For reduction of computational complexity in the IA algorithm, the thinned-out IA algorithm in which only one step size is updated every iteration is proposed and is complementarily switched with the HA algorithm according to the convergence. The switching is determined by using the gradient of the error signal power. These are investigated through the computer simulations.

In this paper, we propose a new design algorithm for nearly linear phase IIR digital filters with prescribed log magnitude response. The error function used here is the sum of the weighted log magnitude-squared error and phase -squared error, and so it is possible to control log magnitude and phase response directly. The gradient vector of the proposed error function is easily calculated as the closed form solution because of its nature, in which the real and imaginary part of the logarithm of a complex transfer transfer function corresponds to the log magnitude and phase response, respectively. This algorithm is simple and converges quickly. Finally, we show the validity of the proposed algorithm with some examples.

This letter proposes a map-matching method for automotive navigation systems. The proposed method utilizes the innovation of the Kalman filter algorithm and can achieve more accurate positioning than the correlation method which is generally used for the navigation systems. In this letter, the performance of the proposed algorithm is verified by some simulations.

This paper describes a narrow pass-band optical filter utilizing a wavelength-sensitive power-transfer characteristic in the directional coupler composed of the K-and Ag-ion exchange waveguides which have greatly different dispersion relations caused by the large mismatch in the index profile of the waveguide cross-section. A narrow pass-band width of about 7 nm is measured in the filter fabricated in the soda-lime glass substrate. The fabrication technique with two-step ion-exchange of the K-and Ag-ions, is also presented together with a quick design method.

To improve the demodulated signal-to-noise ratio, SNR, for colored noise environments, we present a new direct-sequence spread-spectrum receiver system, whose construction is based on the concept of Shaped M-sequence Demodulation (SMD). This receiver has the function for shaping the local dispreading-code waveform. This method can modify the frequency transfer function from a received input to the damp-integrated output according to the power spectrum of colored noise added in the transmission process. SMD performs the combined function of a whitening filter and a matched filter, which can be used to implement an optimum receiver. For the case when the additive colored-noise power spectrum is known and the transmission channel is non-band-limited, a design theory is derived that provides the maximum SNR by choosing the best dispreading-code sequence corresponding to a given signature spreading-code sequence. The noise power component produced in the receiver damp-integrated-output is anayzed by introducing the auto-correlation matrix of the additive noise. The SNR performance of systems, one using non-optimized codes and the other using optimized codes, is examined and compared for various noise models. It is verified by analysis and computer simulation that, compared to a conventional system using non-optimized codes, remarkable SNR improvements can be achieved due to the whitening effect acquired without producing inter-symbol interference. In contrast, if a transversal whitening filter is front-ended, it produces inter-frame interference, degrading the SNR performance. The band-limiting effect of the transmission channel is also analyzed, and we confirmed that the codes optimized for the non-band-limited channel can be applied to the band-limited channel with little degradation of SNR. SMD is inherently tolerant of fast-changing noise such as fading, due to its frame-by-frame operation. Considering this function as a general demodulation scheme, it may be called "Local Code Filtering."

This paper describes the properties of a TE10 metal waveguide filter using two polycrystalline ferrite chips at millimeter wave frequencies. The frequency response of the filter has been analyzed using the mode-matching technique, and optimized by the computer technique. The bandpass filter characteristics with high dynamic range more than 30dB was obtained with insertion loss of 1.5dB and good magnetically tunable response is observed with a quality factor of 200, which agrees considerably well with predicted values.

While high compression ratio has been achieved using recently developed image coding algorithm, the noise removal technique is considered as an important subject. This still holds for very low bitrate video coding, that is, MPEG-4 has defined it as a core experiment which is mainly concerned with block based discrete cosine transform (DCT) coding such as H.263 and MPEG-1. This paper describes a novel and practical technique which attempts to accomplish both noise suppression and detail preservation at the same time. Some of the conventional adaptive filters are designed to search a homogeneous region among the predetermined polygonal subregions, then to apply a smoothing operation within the selected subregion. It shall be, however noted that sometimes the predetermined subregion finally selected may still be hererogeneous. This fact leads us to a novel idea; instead of examining the predetermined regions, define a lot more flexible region likely to be homogeneous. In order to achieve this, we introduce the binary index. each pixel is classified into either the lower intensity group or higher intensity group based on a local statistics. Then a smoothing operation is applied within the pixels having the same group index as the pixel to be processed. Thus our scheme can search a homogeneous region appropriately. The adaptive smoothing adopted in the proposed scheme is also designed to be consistent with an important property of human visual system, i.e., the spatial masking. noise visibility decreases at spatial details such as edges and textures. Another advantage is that it can be realized with significantly low computations. The simulation results show that his approach can suppress the visible artifacts while retaining the fine details such as edge and texture.

This paper proposes and investigates the adaptive multi-user equalizer based on the multi-dimensional IIR adaptive lattice filter in order to suppress the co-channel interference (CCI) in asynchronous DS/CDMA system. An asynchronous DS/CDMA system with multi-user receiver is modeled as multi-dimensional or multi-input/out system with cross-coupling that is co-channel interference. From the system model it is shown that the multi-user detection is reduced into a problem of multi-dimensional equalization for multiaccess interference as well as intersymbol interference. The proposed multi-user equalizer can improve the equalizing error of the filter, comparing with that of the multi-dimensional FIR transversal filter of which number of tap is finite. The multi-dimensional lattice filter can adaptively achieve fast and stable convergence with less taps. Since the filter can resolve correlative multiple input into orthogonal output stage by stage, CCI can be removed. Computer simulations show performance of the proposed scheme.

For direct sequence spread spectrum systems, the performance of PN sequence acquisition can be significantly affected if data modulation is present. However, the data modulation often exists during the reacquisition of a PCS radio channel. This study proposes and analyzes two shemes which are designed to improve acquisition process for PN sequence under data modulation. Both designs are based upon a PN acquisition receiver with parallel I-Q matched filters. The first scheme employs a serial search strategy with verification mode. The second scheme, which is still based upon the same parallel acquisition receiver, employs the parallel search strategy. We show that the second scheme is capable of providing faster acquisition under data modulation than the first serial search scheme using the same number of I-Q matched filter. We believe it should become a very good alternative for the acquisition of data modulated PN sequences in personal communications.

A high-Tc superconducting filter of the planar structure is proposed for handling higher power signals and for miniaturizing the filter configuration. The filter is designed with a single disk-resonator shared by two degenerate modes to operate as a two-stage bandpass filter. Thereby the proposed filter is expected to possess high power handling capability as a conventional filter with two resonator disks does while the filter configuration is about a half in area compared to the conventional one. The Tchebyscheff type filter with 5.1 GHz center frequency and 2% relative bandwidth was fabricated using a high-Tc superconducting thin film. The passband insertion loss, Lo, was approximately 0.8 dB at 77 K. The low loss performance due to the superconductivity was observed at incident signal levels up to 41.2 dBm (around 15 W) at 20 K, which is limited by the power devices in the measurement setup. In addition, good linearity in the filter responses was confirmed by observing the intermodulation distortion with the two-tone method, which indirectly shows a stable operation with higher power incident signals.

An image obtained by ultrasonic medical equipment is poor in quality because of speckle noise, that is caused by the quality of ultrasonic beam and so on. Thus, it is very difficult to detect internal organs or the diseased tissues from a medical ultrasonic image by the processing, which is used only gray-scale of the image. To analyze the ultrasonic image, it is necessary to use not only gray-scale but also appropriate statistical character. In this paper, we suggest a new method to extract regions of internal organs from an ultrasonic image by the discrimination function. The discrimination function is based on gray-scale and statistical characters of the image. This function is determined by using parameters of the multi-dimensional autoregressive model.

This paper first presents the performance analysis of the NACF algorithm. The results show the possibility of the degradation in the convergence speed. To improve the convergence speed, the bias term is introduced into the NACF algorithm and its efficiency is investigated through the computer simulations.

This paper proposes fast FIR digital filter structures using the minimal number of adders. Filter coefficients are expressed with canonic signed digit (CSD) code and Hartley's technique is used to minimize the number of adders and subtractors. The proposed filters implemented as wired logic are fast because the structure having the shortest critical path is selected. Two algorithms are given to obtain such fast structures. In many examples the critical path length of the filter structures obtained using the proposed method is equal to that of the conventional CSD structures. This paper also presents a new design method of FIR filters using the mixed integer programming (MILP). Utilization of common expressions in Hartley's technique widens the CSD coefficient space. Thus the MILP may lead to better frequency responses. Superior frequency responses are actually obtained in many simulations.

The ringing occurred in the step response causes an undesirable stripe pattern in TV signals. A simultaneous approximation with both the frequency and the step response is required in the designing filter which is used in the image signal processing in order to prevent the ringing. The wellknown Remez algorithm for designing FIR filters approximates the response only in the frequency domain. As the result, the filters designed by this algorithm causes the large ringing in the step response. In this paper, we propose the method of design for FIR filters with minimum amplitude in the stopband, under the condition that the step response has no ringing and the prescribed rise characteristics. For this end, we use the constrained successive projections method.

This paper presents a new method for designing IIR Nyquist filters with zero intersymbol interference. It is shown that IIR Nyquist filters with zero intersymbol interference have some constraints on frequency response, i.e., both magnitude and phase error in passband are dependent on stopband error. Therefore, the frequency response is required to optimize only in stopband. The proposed procedure is based on the formulation of an eigenvalue problem by using Remez multiple exchange algorithm in stopband. Then, the filter coefficients can be computed by solving the eigenvalue problem, and the optimal solution with equiripple stopband response is easily obtained by applying an iteration procedure. The proposed procedure is more computationally efficient than the conventional methods.

This paper proposes a three-dimensional (3-D) band-limiting technique for a conversion of Simple Cubic Sampling into Body-Centered Cubic Sampling. Based on spectral distribution of the original signal, the proposed method adaptively varies the passband shape of a band-limiting filter in order to preserve informations of the original picture. By applying this method to 3-D moving pictures, we can preserve resolution on each axis without introducing heavy aliasing error and avoid degradation of picture quality such as ringing in still pictures or blurring in moving pictures. The examples given in this paper demonstrate these advantages.

This paper considers the estimation of speech parameters and their enhancement using an approach based on the estimation-maximization (EM) algorithm, when only noisy speech data is available. The distribution of the excitation source for the speech signal is assumed as a mixture of two Gaussian probability distribution functions with differing variances. This mixture assumption is experimentally valid for removing the residual excitation signal. The assumption also is found to be effective in enhancing noise-corrupted speech. We adaptively estimate the speech parameters and analyze the characteristics of its excitation source in a sequential manner. In the maximum likelihood estimation scheme we utilize the EM algorithm, and employ a detection and an estimation step for the parameters. For speech enhancement we use Kalman filtering for the parameters obtained from the above estimation procedure. The estimation and maximization procedures are closely coupled. Simulation results using synthetic and real speech vindicate the improved performance of our algorithm in noisy situations, with an increase of about 3 dB in terms of output SNR compared to conventional Gaussian assumption. The proposed algorithm also may be noteworthy in that it needs no voiced/unvoiced decision logic, due to the use of the residual approach.

In many actual applications of the adaptive filtering, input signals as well as output signals often contain observation noises. Hence, it is necessary to develop an adaptive filtering algorithm to such an errors-in-variables (EIV) model. One solution for identifying the EIV model is a total least squares (TLS) algorithm based on a singular value decomposition of an off-line processing. However, it has not been considered to identify the EIV IIR system using an adaptive TLS algorithm of which stability has been guaranteed during adaptation process. Hence we propose a normalized lattice IIR adaptive filtering algorithm for the TLS parameter estimation. We also show the effectiveness of the proposed algorithm under noisy circumstances through simulations.

The present paper investigates a two-dimensional (2-D) adaptive lattice filter used for modeling 2-D AR fields. The 2-D least mean square (LMS) lattice algorithm is used to update the filter coefficients. The proposed adaptive lattice filter can represent a wider class of 2-D AR fields than previous ones. Furthremore, its structure is also shown to possess orthogonality in the backward prediction error fields. These result in superior convergence and tracking properties to the adaptive transversal filter and other adaptive 2-D lattice models. Then, the convergence property of the proposed adaptive LMS lattice algorithm is discussed. The effectiveness of the proposed model is evaluated for parameter identification through computer simulation.

A classs of type 1 linear phase FIR digital filters is proposed. The filter can be realized using a parallel, modular and regular array structure. It is shown that, under some simple constraints, the consisting modules of the array can be realized free of multiplier coefficients. Such two dimensional mesh arrays are specially suitable for realization with special-purpose systolic hardware for high-speed digital signal processing tasks. Compared to the array structure, proposed by the authors, for multiplierless realization of maximally flat FIR digital filters, this class needs less adders to fulfill the same magnitude response requirements. Another attractive property of the proposed array is that a number of highpass or lowpass filters with different passband widths can be realized simultaneously in a very economical way.