Yong Ching Lim

Bio: Yong Ching Lim is an academic researcher from Nanyang Technological University. The author has contributed to research in topics: Digital filter & Finite impulse response. The author has an hindex of 30, co-authored 145 publications receiving 3205 citations. Previous affiliations of Yong Ching Lim include Tampere University of Technology & National University of Singapore.

A weighted least squares algorithm for quasi-equiripple FIR and IIR digital filter design

Abstract: It has been demonstrated by several authors that if a suitable frequency response weighting function is used in the design of a finite impulse response (FIR) filter, the weighted least squares solution is equiripple. The crux of the problem lies in the determination of the necessary least squares frequency response weighting function. A novel iterative algorithm for deriving the least squares frequency response weighting function which will produce a quasi-equiripple design is presented. The algorithm converges very rapidly. It typically produces a design which is only about 1 dB away from the minimax optimum solution in two iterations and converges to within 0.1 dB in six iterations. Convergence speed is independent of the order of the filter. It can be used to design filters with arbitrarily prescribed phase and amplitude response. >

The optimum design of one- and two-dimensional FIR filters using the frequency response masking technique

Abstract: An expression for the impulse response up-sampling ratio M, which will produce a minimum complexity design, is derived. It is shown that M approaches e (the base of the natural logarithm) as the number of frequency response masking stages increases; in a K-stage design the complexity of the filter is inversely proportional to the (K+1)th root of the transition width; the frequency response masking technique is effective if the normalized transition width is less than 1/16; and the frequency response masking technique is more efficient than the interpolated impulse response technique if the square root of the normalized transition width is less than the arithmetic mean of the normalized passband edge and stopband edge. An expression for the multistage frequency response ripple compensation is derived. An optimum design relationship for the interpolated impulse response technique is also derived. The design of narrow-band two-dimensional filters using the frequency response masking technique is also presented. >

Design of discrete-coefficient-value linear phase FIR filters with optimum normalized peak ripple magnitude

Abstract: Four methods are presented for optimizing filters in the normalized peak ripple magnitude (NPRM) sense. Two of the methods being to the passband gain sectioning technique. The other two methods make use of the objective function f= delta - alpha b. Several heuristic methods for determining alpha are also presented. The NPRM is an important performance measure; the absolute peak ripple magnitude and passband gain are less important. In these applications, the passband gain need not be fixed at unity but should be a continuous variable to the optimized. Nevertheless, an upper and a lower bound on the passband gain should be imposed to satisfy overflow and roundoff noise performance requirements. >

Signed power-of-two term allocation scheme for the design of digital filters

Abstract: It is well known that if each coefficient value of a digital filter is a sum of signed power-of-two (SPT) terms, the filter can be implemented without using multipliers. In the past decade, several methods have been developed for the design of filters whose coefficient values are sums of SPT terms. Most of these methods are for the design of filters where all the coefficient values have the same number of SPT terms. It has also been demonstrated recently that significant advantage can be achieved if the coefficient values are allocated with different number of SPT terms while keeping the total number of SPT terms for the filter fixed. In this paper, we present a new method for allocating the number of SPT terms to each coefficient value. In our method, the number of SPT terms allocated to a coefficient is determined by the statistical quantization step-size of that coefficient and the sensitivity of the frequency response of the filter to that coefficient. After the assignment of the SPT terms, an integer-programming algorithm is used to optimize the coefficient values. Our technique yields excellent results but does not guarantee optimum assignment of SPT terms. Nevertheless, for any particular assignment of SPT terms, the result obtained is optimum with respect to that assignment.

Single-precision multiplier with reduced circuit complexity for signal processing applications

Abstract: When two numbers are multiplied, a double-wordlength product is produced. In applications where only the single-precision product is required, the double-wordlength result is rounded to single-precision. Hence, in single-precision applications, it is not necessary to compute the least significant part of the product exactly. Instead, it is only necessary to estimate the carries generated in the computation of the least significant part that will ripple into the most significant part of the product. This will produce a single-precision multiplier with significantly reduced circuit complexity. Three novel methods for realizing this class of reduced complexity single-precision multipliers are introduced and their performance analyzed. >

Multirate Systems and Filter Banks

Abstract: The outline of this chapter is as follows. Section 2 reviews various types of existing finite impulse response (FIR) and infinite impulse response (IIR) two-channel filter banks. The basic operations of these filter banks are considered and the requirements are stated for alias-free, perfect-reconstruction (PR), and nearly perfect-reconstruction (NPR) filter banks. Also some efficient synthesis techniques are referred to. Furthermore, examples are included to compare various two-channel filter banks with each other. Section 3 concentrates on the design of multi-channel (M-channel) uniform filter banks. The main emphasis is laid on designing these banks using tree-structured filter banks with the aid of two-channel filter banks and on generating the overall bank with the aid of a single prototype filter and a proper cosine-modulation or MDFT technique. In Section 4, it is shown how octave filter banks can be generated using a single two-channel filter bank as the basic building block. Also, the relations between the frequency-selective octave filter banks and discrete-time wavelet banks are briefly discussed. Finally, concluding remarks are given in Section 5.

Understanding digital signal processing

Abstract: From the Publisher: This is undoubtedly the most accessible book on digital signal processing (DSP) available to the beginner. Using intuitive explanations and well-chosen examples, this book gives you the tools to develop a fundamental understanding of DSP theory. The author covers the essential mathematics by explaining the meaning and significance of the key DSP equations. Comprehensive in scope, and gentle in approach, the book will help you achieve a thorough grasp of the basics and move gradually to more sophisticated DSP concepts and applications.

Splitting the unit delay [FIR/all pass filters design]

Abstract: A fractional delay filter is a device for bandlimited interpolation between samples. It finds applications in numerous fields of signal processing, including communications, array processing, speech processing, and music technology. We present a comprehensive review of FIR and allpass filter design techniques for bandlimited approximation of a fractional digital delay. Emphasis is on simple and efficient methods that are well suited for fast coefficient update or continuous control of the delay value. Various new approaches are proposed and several examples are provided to illustrate the performance of the methods. We also discuss the implementation complexity of the algorithms. We focus on four applications where fractional delay filters are needed: synchronization of digital modems, incommensurate sampling rate conversion, high-resolution pitch prediction, and sound synthesis of musical instruments.

Multirate digital signal processing

Abstract: There has been a great deal of material in the area of discrete-time transforms that has been published in recent years. This book does an excellent job of presenting important aspects of such material in a clear manner. The book has 11 chapters and a very useful appendix. Seven of these chapters are essentially devoted to the Fourier series/transform, discrete Fourier transform, fast Fourier transform (FFT), and applications of the FFT in the area of spectral estimation. Chapters 8 through 10 deal with many other discrete-time transforms and algorithms to compute them. Of these transforms, the KarhunenLoeve, the discrete cosine, and the Walsh-Hadamard transform are perhaps the most well-known. A lucid discussion of number theoretic transforms i5 presented in Chapter 11. This reviewer feels that the authors have done a fine job of compiling the pertinent material and presenting it in a concise and clear manner. There are a number of problems at the end of each chapter, an appreciable number of which are challenging. The authors have included a comprehensive set of references at the end of the book. In brief, this book is a valuable contribution to the general areas of signal processing and communications. It can be used for a graduate level course in perhaps two ways. One would be to cover the first seven chapters in great detail. The other would be to cover the whole book by focussing on different topics in a selective manner. This book by Elliott and Rao is extremely useful to researchers/engineers who are working in the areas of signal processing and communications. It i s also an excellent reference book, and hence a valuable addition to one’s library

The sliding DFT

Abstract: The sliding DFT process for spectrum analysis was presented and shown to be more efficient than the popular Goertzel (1958) algorithm for sample-by-sample DFT bin computations. The sliding DFT provides computational advantages over the traditional DFT or FFT for many applications requiring successive output calculations, especially when only a subset of the DFT output bins are required. Methods for output stabilization as well as time-domain data windowing by means of frequency-domain convolution were also discussed. A modified sliding DFT algorithm, called the sliding Goertzel DFT, was proposed to further reduce the computational workload. We start our sliding DFT discussion by providing a review of the Goertzel algorithm and use its behavior as a yardstick to evaluate the performance of the sliding DFT technique. We examine stability issues regarding the sliding DFT implementation as well as review the process of frequency-domain convolution to accomplish time-domain windowing. Finally, a modified sliding DFT structure is proposed that provides improved computational efficiency.