings are much smaller than one-half wavelength. However, optimum beamforming algorithms can be extremely sensitive to slight errors in array characteristics. The performance are known to degrade significantly if some of underlying assumptions on the sensor array is violated. Therefore, white noise gain constraint is used to improve the robustness of the supergain beamformer against random errors. We show that the design of the weight vector of robust supergain beamformer can be reformulated as a form of second-order cone programming and resolved efficiently via the well-established interior point method. Results of computer simulation for a 24-element circular array confirm satisfactory performance of the approach proposed in this paper.     

A method of improving overall resolution in ultrasonic array imaging using spatio-temporal deconvolution

In this paper a beamforming method for ultrasonic array imaging is presented that performs both spatial and temporal deconvolution based on a minimum mean square error (MMSE) criteria. The presented MMSE receive mode beamformer performs a regularized inversion of the propagation operator for the ultrasonic array system at hand. The MMSE beamformer accounts for the transmit and receive processes, defined in terms of finite array element sizes, transmit focusing laws and electrical transducer characteristics. The MMSE beamformer is compared to the traditional delay-and-sum (DAS) beamformer with respect to both resolution and signal-to-noise ratio. The two algorithms are compared using both simulated and measured data. The simulated data was obtained using ultrasonic field simulations and the measured data was acquired using a linear phased array imaging wire targets in water. The results show that the MMSE beamformer has superior temporal and lateral resolution compared to DAS. It is also shown that the MMSE beamformer can be expressed as a filter bank, which enables parallel processing at high frame rates.     

Eigenspace-based minimum variance beamformer combined with Wiener postfilter for medical ultrasound imaging

In this paper, a new beamformer which combines the eigenspace-based minimum variance (ESBMV) beamformer with the Wiener postfilter is proposed for medical ultrasound imaging. The primary goal of this work is to further improve the medical ultrasound imaging quality on the basis of the ESBMV beamformer. In this method, we optimize the ESBMV weights with a Wiener postfilter. With the optimization of the Wiener postfilter, the output power of the new beamformer becomes closer to the actual signal power at the imaging point than the ESBMV beamformer. Different from the ordinary Wiener postfilter, the output signal and noise power needed in calculating the Wiener postfilter are estimated respectively by the orthogonal signal subspace and noise subspace constructed from the eigenstructure of the sample covariance matrix.We demonstrate the performance of the new beamformer when resolving point scatterers and cyst phantom using both simulated data and experimental data and compare it with the delay-and-sum (DAS), the minimum variance (MV) and the ESBMV beamformer. We use the full width at half maximum (FWHM) and the peak-side-lobe level (PSL) to quantify the performance of imaging resolution and the contrast ratio (CR) to quantify the performance of imaging contrast. The FWHM of the new beamformer is only 15%, 50% and 50% of those of the DAS, MV and ESBMV beamformer, while the PSL is 127.2 dB, 115 dB and 60 dB lower. What is more, an improvement of 239.8%, 232.5% and 32.9% in CR using simulated data and an improvement of 814%, 1410.7% and 86.7% in CR using experimental data are achieved compared to the DAS, MV and ESBMV beamformer respectively. In addition, the effect of the sound speed error is investigated by artificially overestimating the speed used in calculating the propagation delay and the results show that the new beamformer provides better robustness against the sound speed errors. Therefore, the proposed beamformer offers a better performance than the DAS, MV and ESBMV beamformer, showing its potential in medical ultrasound imaging.     

Array gain for a conformal acoustic vector sensor array:An experimental study

An acoustic vector sensor can measure the components of particle velocity and the acoustic pressure at the same point simultaneously, which provides a larger array gain against the ambient noise and a higher angular resolution than the omnidirectional pressure sensor. This paper presents an experimental study of array gain for a conformal acoustic vector sensor array in a practical environment. First, the manifold vector is calculated using the real measured data so that the effects of array mismatches can be minimized. Second, an optimal beamformer with a specific spatial response on the basis of the stable directivity of the ambient noise is designed, which can effectively suppress the ambient noise.Experimental results show that this beamformer for the conformal acoustic vector sensor array provides good signal-tonoise ratio enhancement and is more advantageous than the delay-and-sum and minimum variance distortionless response beamformers.     

Imaging of directional distributed noise sources

This study relates to the acoustic imaging of noise sources that are distributed and strongly directional, such as in turbulent jets. The goal is to generate high-resolution noise source maps with self-consistency, i.e., their integration over the extent of the noise source region gives the far-field pressure auto-spectrum for a particular emission direction. Self-consistency is possible by including a directivity factor in the formulation of the source cross-spectral density. The resulting source distribution is based on the complex coherence, rather than the cross-spectrum, of the measured acoustic field. For jet noise, whose spectral nature changes with emission angle, it is necessary to conduct the measurements with a narrow-aperture array. Three coherence-based imaging methods were applied to a Mach 0.9 turbulent jet: delay-and-sum beamforming; deconvolution of the beamformer output; and direct spectral estimation that relies on minimizing the difference between the measured and modeled coherences of the acoustic field. The delay-and-sum beamforming generates noise source maps with strong spatial distortions and sidelobes. Deconvolution leads to a five-fold improvement in spatial resolution and significantly reduces the intensity of the sidelobes. The direct spectral estimation produces maps very similar to those obtained by deconvolution. The coherence-based noise source maps, obtained by deconvolution or direct spectral estimation, are similar at small and large observation angles relative to the jet axis.     

Broadband optimal contrast resolution beamforming

This paper proposes a novel receive beamformer architecture for broadband imaging systems that uses unique finite impulse response (FIR) filters on each channel. The conventional delay-and-sum (DAS) beamformer applies receive apodization by weighting the signal on each receive channel prior to beam summation. Our proposed FIR beamformer passes the focused receive radio frequency (RF) signals through multi-tap FIR filters on each receive channel prior to summation. The receive FIR filters are constructed to maximize the contrast resolution of the system’s spatial response. The broadband FIR beamformer produces spatial point spread functions (PSFs) with narrower mainlobe widths and lower sidelobe levels than spatial PSFs produced by the conventional DAS beamformer.We present simulation results showing that FIR filters of modest tap lengths (3-7) can yield marked improvement in image contrast and point resolution. Specifically we show that 7-tap FIR filters can reduce sidelobe and grating lobe energy by 30 dB and improve contrast resolution by as much as 20 dB compared to conventional apodization profiles. This improvement in contrast resolution comes at the expense of a decrease in beamformer sensitivity. We investigate the effects of phase aberration and show in simulation results that the multi-tap FIR beamformer outperforms the unaberrated DAS beamformer by 8-12 dB even in the presence of moderate aberration characterized by a root-mean-square strength of 28 ns and a full-width at half-maximum correlation length of 3.6 mm. We show experimental results wherein multi-tap FIR filters decrease sidelobe energy in the resulting 2D spatial response while achieving a narrow mainlobe. We also show results where the FIR beamformer improves the contrast to noise ratio (CNR) in simulated B-mode cyst images by more than 4 dB. Our algorithm has the potential to significantly improve ultrasound beamforming in any application where the system response is reasonably well characterized. Furthermore, this algorithm can be used to increase contrast and resolution in one-way beamforming systems such as acousto-optic and opto-acoustic imaging.     

Optimum beamformer in correlated source environments

A new method for overcoming signal cancellation problems due to correlated interferences which occur in a minimum variance distortionless response beamformer is proposed. Instead of decorrelating the correlated interferences, the coherently combining signal-to-interference plus noise ratio (CC-SINR) beamformer regards them as replicas of the desired signal and coherently combines them with the desired signal. This method uses an eigenvector constraint that suppresses only noise and uncorrelated interferences but retains the desired signal and correlated interferences. The CC-SINR beamformer does not require any preliminary information on correlated interferences. The signal-to-interference plus noise ratio (SINR) of the proposed beamformer output was compared to that of a conventional SINR beamformer when correlated interference, uncorrelated interference, and white noise exist. In addition, various key parameters that affect the performance of the beamformer, such as signal-to-noise ratio, uncorrelated interference-to-noise ratio, angular separation between signals, attenuation factor, phase delay of correlated interference, and the number of sensors were analyzed. All of the experimental results were in good agreement with the analytical results.     

圆柱阵子阵分级处理的稳健超指向性波束形成方法

提出了圆柱阵子阵分级处理的稳健超指向性波束形成方法。首先建立了圆柱阵分两级子阵进行波束形成的信号模型,接着利用空间均匀噪声场中噪声互谱矩阵的循环特性得到基于特征波束分解与综合模型的圆柱阵超指向性的最优解,然后仿真研究了其误差敏感度函数、阵增益和波束图等性能指标,并与圆柱阵的传统全局处理方法进行了对比。提出的分两级子阵处理的超指向性方法与传统全局处理方法相比不仅降低了数据存储量和波束形成计算量,而且进一步提升了稳健性,并且在低频段的阵增益远远高于常规波束形成的值,对水下声呐阵列的设计具有一定的参考价值。      

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. II. Experimental verification of the predictions

A method to predict the amount of noise reduction which can be achieved using a two-microphone adaptive beamforming noise reduction system for hearing aids [J. Acoust. Soc. Am. 109, 1123 (2001)] is verified experimentally. 34 experiments are performed in real environments and 58 in simulated environments and the results are compared to the predictions. In all experiments, one noise source and one target signal source are present. Starting from a setting in a moderately reverberant room (reverberation time 0.42 s, volume 34 m3, distance between listener and either sound source 1 m, length of the adaptive filter 25 ms), eight different parameters of the acoustical environment and three different design parameters of the adaptive beamformer were systematically varied. For those experiments, in which the direct-to-reverberant ratios of the noise signal is +3 dB or less, the difference between the predicted and the measured improvement in signal-to-noise ratio (SNR) is -0.21+/-0.59 dB for real environments and -0.25+/-0.51 dB for simulated environments (average +/- standard deviation). At higher direct-to-reverberant ratios, SNR improvement is systematically underestimated by up to 5.34 dB. The parameters with the greatest influence on the performance of the adaptive beamformer have been found to be the direct-to-reverberant ratio of the noise source, the reverberation time of the acoustic environment, and the length of the adaptive filter.     

Localization of virtual sound sources with bilateral hearing aids in realistic acoustical scenes

Sound localization with hearing aids has traditionally been investigated in artificial laboratory settings. These settings are not representative of environments in which hearing aids are used. With individual Head-Related Transfer Functions (HRTFs) and room simulations, realistic environments can be reproduced and the performance of hearing aid algorithms can be evaluated. In this study, four different environments with background noise have been implemented in which listeners had to localize different sound sources. The HRTFs were measured inside the ear canals of the test subjects and by the microphones of Behind-The-Ear (BTEs) hearing aids. In the first experiment the system for virtual acoustics was evaluated by comparing perceptual sound localization results for the four scenes in a real room with a simulated one. In the second experiment, sound localization with three BTE algorithms, an omnidirectional microphone, a monaural cardioid-shaped beamformer and a monaural noise canceler, was examined. The results showed that the system for generating virtual environments is a reliable tool to evaluate sound localization with hearing aids. With BTE hearing aids localization performance decreased and the number of front-back confusions was at chance level. The beamformer, due to its directivity characteristics, allowed the listener to resolve the front-back ambiguity.