The effects of IF bandpass mismatch errors on adaptive cancellation

The effects of IF bandpass mismatch errors on adaptive cancellers are investigated. Frequency mismatch errors occur because of errors in the synthesis process of the bandpass filters which are designed to be identical and are in each input channel. Tapped-delay line transversal filters can be used to compensate for these frequency mismatches and thus improve cancellation performance. A pole/zero error model of the filters is developed whereby closed-form solutions of the maximum achievable average cancellation are obtained. This cancellation is a function of the order of the ideally matched frequency filters, the number of time-delay taps in the compensating transversal filter, the bandwidth-tapped time-delay product, and the constraints on these parameters. A design procedure is outlined for optimizing the canceller with respect to these parameters and their constraints; specifically, results are presented for Butterworth-type input filters. It is shown that an arbitrarily low output noise residue cannot be achieved by arbitrarily increasing the number of time-delay taps     

The effect of I,Q mismatch errors on adaptive cancellation

The effects of in-phase (I) and quadrature-phase (Q) amplitude errors and low-pass-filter (LPF) errors on adaptive cancellers are investigated. I,Q errors occur because of errors in the synthesis process of the mixers and LPFs designed to be identical for each input channel. These I,Q errors among the channels result in cancellation degradation. Tapped delay line transversal filters have been proposed as a way to compensate for these errors and thus improve cancellation performance. However, it is shown that if there is any LPF mismatch, then transversal filtering has a small effect on improving canceler performance. The use of individual I,Q adaptive transversal filter weighting is suggested as a means of completely eliminating the phase amplitude errors, and making the canceler performance responsive to transversal filter compensation     

Controlled Alteration of Wideband Radar Oata

A procedure is described for obtaining weights for a transversal filter which will degrade the range resolution and alter the sidelobe levels of the sampled version of a time envelope whose spectrum is band. limited and known. The two general cases of super-and sub-Nyquist rate sampling are discussed.     

A Computationally Efficient Approximation to a Discrete-Time Matched Filter

When the number of filter coefficients is large, the solution of the discrete-time matched filter equation can be computationally difficult. In this paper several techniques are presented for approximating the impulse response of a matched filter without actually solving the matched filter equation. The performance of these approximating filters is analyzed and compared with the performance of the matched filter. It is also shown that an approximation which is best in a mean-squared-error sense is not necessarily best in terms of output signal-to-noise ratio.     

Creating complex signal samples from a band-limited real signal

A very efficient method of creating complex signal samples from a band-limited real signal is presented. Because the method employs a simple mixer followed by one analog-to-digital (A/D) converter, plus a finite-duration impulse response (FIR) filter for image band rejection, there is no phase distortion in the resulting sampled signal. The method is more efficient than competing methods based on infinite-duration impulse response (IIR) filters     

一种基于数字综合算法的MTD滤波器设计方法

利用空间阵列方向图合成中的数字综合算法 ,提出了一种新的MTD滤波器设计方法。该方法通过在滤波器频率响应的副瓣区放置大量的干扰信号 ,改变干扰信号的功率强度 ,从而自适应地控制滤波器频率响应的副瓣电平。这种方法所设计的MTD滤波器 ,其频率响应不仅在零频附近有宽而深的零陷 ,而且在其他副瓣区域可以为任意形状 ,具有较好的实用性。设计实例证实了这种方法的有效性。     

低通滤波器的阶跃响应及其对动态校准的影响

动态测试系统在利用阶跃信号进行动态校准时,系统中的低通滤波器环节所产生的衰减振荡可能会误导测试人员,特别是对于带传压管道的动态压力测试系统。仿真与测试结果表明,各种模拟低通滤波器和IIR数字低通滤波器的阶跃响应都会出现一定的衰减振荡。针对带低通滤波器的系统以及其他系统的阶跃激励校准,通过对简单FFT频响分析方法、矩形单脉冲法、冲激响应法三种方法的比较,发现矩形单脉冲法能更正确地对阶跃校准数据进行频响分析。     

A minimax approach to the design of Doppler filters

The detection of a target in correlated clutter, thermal noise, and extraneous interference is considered. The amplitude, phase and Doppler frequency of the signal are not known a priori. A general criterion is presented which measures the performance of a suboptimal test relative to an optimal test. The criterion is encompassed into a design procedure used to design Doppler filters. The procedure allows many design considerations to be taken into account, and results in a design which attempts to minimize the number of filters required. For low dimensionality the procedure results in single filter designs; for higher dimensionality multiple filters are designed. The performances of these systems are compared with the results obtained by Emerson (1978) and Andrews (1974). It is found that the procedure yields good filter designs under general conditions and may reduce the number of filters required compared with classical designs     

Precise navigation using adaptive FIR filtering and time domainspectral estimation

This paper introduces a new low cost, short range, positioning system based on adaptive finite impulse response (FIR) filtering and time domain spectral estimation. The system can determine absolute positions with a high degree of accuracy and is well suited for real time navigation. The approach is based upon signal processing techniques and a priori knowledge of the system transfer function. The first step is to measure the phase response of the linear transfer function and then using a FIR filter the time response of the system can be determined. The FIR filter computes the time response by performing a deconvolution between the measured phase response, and the complex conjugate of the transfer function. By correlating the known input impulse response with the output of the FIR filter, an error term is generated. The time delay of the system is determined by adjusting the FIR filter coefficients to minimize the error term. Simulated analysis of the system indicates a worst case error of ±16 cm     

The Processing of Band-Limited Measurements; Filtering Techniques in the Least Squares Context and in the Presence of Data Gaps

This paper discusses the treatment of correlated measurements in the least squares context. We focus on the processing of band-limited measurements and on long time series with a constant sampling interval. Time domain as well as frequency domain approaches were discussed to offer different ways to integrate the filtering process into the optimization scheme as good as possible. The focus was on long equispaced data sets. The application of discrete filters in the space domain makes it possible to decorrelate the observations during data acquisition. This opens the way to a sequential adjustment procedure, where the design matrix is treated row-by-row. Huge systems with millions of observations can be solved by direct or iterative strategies, and both approaches benefit from well-tailored filter techniques. Because of the sequential access the computational effort of this giant task can be easily distributed to a cluster of parallel processors and offers, in addition, the possibility to treat data gaps in a straightforward way. This revised version was published online in August 2006 with corrections to the Cover Date.     

The effect of the clutter-to-noise ratio on Doppler filterperformance

The effect of the clutter-to-noise ratio on the performance of a Doppler filter is considered. Clutter is assumed to have a power level which is unknown and varies in range. The assessment of the performance of a Doppler filter is based on the gain of the filter, which is the normalized output signal-to-interference ratio improvement at a given Doppler. The gain is generally a complex function of the statistics of the clutter. New upper and lower bounds on the gain differential between the expected design point clutter-to-noise ratio and the actual clutter-to-noise ratio are found. These bounds are independent of the clutter covariance matrix and are only a function of the unknown clutter-to-noise ratio. The bounds are valid for both Gaussian and non-Gaussian noise and for arbitrary linear filters. The upper and lower bounds differ by the theoretical coherent integration gain, 10 logN dB, where N is the number of pulses. A tighter lower bound is found for the case when the filters are matched filters. A simple exact expression is found for matched filters assuming a Gaussian Markov clutter model as the clutter spectral width approaches zero. An easily implementable adaptive procedure is given which improves performance due to the unknown clutter-to-noise ratio. This work extends a previous result, valid for the Emerson filter, that shows the effect of clutter-to-noise ratio on performance in terms of an average quantity, the improvement factor     

Clutter Covariance Matrix Form Effect on Adaptive MTI Performance

The performance of an adaptive moving target indicator (MTI), which employs a Wiener predictor by means of a transversal filter, is discussed, taking into consideration the effect of the form of the clutter covariance matrix on the MTI performance. It is emphasized that the main tap position in the transversal filter is an important factor which provides degrees of freedom in the clutter covariance matrix to improve the MTI performance. Calculation results show that by exploiting these degrees of freedom, excellent performance is feasible, in particular shorter transient response.     

Poly-phase codes and optimal filters for multiple user ranging

A technique is introduced to select poly-phase codes and optimal filters of a pulse compression system that have specific temporal and frequency characteristics. In the particular problem under study, multiple vehicles are assigned unique codes and receiver filters that have nearly orthogonal signatures. Narrowband users, that act as interference, are also present within the system. A code selection algorithm is used to select codes which have low autocorrelation sidelobes and low cross correlation peaks. Optimal mismatched filters are designed for these codes which minimize the peak values in the autocorrelation and the cross correlation functions. An adjustment to the filter design technique produces filters with nulls in their frequency response, in addition to having low correlation peaks. The method produces good codes and filters for a four-user system with length 34 four-phase codes. There is considerable improvement in cross and autocorrelation sidelobe levels over the matched filter case with only a slight decrease in the signal-to-noise ratio (SNR) of the system. The mismatched filter design also allows the design of frequency nulls at any frequency with arbitrary null attenuation, null width, and sidelobe level, at the cost of a slight decrease in processing gain     

DME Interference mitigation for L-DACS1 based on system identification and sparse representation

L-band digital aeronautical communication system 1 (L-DACS1) is a promising candi-date data-link for future air-ground communication, but it is severely interfered by the pulse pairs (PPs) generated by distance measure equipment. A novel PP mitigation approach is proposed in this paper. Firstly, a deformed PP detection (DPPD) method that combines a filter bank, correlation detection, and rescanning is proposed to detect the deformed PPs (DPPs) which are caused by mul-tiple filters in the receiver. Secondly, a finite impulse response (FIR) model is used to approximate the overall characteristic of filters, and then the waveform of DPP can be acquired by the original waveform of PP and the FIR model. Finally, sparse representation is used to estimate the position and amplitude of each DPP, and then reconstruct each DPP. The reconstructed DPPs will be sub-tracted from the contaminated signal to mitigate interference. Numerical experiments show that the bit error rate performance of our approach is about 5 dB better than that of recent works and is closer to interference-free environment.     

巴特沃斯数字滤波器在自动测试中的应用

本文主要研究巴特沃斯低通数字滤波器在自动测试系统中的应用。讨论了高速采样低速输出的自动测试系统中采用低通数字滤波器的必要性;比较了巴特沃斯低通数字滤波器与多点平均滤波的性能,巴特沃斯滤波器具有较理想的特性;介绍了建立巴特沃斯低通数字滤波器的设计方法,以及在实现实时滤波算法时,克服初始瞬变过程和防止滤波器数值不稳的措施。最后,给出了在长时间陀螺漂移测试中的应用结果。     

Optimal FIR filters for time-varying state-space models

An optimal FIR (finite impulse response) filter and smoother is introduced for the time-varying state-space model. The suggested filter has an FIR structure and utilizes finite observation. It is shown that the impulse response of the optimal FIR filter can be obtained by a simple Riccati-type matrix differential equation. Especially for time-invariant systems, this FIR filter reduces to previously known simple forms. For implementation, a recursive form of the optimal FIR filter and smoother is derived by using adjoint variables, and computational algorithms are suggested. It is also shown by sensitivity analysis that the proposed optimal FIR filter alleviates potential divergence characteristics of the standard Kalman filter     

基于多维数值积分的高阶确定采样型滤波方法

基于多维数值积分讨论了确定采样型滤波器,这一类滤波器的不同之处体现在对滤波器中均值和方差的计算,这一问题与数值积分密切相关.针对以往确定采样型滤波器在提高滤波精度的同时会增加计算量,通过分析高斯权值积分,采用完全对称积分公式,计算积分节点、节点个数及权重,相比高斯厄米特滤波减少了计算量,同时滤波精度可达到五阶以上.选取典型算例对新的滤波方法仿真分析,验证了该滤波方法的可行性和有效性.     

电动舵机中的电流环分析及校正

以电动舵机中电流控制问题为研究对象,通过对直流电机电流数学建模及频域分析,阐述了常规电流环控制器校正后的电流频率响应特性。针对电流负反馈容易引入测量噪声、采样延时带来不稳定等问题,提出呆用陷波滤波器串联校正电流的频率特性,并给出了滤波器参数设计方法。分别对串联校正和电流负反馈控制进行了实验验证,结果表明采用陷波滤波器串联校正能够抑制峰值电流,与电流环比例控制效果接近,其实现简单、稳定性好。     

Optimal Detection of a Signala with Time-Varying Carrier Phase

The problem considered in this paper is the detection of a signal known except for time-varying carrier phase in white Gaussian noise. The method of attacking this problem is to model the time-varying carrier phase as a Markov process. Fourier transform techniques are then applies to yield a simple time-wise adaptive form for the phasetracking detector. Optimal accounting for the time variations in phase is accomplished via a simple algorithm which serves to update the detector memory. Furthermore, it is shown that this memory updating operation is a discrete linear filter whose impulse response is a simple function of the previous memory state and the Markov transitional statistics on the phase. A priori knowledge regarding the phase is summarized in the initial impulse response of the updating filter.     

空间三轴激光陀螺高精度抖动剥除算法研究CSCD

激光陀螺的输出信号中包涵外界输入角速度、机械抖动角速度两部分信息,机械抖动角速度是一个叠加了一定噪声的标准正弦振动。针对空间三轴机抖激光陀螺仪,提出了一种高精度的新型正弦抖动信号滤除算法,通过自适应陷波器和有限冲击响应数字滤波器的组合,能极大地衰减激光陀螺仪零偏输出波形中的正弦分量,实现外界输入信号的高精度准确提取。实验结果表明,该抖动剥除算法效果显著,在保证快速响应外界输入的条件下能够实现高精度地提取角速度信号,可有效降低惯性系统的成本和复杂度,进而提高产品质量可靠性,具有很强的工程实用价值。