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1.
提出了一种MVDR(最小方差无失真响应)的改进算法,用以解决常规MVDR算法由于阵形时变而出现的性能下降问题。在获得时变阵形估计数据的基础上,该算法以统计时段内的平均阵形为基准阵形,在每个扫描方向上根据实际阵形和基准阵形的差异对阵列互谱矩阵多样本进行相位补偿,从而实现统计时段内的互谱矩阵多样本相干累加和目标检测。数值仿真与海上实验数据处理结果表明:与传统MVDR算法相比,改进算法有效缓解了时变阵形下的目标测向角度模糊问题,可提高拖线阵目标左右舷分辨性能、增强弱目标检测能力。  相似文献   

2.
十字阵短时宽带声源实时定向算法   总被引:5,自引:1,他引:4       下载免费PDF全文
程萍  陈建峰  马驰  张竹 《应用声学》2012,31(2):123-129
面向短时宽带声源实时定向问题,提出了一种基于互功率谱时延估计的十字阵定向优化算法。针对该方法估计结果离散且呈不均匀分布的特点,将观测平面划分为四个测量区域,并利用不同阵元组合分别处理,解决实时性与估计成功率的矛盾;依据互相关函数的特点,设计了若干判断准则,排除由于数据取样短造成的异常时延估计,改善算法的可靠性;采用频域插值方法,一定程度上提高时延估计精度,从而提高定向精度。MATLAB仿真和DSP系统实验表明,这种方法在实际应用中有效提高了对短时宽带声源定向的性能。  相似文献   

3.
快速收敛最小方差无畸变响应算法研究及应用   总被引:2,自引:0,他引:2  
常规最小方差无畸变响应(MVDR)自适应波束形成是一种高分辨窄带波束形成器,它是利用实际声场的窄带互谱密度矩阵(CSDM)估计出自适应波束形成权向量.在实际应用中,MVDR算法需要较长的观测时间估计协方差矩阵,不利于对高速运动目标进行定位;对于宽带目标信号, MVDR算法需要对每一个CSDM进行求逆运算,计算量较大;在相干源条件下,目标信号之间会发生"对消"现象,MVDR算法性能急剧恶化.本文提出了基于子带子阵处理的快速收敛MVDR自适应波束形成方法.首先将全频带划分成一组子带,将接收线阵划分成一组子阵,然后对每一子带计算降维的驾驶协方差矩阵(STCM),从而得到快速收敛MVDR自适应波束形成的权值和空间谱估计结果.同时采用双向空间平滑方法对相干源进行MVDR空间谱估计.仿真和海试数据处理结果表明该算法在保证高分辨力的同时,具有瞬时收敛的性能,双向空间平滑技术具有良好的解相干性能.  相似文献   

4.
基于两次谱分析的时延估计方法研究   总被引:2,自引:0,他引:2       下载免费PDF全文
张卫平  张合  王伟策  刘强  方向 《应用声学》2008,27(3):222-226
时延估计是目标定位跟踪系统的关键技术之一,在水声、雷达、声探测等领域广泛应用。时延估计的基本方法是互相关法和相位谱法。互相关法时延估计分辨率与信号带宽近似成反比,因此很难估计多目标时延。相位谱时延估计只能估计单目标时延,并且存在相位解绕问题。本文提出了两次谱分析时延估计方法,即将互功率谱函数再次进行谱估计,二次谱峰值位置间距即为时延估计,这种方法既能够估计单目标时延,又能够估计多目标时延,并且不用相位解绕。仿真计算验证了两次谱时延估计方法的可行性。  相似文献   

5.
一种利用高阶谱相位数据进行时延估计的新方法   总被引:3,自引:0,他引:3  
本文提出了一种利用高阶谱相位数据进行时延估计的新方法。对于无噪声干扰或是不相关高斯噪声干扰的信号来说,互谱相位数据时延估计能给出较好的估计结果。但是若存在相关高斯噪声干扰时,利用高阶谱相位数据进行时延估计就有较突出的优点,其精度高于互谱相位数据时延估计法。在频域内经过适当加权,能更有效地提高估计精度。  相似文献   

6.
基于二次相关的语音信号时延估计改进算法   总被引:1,自引:1,他引:0  
刘敏  曾毓敏  张铭  李晨 《应用声学》2016,35(3):255-264
目前语音信号的时延估计研究,大部分采用的是广义互相关算法。然而,广义互相关时延估计算法易受噪声和混响环境影响。为此,本文提出了一种基于二次相关的语音信号时延估计改进算法,该算法对语音信号进行二次互相关运算,并结合Hilbert变换,对二次互相关峰值进行进一步的锐化处理,使得反映时延的峰值点检测更为准确。实验结果表明,改进的时延估计方法在非平稳的语音信号中能够有效地抑制噪声干扰,且在不同混响条件下时延估计具有更好的性能。  相似文献   

7.
通过分析复杂环境中不同频带声信号时延估计的差异,提出多频带期望值最大时延估计方法。为了使各频带之间无重叠,该方法采用独立分带划分声信号不同频带,然后计算各频带广义互相关函数,并对子带广义互相关函数建立最大似然模型,最后利用期望值最大算法将多维优化转为一维优化的迭代式,获得最优子带广义互相关函数,在此基础上估计声信号的时延信息。数据仿真和实际实验结果表明,多频带期望值最大化时延估计相较常规时延估计有效估计值的百分比提升了10%,并将最优频带互相关函数应用到该定位算法中,在网格间距为0.3 m时,得到的峰值区域汇聚更明显,定位效果更好。  相似文献   

8.
时胜国  李赢 《应用声学》2019,38(4):530-539
针对宽带相干目标的远程探测问题,本文提出一种基于声压振速联合处理和矢量重构的声矢量圆阵MVDR波束形成方法。该方法利用相位模态变换技术,将声矢量圆阵变换为与信号频率无关的虚拟线阵,并构建虚拟线阵声压与组合振速的互协方差矩阵,利用声压与振速各分量间的空间相关性有效地抑制各向同性环境噪声;并对宽带相干信号的互协方差矩阵进行矢量重构,即将最大特征值对应的特征向量划分为相互重叠的子向量,从而构建前/后向Hermitian矩阵;最后,基于MVDR波束形成器实现宽带相干目标的方位估计。仿真计算和实验数据处理结果表明,该方法具较强的解相干能力和噪声抑制能力以及较高的方位估计性能。  相似文献   

9.
针对无源雷达中时延估计辐射源信号未知的情况,构建了一种新的时延最大似然估计模型.根据模型特点利用快速傅里叶变换(FFT)的计算方法实现时延估计.为了提高估计的精度,采用马尔科夫链蒙特卡罗(MCMC)抽样的方法估计时延值.该方法不需峰值检测,可直接给出时延估计结果.并推导了该模型下的时延估计的克拉美罗界(CRLB).仿真实验表明,MCMC算法可适用于窄带和宽带信号的时延估计;在样本相同的条件下,MCMC算法估计精度高于重要性采样(IS)算法和基于峰值检测的ML算法,计算复杂度低于IS算法,且MCMC算法可直接估计采样间隔非整数倍的时延.  相似文献   

10.
有源声呐感兴趣的参量是目标距离和径向速度,它们无法直接观测得到,需要通过估计而获得。利用波导多路径环境多目标时延-多普勒模型,可以导出采样互模糊度函数均值是发射信号自模糊度函数与广义目标反射性密度函数的两维卷积,其中广义目标反射性密度函数为信道扩展函数与目标反射性密度函数的两维卷积。依据信息理论最小Csiszar鉴别准则,可导出R-L(Richardson-Lucy)迭代解卷算法,对采样互模糊度函数均值进行两维迭代解卷积,消除发射信号和信道引入的模糊,序贯地实现时延-多普勒两维像的估计,进而获得多目标的时延和多普勒参量估计。仿真结果和海上实验数据分析验证了R-L解卷算法的可行性和有效性,较之常规的匹配滤波和维纳滤波算法,R-L算法有效地提高了时延和多普勒估计的分辨力和精度。  相似文献   

11.
针对噪声环境下微小气体泄漏难以准确定位的问题,提出了一种基于改进最小方差无失真响应角度谱算法的气体泄漏定位方法。该算法通过引入信噪比追踪加权的方式,提取受噪声影响较小且单个声源能量占优的时频支撑域,并通过Softplus激活函数自适应地调整不同频率分量对角度谱函数的贡献,增加泄漏声源占优的时频域权重;此外,引入基于时频稀疏性的分频带处理,使各子频带内存在一个主导声源能量占优,抑制低频段噪声能量的积累同时避免高频混叠现象。通过软件仿真计算以及实验验证算法的性能,结果表明改进最小方差无失真响应角度谱算法可以实现气体泄漏源的精准定位,定位结果的最大误差在3.5°以内。相比传统算法,该方法在低信噪比和低采样点数下有更高的稳定性、抗噪能力及准确率,可为气体泄漏定位的实际应用提供一定的参考价值。  相似文献   

12.
Direction of arrival(DOA) estimation and signal recovery is the base of the underwater target localization,tracking and recognition.Based on the compressed sensing theory,a method for DOA estimation and source signal recovery is proposed using the single snapshot processing of the received array signal in frequency domain.The received array signal are transformed to frequency domain,and the single snapshot data in frequency domain are regarded as the measured data of the compressed sensing.According to the frequency,searching orientation and array manifold,the overcomplete array manifold is constructed as the sensing matrix of the compressed sensing.Both the target signal and power of the searching orientation are estimated by the basis pursuit method to complete DOA estimation and signal recovery.Simulation results show that the proposed method has a number of advantages over the minimum variance distortionless response(MVDR) method,including improved robustness to noise,fewer requirement in number of sensors and snapshots.And the correlation coefficient of the signal reaches up to 0.89.Experiment results in real environments verify that the proposed method performs more effectively in the detection of weak targets than the MVDR method and can be applied to real sonar system.  相似文献   

13.
王璐  许录平  张华  罗楠 《物理学报》2013,62(13):139702-139702
为了提高脉冲星辐射脉冲信号的检测速度和在低信噪比下的检测效果, 提出了一种基于S变换的脉冲星辐射脉冲信号检测算法. 文中证明了高斯白噪声S变换域功率谱服从自由度为2的卡方分布, 基于此对累积信号S变换域功率谱进行阈值处理,累加阈值处理后的时频功率谱作为统计量进行检测. 此外阈值处理后的功率谱也可用来测量脉冲星信号的时间延迟. 仿真实验验证了本文算法的有效性,其检测性能优于同类的基于高斯分布模型的检测算法, 同时还可以在一定精度下给出脉冲星信号的时间延迟值. 关键词: 脉冲星 卡方分布 S变换域检测 时延测量  相似文献   

14.
针对宽带高分辨方位估计存在方位估计偏差大、算法复杂度高等问题,提出了一种基于条件波数谱密度(Conditional Wavenumber Spectral Density based,CWSD-based)的宽带高分辨方位谱估计算法.该算法利用条件波数谱密度将阵列信号转换到频率-波数空间,宽带信号能量在该空间的坐标呈现与入射角相关的线性分布,通过借鉴直线检测原理,实现邻近目标的高分辨方位估计,且无需预估角度和信源数等信息。仿真结果表明,该算法理论分辨率与处理最高频率成反比,估计均方误差约为0.1°,对阵形畸变鲁棒,运算效率高。海上试验数据表明,本文方法在方位分辨率、弱目标检测、非目标向噪声抑制、稳健性等方面都优于宽带常规波束形成和最小方差无畸变算法,在实际海洋中可实现超低旁瓣高分辨波达方向估计。   相似文献   

15.
王超  笪良龙  韩梅  孙芹东  王文龙 《声学学报》2021,46(6):1050-1058
针对单矢量水听器海上目标探测问题,利用稀疏近似最小方差(Sparse Asymptotic Minimum Variance,SAMV)算法进行目标方位估计,该算法利用单矢量水听器自身具有阵列流形的特点,将整个扫描空间离散化,目标方位分布于某一离散方向位置上,利用空间信号的稀疏性可提高目标方位估计性能。仿真结果表明,SAMV算法在各信噪比条件下方位估计噪声背景级明显优于常规波束形成(Conventional Beam Forming,CBF)算法和最小方差无失真响应(Minimum Variance Distortionless Response,MVDR)算法,当信噪比大于0dB时,该算法测向结果均方根误差小于2°,且SAMV算法具有更好的空间方位分辨能力。消声水池和海上声学浮标海上试验数据处理结果表明,SAMV算法给出了噪声背景级更低的目标方位历程图,有效验证了SAMV算法对海上目标的探测性能及其有效性。   相似文献   

16.
In this paper, a single-channel speech enhancement algorithm based on non-linear and multi-band Adaptive Gain Control (AGC) is proposed. The algorithm requires neither Signal-to-Noise Ratio (SNR) nor noise parameters estimation. It reduces the background noise in the temporal domain rather than the spectral domain using a non-linear and automatically adjustable gain function for multi-band AGC. The gain function varies in time and is deduced from the temporal envelope of each frequency band to highly compress the frequency regions where noise is present and lightly compress the frequency regions where speech is present. Objective evaluation using the PESQ (Perceptual Evaluation of Speech Quality) metric shows that the proposed algorithm performs better than three benchmarks, namely: the spectral subtraction, the Wiener filter based on a priori SNR estimation and a band-pass modulation filtering algorithm. In addition, blind subjective tests show that the proposed algorithm introduces less musical noise compared to the benchmark algorithms and was preferred 78.8% of the time in terms of signal quality. The proposed algorithm is implemented in a miniature low power digital signal processor to validate its feasibility and complexity for smart hearing protection in noisy environments.  相似文献   

17.
In this paper, a fundamental frequency (F(0)) tracking algorithm is presented that is extremely robust for both high quality and telephone speech, at signal to noise ratios ranging from clean speech to very noisy speech. The algorithm is named "YAAPT," for "yet another algorithm for pitch tracking." The algorithm is based on a combination of time domain processing, using the normalized cross correlation, and frequency domain processing. Major steps include processing of the original acoustic signal and a nonlinearly processed version of the signal, the use of a new method for computing a modified autocorrelation function that incorporates information from multiple spectral harmonic peaks, peak picking to select multiple F(0) candidates and associated figures of merit, and extensive use of dynamic programming to find the "best" track among the multiple F(0) candidates. The algorithm was evaluated by using three databases and compared to three other published F(0) tracking algorithms by using both high quality and telephone speech for various noise conditions. For clean speech, the error rates obtained are comparable to those obtained with the best results reported for any other algorithm; for noisy telephone speech, the error rates obtained are lower than those obtained with other methods.  相似文献   

18.
Time delay estimation (TDE) plays an important role in many engineering appli-cations. A new time delay estimation configuration, the quadratic weighting of the frequency domain adaptive TDE model, is put forward. The quadratic weighting of the frequency domainSCOT (Smoothed Coherence Transform) and ML (Maximum Likelihood) adaptive TDE algo-rithms are presented, respectively. The variance of the quadratic weighting of the frequency domain SCOT algorithm is derived. Then the proposed algorithms are applied in the TDE of helicopter passive acoustic location. The simulation results are presented which verify that the proposed algorithm has better performance in the low signal to noise ratio.  相似文献   

19.
This paper addresses the problem of noise reduction in the time domain where the clean speech sample at every time instant is estimated by filtering a vector of the noisy speech signal. Such a clean speech estimate consists of both the filtered speech and residual noise (filtered noise) as the noisy vector is the sum of the clean speech and noise vectors. Traditionally, the filtered speech is treated as the desired signal after noise reduction. This paper proposes to decompose the clean speech vector into two orthogonal components: one is correlated and the other is uncorrelated with the current clean speech sample. While the correlated component helps estimate the clean speech, it is shown that the uncorrelated component interferes with the estimation, just as the additive noise. Based on this orthogonal decomposition, the paper presents a way to define the error signal and cost functions and addresses the issue of how to design different optimal noise reduction filters by optimizing these cost functions. Specifically, it discusses how to design the maximum SNR filter, the Wiener filter, the minimum variance distortionless response (MVDR) filter, the tradeoff filter, and the linearly constrained minimum variance (LCMV) filter. It demonstrates that the maximum SNR, Wiener, MVDR, and tradeoff filters are identical up to a scaling factor. It also shows from the orthogonal decomposition that many performance measures can be defined, which seem to be more appropriate than the traditional ones for the evaluation of the noise reduction filters.  相似文献   

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