Fast recursive estimation using the lattice structure

E. Shichor
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引用次数: 11

Abstract

This paper presents the theory for a rapidly converging adaptive linear digital filter. The filter weights are updated for every new input sample. This way the filter is optimal (in the minimum mean square error sense) for all past data up to the present, at all instants of time. This adaptive filter has thus the fastest possible rate of convergence. Such an adaptive filter, which is highly desirable for use in dynamical systems, e.g., digital equalizers, used to require on the order of N2 multiplications for an N-tap filter at each instant of time. Recent “fast” algorithms have reduced this number to like 10 N. One of these algorithms has the lattice form, and is shown here to have some interesting properties: It decorrelates the input data to a new set of orthogonal components using an adaptive, Gram-Schmidt like, transformation. Unlike other fast algorithms of the Kalman form, the filter length can be changed at any time with no need to restart or modify previous results. It is conjectured that these properties will make it less sensitive to digital quantization errors in finite word-length implementation.
使用点阵结构的快速递归估计
提出了一种快速收敛的自适应线性数字滤波器的原理。每个新的输入样本都会更新过滤器权重。这样,对于所有过去到现在的数据,在所有时刻,过滤器都是最优的(在最小均方误差意义上)。因此,这种自适应滤波器具有最快的收敛速度。这种自适应滤波器非常适合用于动态系统,例如数字均衡器,用于在每个瞬间对n个抽头滤波器进行N2次乘法。最近的“快速”算法已经将这个数字减少到10n,其中一种算法具有晶格形式,并且在这里展示了一些有趣的特性:它使用自适应的Gram-Schmidt - like变换将输入数据解关联到一组新的正交分量。与其他卡尔曼形式的快速算法不同,滤波器长度可以随时改变,无需重新启动或修改先前的结果。据推测,这些特性将使它在有限字长实现中对数字量化误差不那么敏感。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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