ScholarBank@NUShttps://scholarbank.nus.edu.sgThe DSpace digital repository system captures, stores, indexes, preserves, and distributes digital research material.Thu, 16 Jul 2020 17:15:44 GMT2020-07-16T17:15:44Z50101- Finite-time-domain synthesis of linear time-variant digital filters by difference equationshttps://scholarbank.nus.edu.sg/handle/10635/62201Title: Finite-time-domain synthesis of linear time-variant digital filters by difference equations
Authors: Li, D.
Abstract: This paper presents a method for synthesizing a recursive linear time-variant digital filter in finite time domain. The desired filter can be nonrecursive and is given by its generalized frequency function. The synthesized filter is described by a linear difference equation with time-variant coefficients. These coefficients at a given time instant are derived recursively from those at the previous instants. In addition, for the determined coefficients at the previous instants, the coefficients at the given instant minimize the error between the generalized frequency function of the desired filter and that of the synthesized one at the given instant. The performance of this method is illustrated and compared with another available method through numerical examples. The results show an improvement of nearly two order of magnitude in the normalized mean squared error between the amplitudes of the generalized frequency function of the desired filter and that of the synthesized. © 1994.
Thu, 01 Sep 1994 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/622011994-09-01T00:00:00Z
- Signed power-of-two (SPT) term allocation scheme for the design of digital filtershttps://scholarbank.nus.edu.sg/handle/10635/81739Title: Signed power-of-two (SPT) term allocation scheme for the design of digital filters
Authors: Lim, Yong-Ching; Yang, Rui; Li, Dongning; Song, Jianjian
Abstract: It is well known that if each coefficient value of a digital filter is a sum of SPT terms, the filter can be implemented without using multipliers. In the past decade, several methods had 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. In this paper, we present a new method for allocating different number of SPT terms to each coefficient value keeping the total number of SPT terms fixed. Our technique yields excellent results.
Thu, 01 Jan 1998 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/817391998-01-01T00:00:00Z
- Multiplierless realization of adaptive filters by nonuniform quantization of input signalhttps://scholarbank.nus.edu.sg/handle/10635/81559Title: Multiplierless realization of adaptive filters by nonuniform quantization of input signal
Authors: Li, Dongning; Ching Lim, Yong
Abstract: This paper presents a multiplierless realization structure for the implementation of adaptive digital filters. Multiplierless realization is achieved by quantizing input signals in such way that each input sample is expressed as a sum of a fixed number of signed power-of-two (SPT) terms. This simplifies all multiplications involving input signal data. Simulation results for suppressing a narrow-band interference in a direct-sequence spread-spectrum communication system show that the performance of our technique with input signal quantized to 2 SPT terms is comparable to a conventional filter with input uniformly quantized to 10 bits. However, our technique reduces the number of transistors required for realization in a custom VLSI circuit to about 1/3 of that required by conventional realization.
Sat, 01 Jan 1994 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/815591994-01-01T00:00:00Z
- Polynomial-time algorithm for designing digital filters with power-of-two coefficientshttps://scholarbank.nus.edu.sg/handle/10635/81678Title: Polynomial-time algorithm for designing digital filters with power-of-two coefficients
Authors: Li, Dongning; Song, Jianjian; Lim, Yong Ching
Abstract: This paper presents an algorithm for designing digital filters with coefficients expressible as sums of signed power-of-two (SPT) terms. For each filter gain, the time complexity of the algorithm is a second-order polynomial in the filter order and is a first-order polynomial in the filter wordlength. Unlike conventional methods where each coefficient is allocated a fixed number of SPT terms, our method allows the number of SPT terms for each coefficient to vary subject to the number of SPT terms for the entire filter. This provides us with the possibility of finding a better filter without increasing the number of adders, which determines the realization cost for a given filter length. Application of the algorithm to FIR filter designs shows that it achieves up to 8.9 dB improvement over simulated annealing and mixed integer linear programming on the normalized peak ripples of example filters.
Fri, 01 Jan 1993 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/816781993-01-01T00:00:00Z
- Finite-time-domain synthesis of linear time-variant digital filters by difference equationshttps://scholarbank.nus.edu.sg/handle/10635/80433Title: Finite-time-domain synthesis of linear time-variant digital filters by difference equations
Authors: Li, D.
Abstract: This paper presents a method for synthesizing a recursive linear time-variant digital filter in finite time domain. The desired filter can be nonrecursive and is given by its generalized frequency function. The synthesized filter is described by a linear difference equation with time-variant coefficients. These coefficients at a given time instant are derived recursively from those at the previous instants. In addition, for the determined coefficients at the previous instants, the coefficients at the given instant minimize the error between the generalized frequency function of the desired filter and that of the synthesized one at the given instant. The performance of this method is illustrated and compared with another available method through numerical examples. The results show an improvement of nearly two order of magnitude in the normalized mean squared error between the amplitudes of the generalized frequency function of the desired filter and that of the synthesized. © 1994.
Thu, 01 Sep 1994 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/804331994-09-01T00:00:00Z
- Signed power-of-two (SPT) term allocation scheme for the design of digital filtershttps://scholarbank.nus.edu.sg/handle/10635/50640Title: Signed power-of-two (SPT) term allocation scheme for the design of digital filters
Authors: Lim, Yong-Ching; Yang, Rui; Li, Dongning; Song, Jianjian
Abstract: It is well known that if each coefficient value of a digital filter is a sum of SPT terms, the filter can be implemented without using multipliers. In the past decade, several methods had 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. In this paper, we present a new method for allocating different number of SPT terms to each coefficient value keeping the total number of SPT terms fixed. Our technique yields excellent results.
Thu, 01 Jan 1998 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/506401998-01-01T00:00:00Z
- Polynomial-time algorithm for designing digital filters with power-of-two coefficientshttps://scholarbank.nus.edu.sg/handle/10635/72865Title: Polynomial-time algorithm for designing digital filters with power-of-two coefficients
Authors: Li, Dongning; Song, Jianjian; Lim, Yong Ching
Abstract: This paper presents an algorithm for designing digital filters with coefficients expressible as sums of signed power-of-two (SPT) terms. For each filter gain, the time complexity of the algorithm is a second-order polynomial in the filter order and is a first-order polynomial in the filter wordlength. Unlike conventional methods where each coefficient is allocated a fixed number of SPT terms, our method allows the number of SPT terms for each coefficient to vary subject to the number of SPT terms for the entire filter. This provides us with the possibility of finding a better filter without increasing the number of adders, which determines the realization cost for a given filter length. Application of the algorithm to FIR filter designs shows that it achieves up to 8.9 dB improvement over simulated annealing and mixed integer linear programming on the normalized peak ripples of example filters.
Fri, 01 Jan 1993 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/728651993-01-01T00:00:00Z
- Multiplierless realization of adaptive filters by nonuniform quantization of input signalhttps://scholarbank.nus.edu.sg/handle/10635/72764Title: Multiplierless realization of adaptive filters by nonuniform quantization of input signal
Authors: Li, Dongning; Ching Lim, Yong
Abstract: This paper presents a multiplierless realization structure for the implementation of adaptive digital filters. Multiplierless realization is achieved by quantizing input signals in such way that each input sample is expressed as a sum of a fixed number of signed power-of-two (SPT) terms. This simplifies all multiplications involving input signal data. Simulation results for suppressing a narrow-band interference in a direct-sequence spread-spectrum communication system show that the performance of our technique with input signal quantized to 2 SPT terms is comparable to a conventional filter with input uniformly quantized to 10 bits. However, our technique reduces the number of transistors required for realization in a custom VLSI circuit to about 1/3 of that required by conventional realization.
Sat, 01 Jan 1994 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/727641994-01-01T00:00:00Z
- Finite-time-domain synthesis of recursive linear time-variant causal digital filters by separable sequenceshttps://scholarbank.nus.edu.sg/handle/10635/72638Title: Finite-time-domain synthesis of recursive linear time-variant causal digital filters by separable sequences
Authors: Li, Dongning
Abstract: Linear time-variant (LTV) digital filters whose impulse responses are separable sequences have been proved to be recursive. Such filters have the potential in saving computation time and storage. This paper presents a method to synthesize a desired LTV digital filter given in numerical form in a finite time domain by a separable sequence of a given order. The method finds the separate sequence by alternately using two nonlinear restrictions that an optimal recursive LTV filter approximating the desired filter must satisfy. The performance of the proposed method is illustrated and compared with other available ones through numerical examples. The results of the comparisons show that our method provides smaller synthesis errors than the others.
Fri, 01 Jan 1993 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/726381993-01-01T00:00:00Z
- Finite-time-domain synthesis of recursive linear time-variant causal digital filters by separable sequenceshttps://scholarbank.nus.edu.sg/handle/10635/62202Title: Finite-time-domain synthesis of recursive linear time-variant causal digital filters by separable sequences
Authors: Li, Dongning
Abstract: A linear time-variant (LTV) digital filter having a separable sequence as its impulse response has been proved to be recursive. Such filters have the potential of saving computation time and storage. Two techniques are presented for synthesizing a desired LTV digital filter given in numerical form in a finite time domain by a separable sequence. The first is a realization technique which decomposes the impulse response of a desired filter into a separable sequence and theoretically leads to the solution of a separable sequence with the lowest order. For a numerical solution, the order of the separable sequence is dependent on an error tolerance that reflects the realization accuracy. If the desired filter is recursive, then the exact order can be solved. The solved order, as well as the decomposition error, is independent of the error tolerance in a wide range. The second is an approximation technique which finds a separable sequence of a given order by minimizing the normalized mean squared error between the impulse response of the desired filter and the separable sequence. The technique searches the separable sequence by alternately using two nonlinear restrictions that an optimally approximated filter must satisfy. The performances of the proposed techniques are illustrated and compared with other available techniques through numerical examples. The results of the comparisons show that our approximation technique results in smaller approximation errors than those of the others.
Mon, 01 Nov 1993 00:00:00 GMThttps://scholarbank.nus.edu.sg/handle/10635/622021993-11-01T00:00:00Z