We present a finite-difference time-domain (FD-TD) method with the perfectly matched layers (PMLs) absorbing boundary condition (ABC) based on the multidimensional wave digital filters (MD-WDFs) for discrete-time modelling of Maxwell's equations and show its effectiveness. First we propose modified forms of the Maxwell's equations in the PMLs and its MD-WDFs' representation by using the current-controlled voltage sources. In order to estimate the lower bound of numerical errors which come from the discretization of the Maxwell's equations, we examine the numerical dispersion relation and show the advantage of the FD-TD method based on the MD-WDFs over the Yee algorithm. Simultaneously, we estimate numerical errors in practical problems as a function of grid cell size and show that the MD-WDFs can obtain highly accurate numerical solutions in comparison with the Yee algorithm. Then we analyze several typical dielectric optical waveguide problems such as the tapered waveguide and the grating filter, and confirm that the FD-TD method based on the MD-WDFs can also treat radiation and reflection phenomena, which commonly done using the Yee algorithm.

In this paper, two universal building blocks for complex filter using CCIIs, CFCCIIs, grounded resistors and grounded capacitors are presented. These can be used to realize various complex bandpass filters with arbitrary order. The paper shows that the response error of the proposed circuit caused by nonideality of active components is more easily compensated than that of the conventional one employing op-amps, and that the sensitivities for all components are relatively small. Experimental results are used for verifying the validity of the proposed circuits.

This letter proposes a Perfect-Reconstruction (PR) encryption scheme based on a PR QMF bank. Using the proposed scheme, signals can be encrypted and reconstructed perfectly by using two Periodically Time-Varying (PTV) digital filters respectively. Also we find that the proposed scheme has a "good" encryption effect and compares favorably with frequency scramble in the aspects of computation complexity, PR property, and degree of security.

An adaptive infinite impulse response (IIR) filter implemented using an allpass and a minimum phase system has an advantage of its poles converging to the poles of the unknown system when the input is a white signal. However, when the input signal is colored, convergence speed deteriorates considerably, even to the point of lack of convergence for certain colored signals. Furthermore with a colored input signal, there is no guarantee that the poles of the adaptive digital filter (ADF) will converge to the poles of the unknown system. In this paper we propose a method which uses a linear predictor filter to whiten the input signal so as to improve the convergence characteristic. Computer simulation results confirm the increase in convergence speed and the convergence of the poles of the ADF to the poles of the unknown system even when the input is a colored signal.

This paper presents two types of two-dimensional (2-D) adaptive beamforming algorithm which have high rate of convergence. One is a linearly constrained minimum variance (LCMV) beamforming algorithm which minimizes the average output power of a beamformer, and the other is a generalized sidelobe canceler (GSC) algorithm which generalizes the notion of a linear constraint by using the multiple linear constraints. In both algorithms, we apply a 2-D lattice filter to an adaptive filtering since the 2-D lattice filter provides excellent properties compared to a transversal filter. In order to evaluate the validity of the algorithm, we perform computer simulations. The experimental results show that the algorithm can reject interference signals while maintaining the direction of desired signal, and can improve convergent performance.

We have developed an efficient recursive algorithm based on the interacting multiple model (IMM) for enhancing speech degraded by additive white noise. The clean speech is modeled by the hidden filter model (HFM). The simulation results shows that the proposed method offers performance gains relative to the previous one with slightly increased complexity.

In addition to additive thermal noise, a received direct-sequence spread spectrum (DS/SS) signal may suffer from intersymbol interference (ISI) and interference caused by cochannel narrowband users or other narrowband radio frequency interference (RFI). This paper presents a workable solution for removing narrowband interference (NBI) and reducing ISI or inter-chip interference (ICI) when the communication channel can be modeled as an FIR filter and the NBI comes from multiple CW tones, an AR-modeled Gaussian process, or a BPSK signal. Unlike earlier solutions, the proposed scheme is capable of performing the functions of NBI-rejection, ISI/ICI suppression and data detection (code despreading) simultaneously. It is easy to implement and, more importantly, it yields lower bit error rate (BER) and smaller mean squared error (MSE).

This paper proposes a novel synchronous acquisition method with an adaptive filter in asynchronous direct sequence/code division multiple access (DS/CDMA) communication systems. An adaptive filter is used in a single-user receiver, in complete synchronization of desired user's signal, the tap coefficients of the filter are controlled to orthogonalize to all other user's spreading sequences without knowledge of the sequences, amplitude and time delays of the signals. While, in the proposed system for synchronous acquisition, the tap coefficients are controlled to orthogonalize to all user's sequences including desired user's signal. The synchronous acquisition can be achieved by using the difference of cross-correlation function value between desired user's sequence of inphase and the tap coefficients for each phase. The principle and performance evaluation for the proposed method are shown. As a result, compared to an acquisition method of conventional sliding correlator, considerable improvement of the average acquisition time can be achieved in large power multiple access interference environment.

A novel voltage-tunable linear 3-input CMOS Operational Transconductance Amplifier (OTA) suitable for onchip integration of advanced monolithic systems is proposed. When a 3-input OTA is needed, a conventional 3-input OTA uses two 2-input OTA's and either grounds one of the 4 input terminals or ties two terminals. This paper presents a method to reduce the number of MOS transistors and to save chip area by designing a 3-input OTA directly. A CMOS pair technique is introduced s a solution to minimize a matching problem for control voltage sources. Simulation results show that the active chip area of the proposed 3-input OTA is reduced by 25% compared to that of a conventional one. The proposed 3-input OTA is applied to a realization of an OTA-C filter to verify the effectiveness.

A wireless communications system with a transmission rate of 10 Mbit/s using Japanese ISM band (2471-2497 MHz) is presented. This system employs a novel spread spectrum multiple access method named "CFO-SS (Carrier Frequency Offset-Spread Spectrum)" method. In the CFO-SS system, a single PN code is commonly assigned to all the multiple carriers, and the frequency offset between the carriers is determined by the information symbol rate, which is small as compared with the spread bandwidth of the signal. Bit error rate performance of the proposed CFO-SS system under multipath environments is investigated by computer simulation, and the performance of the CFO-SS method is confirmed for wireless LAN systems using the 2.4 GHz ISM band.

We consider the situation that plural degraded images are obtained. When no prior knowledge about original images are known, these images are individually restored by an optimum restoration filter, for example, by Wiener Filter or by Projection Filter. If correlations between original images are obtained, some restoration filters based on Wiener Filter or Projection Filter are proposed. In this paper, we deal with the case that some pixels or some parts of original images overlap, and propose a restoration method using a formulae for overlapping. The method is based on Partial Projection Filter. Moreover, we confirm an efficacy of the proposed method by numerical examples.

A novel neural network architecture for image texture classification is introduced. The proposed model (Kernel Modifying Neural Network: KM Net) which incorporates the convolution filter kernel and the classifier in one, enables an automated texture feature extraction in multichannel texture classification through the modification of the kernel and the connection weights by the backpropagation-based training rule. The first layer units working as the convolution kernels are constrained to be an array of Gabor filters, which achieves a most efficient texture feature localization. The following layers work as a classifier of the extracted texture feature vectors. The capability of the KM Net and its training rule is verified using a basic problem on a synthetic texture image. In addition, the possibilities of applying the KM Net to natural texture classification and biological tissue classification using an ultrasonic echo image have been tried.

Polarization transformation characteristics of a statified slab consisting of uniaxial chiral layers are investigated. It is assumed that a plane electromagnetic wave with arbitrary polarization is normally incident from free space on the stratified slab, which is located on a dielectric substrate. Note that the electric field inside a uniaxial chiral layer is expressed as a sum of four plane waves with different wavenumbers. The wavenumbers are found by seeking non-trivial solutions of the constitutive relations with Maxwell's equations. The electric field components of the transmitted and reflected waves can be obtained from a chainmatrix formalism. The powers and the Stokes parameters of the two waves are represented in terms of their electric field components. As is well known, the Stokes parameters uniquely describe every possible state of polarization of a plane wave. Numerical results are presented for two types of uniaxial chiral structure. The cross- and co-polarized powers and the Stokes parameters of the transmitted and reflected waves are computed for the incident plane wave of linear polarization. The results demonstrate a significant polarization transformation of the transmitted wave. Then it is shown that the stratified slab can be used as efficient polarization-transformation transmission filters active at some frequency band.

In this paper, we propose a two-dimensional (2-D) least-squares lattice (LSL) algorithm for the general case of the autoregressive (AR) model with an asymmetric half-plane (AHP) coefficient support. The resulting LSL algorithm gives both order and space recursions for the 2-D deterministic normal equations. The size and shape of the coefficient support region of the proposed lattice filter can be chosen arbitrarily. Furthermore, the ordering of the support signal can be assigned arbitrarily. Finally, computer simulation for modeling a texture image is demonstrated to confirm the proposed model gives rapid convergence.

The numerical property of the recursive least squares (RLS) algorithm has been extensively, studied. However, very few investigations are reported concerning the numerical behavior of the predictor-based least squares (PLS) algorithms which provide the same least squares solutions as the RLS algorithm. In Ref. [9], we gave a comparative study on the numerical performances of the RLS and the backward PLS (BPLS) algorithms. It was shown that the numerical property of the BPLS algorithm is much superior to that of the RLS algorithm under a finite-precision arithmetic because several main instability sources encountered in the RLS algorithm do not appear in the BPLS algorithm. This paper theoretically shows the stability of the BPLS algorithm by error propagation analysis. Since the time-variant nature of the BPLS algorithm, we prove the stability of the BPLS algorithm by using the method as shown in Ref. [6]. The expectation of the transition matrix in the BPLS algorithm is analyzed and its eigenvalues are shown to have values within the unit circle. Therefore we can say that the BPLS algorithm is numerically stable.

In this paper, implementation of a first-order active complex filter with variable parameter using operational transconductance amplifiers (OTAs) and grounded copacitors is presented. The proposed configurations can be used as s key building block to realize high-order active complex filters with variable parameter in cascade and leapfrog configuration. Experimental results which are in good agreement with theoretical responses are also given o demonstrate the feasibility of the proposed configurations.

This paper proposes an FFT (Fast Fourier Transform) interference detection for interference suppression which combines notch filtering and FEC (forward error correction) to improve the Pe (probability of error) performance degradation due to co-channel interference in digital satellite communication systems. The proposed FFT interference detection scheme can determine the co-channel interference carrier frequency, power, and bandwidth precisely by using the power detection threshold suitable for the desired signal power spectrum, and the notch filter characteristic can be set according to the results. The interference suppression with the proposed scheme achieves the degradation in required Eb/No to only 1.0 dB at a Pe of 10-4 compared to that with the optimum notch filter (ideal detection) in unknown CW (continuous wave) and FM (frequency modulation) co-channel interference environments. Moreover, the proposed scheme improves the required Eb/No by 6.5 dB compared to that without a notch filter in an FM interference environment with interference carrier frequency offset normalized by the desired signal clock rate of 0.52, desired to undesired (interference) signal power ratio of 3 dB and interference bandwidth at 10 dB down power point from the peak normalized by the desired signal clock rate of 0.25.

A lot of optimum filters have been proposed for an image restoration problem. Parametric filter, such as Parametric Wiener Filter, Parametric Projection Filter, or Parametric Partial Projection Filter, is often used because it requires to calculate a generalized inverse of one operator. These optimum filters are formed by a degradation operator, a covariance operator of noise, and one of original images. In practice, these operators are estimated based on empirical knowledge. Unfortunately, it happens that such operators differ from the true ones. In this paper, we show the unified formulae of inducing them to clarify their common properties. Moreover, we investigate their properties for perturbation of a degradation operator, a covariance operator of noise, and one of original images. Some numerical examples follow to confirm that our description is valid.

This paper describes a design technique of perfect reconstruction (PR) two-channel IIR filter bank. M.J.T. Smith et al., gave two types of PR IIR filter bank systems. One is the system such that the analysis and synthesis filters with nonlinear phase are implemented with all-pass polyphase filters and satisfy the power complementary condition approximately. The other is the system such that all the analysis and synthesis filters have liner phase responses and do not satisfy the power complementary condition. To improve coding performance, we propose a filter bank system such that all the analysis and synthesis filters have linear phase and satisfy the power complementary condition approximately.

This paper presents a new method for constructing orthonormal wavelet bases with vanishing moments based on general IIR filters. It is well-known that orthonormal wavelet bases can be generated by paraunitary filter banks. Then, synthesis of orthonormal wavelet bases can be reduced to design of paraunitary filter banks. From the orthonormality and regularity of wavelets, we derive some constraints to IIR filter banks, and investigate relations between the constrained filter coefficients and its zeros and poles. According to these relations, we can apply Remez exchange algorithm in stopband directly, and formulate the design problem in the form of an eigenvalue problem. Therefore, a set of filter coefficients can be easily computed by solving the eigenvalue problem, and the optimal filter coefficients with an equiripple response can be obtained after applying an iteration procedure. The proposed procedure is computationally efficient, and the number of vanishing moments can be arbitrarily specified.