Analysis of the frequency domain adaptive filter

The purpose of this note is to demonstrate significant analytical simplifications for studying the behavior of adaptive filtering in the frequency domain as opposed to studying the behavior of adaptive filtering in the time domain. A closed form expression, for the single complex weight in the frequency domain adaptive filter, is presented which allows significant statistical analysis to be performed. The mean-square error of the filter is evaluated as a function of the algorithm step size and the signal and noise powers.     

Low-distortion switched-capacitor filter design techniques

The distortion mechanism in switched-capacitor (SC) filters are considered, and closed-form expression relating switched-capacitors filter distortion to circuit parameters are derived. Design techniques for low-distortion applications are discussed and are applied to a sixth-order experimental filter. The filter design uses a fully differential class A/B op amp with a continuous-time common-mode feedback circuit. Distortion measurements show that for 82-dB dynamic range (relative to noise floor) the total harmonic distortion of 0.02% within the whole 4-kHz bandwidth and 0.07% within 20-kHz bandwidth.     

A method of digital filter synthesis in the frequency domain

In this letter, a new method of digital filter synthesis is presented. The transfer function of the resulting digital filter closely approximates the corresponding continuous transfer function of the correcting filter with steep transition regions.     

Digital filter bank design quadratic-constrained formulation

Formulate the filter bank design problem as an quadratic-constrained least-squares minimization problem. The solution of the minimization problem converges very quickly since the cost function as well as the constraints are quadratic functions with respect to the unknown parameters. The formulations of the perfect-reconstruction cosine-modulated filter bank, of the near-perfect-reconstruction pseudo-QMF bank, and of the two-channel biorthogonal linear-phase filter bank are derived using the proposed approach. Compared with other design methods, the proposed technique yields PR filter banks with much higher stopband attenuation. The proposed technique can also be extended to design multidimensional filter banks     

基于频域滤波数字均衡器的设计

为了解决多频段数字均衡滤波器处理过程中数据计算量的问题,通过对数字均衡器设计的分析,将数字音频信号进行频域滤波处理,最终设计出一种高效的数字均衡滤波器。通过将数字信号在频域中进行傅里叶变换,提出了一种基于快速傅里叶变换原理的算法,该算法中码位倒置和蝶形运算方法的处理与通常的快速傅里叶变换相比,更有效地减少了数据的运算量,减少了数据处理的时间。结果表明,使用该种算法设计的数字均衡滤波器与传统的时域滤波方法相比,具有很好的实时处理效果。     

RS码的时域编码/频域译码技术

提出一种RS混合译码技术－时域编码／频域译码，对照单纯的时域编译 码和频域编译码分析了其优越性。这种编译码技术可广泛应用于数字电视信号的传输及其它数字通信场合。     

FIR design by non-uniform sampling in frequency domain

In this paper an expression for H(z) has been derived in terms of N independent samples {H_k }. It has then been used to examine effects of non-uniform frequency sampling on the resultant frequency response of FIR filters. This has been applied first to the non-uniform sampling in both the passband and stopband, of a desired filter, and then to non-uniform sampling in the stopband only. Several lowpass filter designs have been examined by these methods and results discussed.     

基带成形滤波器的数字设计与实现

根据基带成型滤波器的工作原理,文中设计出了一种基带成型滤波器的数字实现方案。该方案首先运用MATALB仿真工具得到信号基带成型后的仿真数据,并将仿真数据存储在FPGA中,然后通过查表操作实现了数字基带成型滤波器的功能。文中还给出了通过MODELSIM得到的信号基带成型后的仿真结果,仿真结果表明,由该方案所设计的基带成型滤波器可以很好地完成通信系统中信号的成型特性。     

A multistep size (MSS) frequency domain adaptive filter

A multistep size frequency-domain adaptive filter capable of tracking both stationary and nonstationary signals is proposed. This algorithm makes use of the simple structure of the least mean square error algorithm to update the filter coefficients and process the signal. It then proceeds further by incorporating a set of three step sizes and a knowledge-based strategy to select the best set of filter coefficients and the optimum step size iteratively. The main advantage of the MSS algorithm over the conventional LMS algorithm is that better performance is achieved without the knowledge of the input signal characteristics, such as signal power, degree of nonstationarity, signal-to-noise ratio, and stability bounds. Experimentally, the MSS algorithm is tested under various signal environments. The transient characteristics of the step sizes are found to be in agreement with previous theoretical studies of nonstationary characteristics of the LMS algorithm. In order to reduce the complexity, the conventional frequency-domain block LMS structure is also modified so that the MSS algorithm can be embedded more efficiently by exploiting the block structure     

多相码信号数字脉压滤波器设计

对较小时宽带宽积相位编码信号的时域副瓣抑制滤波器设计问题进行了研究。在约束主瓣输出为常数时使积分副瓣电平最小化，利用最小二乘算法和加权迭代方法，适当放宽主瓣宽度，以使主副比达到最优。仿真结果表明，该方法的设计结果在一定多普勒频率范围内，与匹配滤波相比，能达到10dB以上的主副比改善，且主瓣宽度展宽不大，信噪比损失小于0．5dB。     

数字可调中频噪声源设计

介绍了用FPGA和AD9739产生可调中频噪声源的方法.采用Combined Tausworthe方法产生均匀分布,根据均匀分布和高斯分布之间的映射关系,使用折线逼近的方法产生基带的高斯分布.采用可配置HR IP核实现中心频率和带宽可调的中频噪声.FPCA产生的中频噪声源通过LVDS接口输出给AD9739.     

Iterative design and detection of a DFE in the frequency domain

Error-propagation phenomena and computational complexity of the filters' design are important drawbacks of existing decision-feedback equalizers (DFE) for dispersive channels. In this paper, we propose a new iterative block DFE (IBDFE) which operates iteratively on blocks of the received signal. Indeed, a suitable data-transmission format must be used to allow an efficient implementation of the equalizer in the frequency domain, by means of the discrete Fourier transform. Two design methods are considered. In the first method, hard detected data are used as input of the feedback, and filters are designed according to the correlation between detected and transmitted data. In the second method, the feedback signal is directly designed from soft detection of the equalized signal at the previous iteration. Estimators of the parameters involved in the IBDFE design are also derived. From performance simulations on a wireless dispersive fading channel, we observed that the IBDFE outperforms existing DFEs. Moreover, the IBDFE exhibits a reduction of the computational complexity when compared against existing schemes, both in signal processing and in filter design.     

离焦模糊数字图像的Wiener滤波频域复原

离焦模糊图像的退化模型可用均匀分布的圆盘函数表示 ,其对应的圆盘半径是需辨识的退化模型参数 (即模糊半径 )。利用模糊图像的频率域中的零点位置来估计模糊半径 ,采用简化Wiener滤波对模糊图像进行复原。实验结果表明该方法能够以较少的运算时间代价获取较好的复原效果 ,适用于信噪比高的离焦模糊图像的快速复原     

FIR digital filter design techniques using weighted Chebyshev approximation

This paper discusses the various approaches to designing FIR digital filters using the theory of weighted Chebyshev approximation. The different design techniques are explained and compared on the basis of their capabilities and limitations. The relationships between filter parameters are briefly discussed for the case of low-pass filters. Extensions of the theory to the problems of magnitude and complex approximation are also included, as are some recent results on the design of two-dimensional FIR filters by transformation.     

Secondary LC filter analysis and design techniques forcurrent-mode-controlled converters

Small-signal characteristics of current-mode-controlled PWM converters with a second-stage LC filter are analyzed. It is shown that a secondary filter can be designed to provide good attenuation of the switching ripple while maintaining adequate stability margins with capacitive loading. The resonant frequencies and damping coefficients of the second filter are derived, and design guidelines are given. It is shown that the current-loop gain of the buck converter is not affected by the addition of the second-stage filters when a small filter inductance is used. Three design examples are presented to demonstrate the use of analysis results. Two filter examples are designed for a buck converter. One of the second filters shows the problems that arise with a poor design. A third example is the design of a second-stage filter for a buck-boost converter. In each of the design examples, the small-signal analysis was performed using EASY5 software and the circuits were simulated using the state-space simulation program COSMIR     

基于频域递推算法的窄带滤光片光束传输特性研究

多腔窄带薄膜滤光片在倾斜入射时中心透射波长会向短波方向移动,由此可制备角度调谐滤波器。但其透射光斑在倾斜入射时会出现明显的斑展宽现象,由此产生的接收端耦合度及干涉级次的降低会导致其透射光谱的插入损耗迅速增大,会限制其有效波长调谐范围。传统的多光束叠加法可以对窄带滤光片斜入射时的透射光斑展宽现象进行仿真,但是在光强分布上与实验结果有着较大的误差。本文基于快速傅里叶变换及菲涅尔公式,提出了一种新颖的频域递推算法来计算光斑的展宽程度及透射光强分布,计算了从0度到60度倾斜入射范围内光斑展宽以及透射率的变化情况。理论计算和实验结果都表明该方法较之传统的多光束叠加算法结果更准确且计算效率高。     

同时实现频率滤波和偏振分束的结构

通过设计由介质薄膜组成的二周期结构,在特定的入射条件下,利用特征矩阵的方法计算了TE光和TM光的透射率,发现该结构对TE光和TM光存在一系列分离的频率通道。这些分离的频率通道使该结构既可以用作频率滤波器,又可以用作偏振分束器。     

频率特性有间断点的滤波器的全相位设计

本文提出了频率特性有间断点的滤波器伞相位设计方法，给出全相位数字滤波器与原FIR滤波器的结构，推导出了全相位数字滤波器的内插函数，并用实验证明全相位陷波器具有良好的陷波频率特性。     

FIR滤波器设计:基于进化规划的频率采样技术

本文介绍了进化规划在要样技术中的应用,结合ＦＩＲ数字低通、带通滤波器设计的两个例子,给出了算法实现的具体操作步骤和实验结果。实验数据表明采用进化规划确定的频率过渡带样本值是最优的,设计的ＦＩＲ滤波器的性能优于查表法。