This paper proposes a new combined fast algorithm for transversal adaptive filters. The fast transversal filter (FTF) algorithm and the normalized LMS (NLMS) are combined in the following way. In the initialization period, the FTF is used to obtain fast convergence. After converging, the algorithm is switched to the NLMS algorithm because the FTF cannot be used for a long time due to its numerical instability. Nonstationary environment, that is, time varying unknown system for instance, is classified into three categories: slow time varying, fast time varying and sudden time varying systems. The NLMS algorithm is applied to the first situation. In the latter two cases, however, the NLMS algorithm cannot provide a good performance. So, the FTF algorithm is selected. Switching between the two algorithms is automatically controlled by using the difference of the MSE sequence. If the difference exceeds a threshold, then the FTF is selected. Other wise, the NLMS is selected. Compared with the RLS algorithm, the proposed combined algorithm needs less computation, while maintaining the same performance. Furthermore, compared with the FTF algorithm, it provides numerically stable operation.

We propose a third-order low-pass notch filter realized by a single operational amplifier and a minimum number of equal-valued capacitors. As a design example we realize a Chebyshev filter with a ripple of 0.5 dB and it is shown that the experiment result is very good.

In multidimensional signal sampling, the orthogonal sampling scheme is the simplest one and is employed in various applications, while a non-orthogonal sampling scheme is its alternative candidate. The latter sampling scheme is used mainly in application where the reduction of the sampling rate is important. In three-dimensional (3-D) signal processing, there are two typical sampling schemes which belong to the non-orthogonal samplings; one is face-centered cubic sampling (FCCS) and the other is body-centered cubic sampling (BCCS). This paper proposes a new design method for 3-D band-limiting FIR filters required for such non-orthogonal sampling schemes. The proposed method employs the McClellan transformation technique. Unlike the usual 3-D McClellan transformation, however, the proposed design method uses 2-D prototype filters and 2-D transformation filters to obtain 3-D FIR filters. First, 3-D general sampling theory is discussed and the two types of typical non-orthogonal sampling schemes, FCCS and BCCS, are explained. Then, the proposed design method of 3-D bandlimiting filters for these sampling schemes is explained and an effective implementation of the designed filters is discussed briefly. Finally, design examples are given and the proposed method is compared with other method to show the effectiveness of our methos.

An adaptive multichannel IIR lattice predictor for k-step ahead prediction is constructed and the effectiveness of the proposed predictor is evaluated using digital simulations.

This letter presents a technique to cancel the parasitic effects of operational amplifier (op amp) in active filter design. To minimize the effects, an op amp model considering the parasitics (i.e. both parasitic poles and zeros) is utilized. It is shown that undesirable factors in the transfer function due to the parasitics can be canceled well by predistorting the passive element values of the circuit. As an example, an active-R highpass filter is evaluated both theoretically and numerically. In this way, the proposed technique can be effectively incorporated into the design of active filters.

A least squares approximation method of recursive digital filters for finite interval response with zero value outside the interval is presented. According to the characteristic of the method, the modified Gauss Method is utilized in iteratively determining design parameters. Convergence, together with the stability of the resulting filter, are guaranteed.

This paper studies multidimensional linear periodically shift-variant digital filters (LPSV filters). The notion of a generalized multidimensional transfer function is presented for LPSV filters. The frequency characteristic of the filters is discussed in terms of this transfer function. Since LPSV filters can decompose the spectrum of an input signal into some spectral partitions and rearrange the spectrum, LPSV filters can serve as a frequency scrambler. To show the effect of multidimensional frequency scramble, 2-D LPSV filters are designed based on the 1-D Parks-McClellan algorithm. The resultant LPSV filters divide the input spectrum into some components that are permuted and possibly inverted with keeping the symmetric of the spectrum. Experimental results are presented to illustrate the effectiveness of frequency scramble for real images.

In this paper, a design method is described for very low sensitivity fully-balanced narrow-band band-pass switched-capacitor filters (SCF's) whose worst-case sensitivities of the amplitude responses become zero at every reflection zero. The proposed method is based on applying the low-pass to high-pass transformation, the pseudo two-path technique and the capacitance-ratio reduction technique to very low sensitivity low-pass SC ladder filters. A design example of the band-pass SCF with a quality factor Q250 is given to verify the proposed method. The remarkable advantages of this approach are very low sensitivity to element-value variations, a small capacitance spread, a small total capacitance, and clock-feedthrough noise immunity inside the passband.

An integrator is quite a suitable active element for high-speed filters. The effect of its excess phase shifts, however, is severe in the case of high-Q filter realization. The deterioration due to the excess phase shifts cannot be avoided when only integrators are used as frequency-dependent elements like in leapfrog realization. This paper describes a design of second-order high-speed and high-Q filters with low sensitivity to excess phase shifts of integrators by adding a passive RC circuit. The proposed method can drastically reduce the effect due to the undesirable pole of an integrator, which is the cause of the excess phase shifts, compared to conventional filters using only integrators. As an example, a fourth-order bandpass filter with 5-MHz center frequency and Q=25 is implemented by the proposed method on a monolithic chip. The results obtained here show quite good agreement with the theoretical values. This demonstrates effectiveness of the proposed method and feasibility of high-speed and high-Q filters on a monolithic chip.

The design of N-dimensional (N-D) FIR filters requires in general an enormous computational effort. One of the most successful methods for design and implementation is the McClellan transformation. In this paper a numerically simple technique for determining the coefficients of the transformation is suggested. This appears to be the simplest available method for the design of N-D hyperspherically symmetric FIR filters with excellent symmetry.

Inverse filters can be designed in order to enhance the accuracy with which signals recorded in a given space can be reproduced in a given listening space. The problem is considered here of the design of an inverse filter matrix which enables K recorded signals to be accurately reproduced at K points in the listening space when transmitted via M loudspeaker channels. The analysis is sufficiently general to incorporate the case when the best (least squares) approximation is sought to the reproduction of K signals at L points in the space when LK. An analysis is presented which demonstrates that the approach suggested by the Multiple-Input/Output Inverse Filtering theorem of Miyoshi and Kaneda can be realised adaptively by using the Multiple Error LMS algorithm of Elliott et al.

In case of the system identification problem, such as an echo canceller, estimated impulse response obtained by the frequency-domain adaptive filter based on the circular convolution has estimation error because the unknown system is based on the linear convolution in the time domain. In this correspondence, we consider a sufficient condition to reduce the estimation error.

This paper proposes a new design method of variable FIR digital filters. The method uses a multi-dimensional linearphase FIR filter as a prototype. The principle of the proposed method is based on the fact that the crosssectional characteristics of a 2-D filter along with a line vary if the intersection of this line is changed. The filter characteristics can be varied by recalculating all the filter coefficients from proposed equations, which leads to an advantage that the variable range is very wide. Another advantage is that the passband and stopband deviations are completely predetermined in the design procedures and that the passband edge can be accurately settled to a desired frequency while keeping the transition band width unchanged. First the proposed design method is explained and the effect of the transition band of 2-D filters is discussed. Then the calculation cost required in updating the filter coefficients are considered. Finally two design examples are presented and the proposed method is compared with the existing one, which shows the usefulness of our method.

This paper designs and evaluates highly parallel VLSI processors for real time 2-D state-space digital filters using hierarchical behavioral description language and synthesizer. The architecture of the 2-D state-space digital filtering system is a linear systolic array of homogeneous VLSI processors, each of which consists of eight processing elements (PEs) executing 1-D state-space digital filtering with multi-input and multi-output. Hierarchical behavioral description language and synthesizer are adopted to design and evaluate PE's and the VLSI processors. One 16 bit fixed-point PE executing a (4, 4)-th order 2-D state-space digital filtering is described on the basis of distributed arithmetic in about 1,200 steps by the description language and is composed of 15 K gates in terms of 2 input NAND gate. One VLSI processor which is a cascade connection of eight PEs is composed of 129 K gates and can be integrated into one 1515 [mm2] VLSI chip using 1 µm CMOS standard cell. The 2-D state-space digital filtering system composed of 128 VLSI processors at 25 MHz clock can execute a 1,0241,024 image in 1.47 [msec] and thus can be applied to real-time conventional video signal processing.

This paper proposes a designing algorithm for quadrilateral recursive filters which consist of four quarter-plane filters in the four quadrants. This can realize a perfect zero-phase filtering which is essential for image processing. Furthermore, several parallel processing algorithms capable of performing under very high parallel efficiency are developed on line-connected and mesh-connected processor arrays. By these proposals, the advantage of two-dimensional non-causal zero-phase recursive digital filters is made clear.

A new MOS linear operational transconductance amplifier (OTA) for the up-to-4 MHz range OTA-C filters is proposed. The proposed OTA is designed using a new linearizing technique based on bias-current modulation, to compensate nonlinearities in the transfer characteristic of the conventional current-source-biased source-coupled pair. The design and SPICE simulation are presented in detail, assuming the implementation by the typical p-well CMOS process. The simulation of a 3.58 MHz OTA-C band-pass filter built with the proposed OTAs showed close agreement with the desired performance.