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.

A Chebyshev type bandpass filter using four TM01δ-mode dielectric rod resonators oriented axially in a high-Tc superconductor cylinder is designed with 3 dB bandwidth 36 MHz at 11.958 GHz. The single resonator which contains a Ba (MgTa) O3 ceramic rod of εγ=24 and a YBa2Cu3Oy bulk cylinder is designed to realize temperature coefficient of f0, τf=0 ppm/K at 20 K. The unloaded Q, Qu measured at 20 K is 150,000 which is higher than Qu=100,000 for a TM01δ-mode resonator with a copper cylinder. When the constructed filter is cooled from room temperature to below 50 K, the center frequency shifted only 5 MHz which corresponds to τf=1.5 ppm/K and the insertion loss IL0 at the center freqency reduced from 3.0 dB to about 0 dB, the designed value of which is 0.04 dB, which is too small to be measured accurately.

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.

It often occurs in an environmental phenomenon in our daily life that a specific signal is partially or completely contaminated by the additional external noise. In this study, a digital filter for estimating a specific signal fluctuating impulsively under the existence of an actual external noise with various kinds of probability distribution forms is proposed in an improved form of already reported digital filter. The effectivenss of the proposed theory is experimentally confirmed by applying it to the estimation of an actual impulsve signal in a room acoustic.

This paper treats the problem of realizing high speed 2-D denominator separable digital filters. Partitioning a 2-D data plane into square blocks, filtering proceeds block by block sequentially. A fast intra-block parallel processing method was developed using block state space realization, which allows simultaneous computation of all the next block states and the outputs of one block. As the block state matrix of the filter has high sparsity, the rows and columns are interchanged respectively to reduce the matrix size. The filter is implemented by a multiprocessor system, where for each matrix's row one processor is assigned to perform the row-column vector multiplication. All processors wirk in synchronized fashion. Number of processors of this implementation are equal to the number of rows of the reduced state matrix and throughput is raised with block lengths.

This paper proposes an efficient design method of three-dimensional (3-D) recursive digital filters for video signal processing via decomposition of magnitude specifications. A given magnitude specification of a 3-D digital filter is decomposed into specifications of 1-D digital filters with three different (horizontal, vertical, and temporal) directions. This decomposition can reduce design problems of 3-D digital filters to design problems of 1-D digital filters, which can be designed with ease by conventional methods. Consequently, design of 3-D digital filters can be efficiently performed without complicated tests for stability and large amount of computations. In order to process video signal in real time, the 1-D digital filters with temporal direction must be causal, which is not the case in horizontal and vertical directions. Since the proposed method can approximate negative magnitude specifications obtained by the decomposition with causal 1-D R filters, the 1-D digital filters with temporal direction can be causal. Therefore the 3-D digital filters designed by the proposed method is suitable for real time video signal processing. The designed 3-D digital filters have a parallel separable structure having high parallelism, regularity and modularity, and thus is suitable for high-speed VLSI implementation.

One of the problems with subband image coding is the increase in image sizes caused by filtering. To solve this, it has been proposed to process the filtering by transforming input sequence into a periodic one. Then filtering is implemented by circular convolution. Although this technique solves the problem, there are very strong restrictions, i.e., limitation on the filter type and on the filter bank structure. In this paper, development of this technique is presented. Consequently, any type of linear phase FIR filter and any structure of filter bank can be used.

Based on the Fornasini-Marchesini second model, an efficient algorithm is developed to derive the characteristic polynomial and the inverse of the system matrix from the state-space parameters. As a result, the external description of the Fornasini-Marchesini second model is clarified. A technique for designing 2-D recursive digital filters in the frequency domain is then presented by using the Fornasini-Marchesini second model. The resulting filter approximates both magnitude and group delay specifications and its stability is always guaranteed. Finally, three design examples are given to illustrate the utility of the proposed technique.

Based on the Fornasini-Marchesini second local state-space (LSS) model, the coefficient sensitivities of two-dimensional (2-D) digital filters are analyzed in conjunction with frequency weighting functions. The overall sensitivity called the frequency-weighting sensitivity is then evaluated using the 2-D generalized Gramians that are newly introduced for the Fornasini-Marchesini second LSS model. Next, the 2-D filter structures that minimize the frequency-weighting sensitivity are synthesized for two cases of no constraint and scaling constraints on the state variables. Finally, an example is given to illustrate the utility of the proposed technique.

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.

The optimal coding strategy for signal detection in the correlated gaussian noise is established for the distributed sensors system with essentially zero transmission rate constraint. Specifically, we are able to obtain the same performance as in the situation of no restriction on rate from each sensor terminal to the fusion center. This simple result contrasts with the previous ad hoc studies containing many unnatural assumptions such as the independence of noises contaminating received signal at each sensor. For the design of optimal coder, we can use the classical Levinson-Wiggins-Robinson fast algorithm for block Toeplitz matrix to evaluate the necessary weight vector for the maximum-likelihood detection.

A dual transformation incorporating the frequency-dependent scaling factor with the impedance dimension is proposed to synthesize the current-mode counterpart of a voltage-mode original. A general class of current-mode active-RC biquadratic filters and a switched-capacitor low-pass biquad are derived to demonstrate the synthesis procedure. Their simulation and test results show that the current transfer functions are the same as the voltage transfer functions of the originals, and thus confirm the validity of the procedure. The dual trasformation described herein is general in that with the scaling factor chosen appropriately it can meet a wide variety of circuit transformation, and thus useful also for circuit classification and identification.

This paper examines the key technologies and applications of optical frequency division multiplexing (OFDM) systems. It is clarified that a 100-channel OFDM system is feasible as a result of multichannel frequency stabilization, common optical amplification and channel selection utilizing a tunable optical filter. Transmission limitation due to fiber four-wave mixing is also described. Major functions and applications of the OFDM are summarized and the applicability of OFDM add/drop multiplexing is examined.

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.

It is sometimes required to change the frequency characteristics of a digital filter during its operation. In this paper a new synthesis of variable even-order IIR digital filters is proposed. The cut-off frequency of the filter can be changed by a single parameter. The fundamental filter structure is a cascade of second-order sections. The multiplier coefficients of each section are determined by using the Taylor series expansion of the lowpass to lowpass frequency transformation. For this method any second-order section can be used as a prototype, but here in this paper only the direct form and the lattice form are described. Unlike the conventional method, any transfer functions can be used for the proposed method. Finally a designed example shows that the proposed filter has wider tuning range than the conventional filter, and the advantage of the proposed filters is confirmed.

Unlike a noise removal recursive or averaging filter, this letter presents a temporal filter which attenuates temporal high frequency components and improves visual effects. Although temporal aliasing occurs, the proposed filter proceeds temporal bandlimitation not affected by them. To reduce effects caused by aliasing components, a spatial filtering which is applied along the trajectory of motion is investigated. The proposed filter presents a de-aliasing and effective bandlimiting characteristics as well as reducing of noises.

Binaural effects in two measures are studied. One measure is the detectable limen of click sounds under lateralization of diotic or dichotic noise signals, and the other is phoneme articulation score under localization or lateralization of speech and noise signals. The experiments use a headphones system with listener's own head related transfer function (HRTF) filters. The HRTF filter coefficients are calculated individually from the impulse responses due to the listener's HRTF measured in a slightly sound reflective booth. The frequency response of the headphone is compensated for using an inverse filter calculated from the response at the subject's own ear canal entrance point. Considering the speech frequency band in tele-communication systems is not sufficiently wide, the bandwidth of the HRTF filter is limited below 6.2 kHz. However, the experiments of the localization simulation in the horizontal plane show that the sound image is mostly perceived outside the head in the simulated direction. Under simulation of localization or lateralization of speech and noise signals, the phoneme articulation score increases when the simulation spatially separates the phonemes from the noise signals while the total signal to noise ratio for both ears is maintained constant. This result shows the binaural effect in speech intelligibility under the noise disturbance condition, which is regarded as a part of the cocktail party effect.

This paper proposes a technique for designing two-dimensional (2-D) digital filters approximating an arbitrary magnitude function. The technique is based on 2-D spectral factorization and rational approximation of the complex exponential function. A 2-D spectral factorization technique is used to obtain a recursively computable and stable system with nonsymmetric half-plane support from the desired 2-D magnitude function. Since the obtained system has an exponential function type transfer function and cannot be realized directly in a rational form, a class of realizable 2-D digital filters is introduced to approximate the exponential type transfer function. This class of filters referred to as two-dimensional log magnitude approximation (2-D LMA) filters can be viewed as an extension of the class of 1-D LMA filters to the 2-D case. Filter coefficients are given by the 2-D complex cepstrum coefficients, i.e., the inverse Fourier transform of the logarithm of the given magnitude function, which can be efficiently computed using 2-D FFT algorithm. Consequently, computation of the filter coefficients is straightforward and efficient. A simple stability condition for the 2-D LMA filters is given. Under this condition, the stability of the designed filter is guaranteed. Parallel implementation of the 2-D LMA filters is also discussed. Several examples are presented to demonstrate the design capability.

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.