The requirement of structural boundedness or passivity leads to important classes of digital filters among which are the wave digital (WD) filters and the LBR cascade structures having low coefficient sensitivity. Contrary to the WD filters, the LBR filters are directly synthesized in z-domain and several authors presented different approaches for a better understanding of the synthesis procedure especially for complex transfer functions. Some tentatives were also made to give parallels between passive analog and digital filters (i.e. WD or LBR filters). A general approach to LBR synthesis with transmission zeros not necessarily on the unit circle is presented along with some explicit expressions for the LBR (and the generalized complex counterpart LBC) filter parameters for the realization of an input transfer function. The results can be of interest in automated procedures for low sensitivity digital filter design.

The aim of this study is to evaluate mental workload (MWL) quantitatively by HRV (Heart Rate Variability) measures. The electrocardiography and the respiration curve were recorded in five different epochs (1) during a rest condition and (2) during mental arithmetic tasks (addition). In the experiment, subjects added two numbers. The work levels (figures of the number in the addition) were set to one figure, two figures, three figures and four figures. The work level had effects on the mean percent correct, the number of answers and the mean processing time. The psychological evaluation on mental workload obtained by the method of paired comparison increased with the work level. Among the statistical HRV measures, the number of peak and trough waves could distinguish between the rest and the mental loading. However, mental workload for each work level was not evaluated quantitatively by the measure. The HRV measures were also calculated from the power spectrum estimated by the autoregressive (AR) model identification. The ratio of the low frequency power to the high frequency power increased linearly with the work level. In conclusion, the HRV measures obtained by the AR power spectrum analysis were found to be sensitive to changes of mental workload.

This paper presents a design of low-power CMOS OTA-C filters suitable for on-chip integration of advanced monolithic system LSIs that have analog I/O and digital signal processing capability. First, we discuss the distortion of MOS cross-coupled circuits which have a quite low distortion when the MOS FETs have the square law characteristics. Considering the nonidealities of MOS FET, however, we find that the third harmonic component of signal dominates the total harmonic distortion (THD) of the cross-coupled pair circuit. We propose a new architecture to reduce the 3rd harmonic component. Low distortion operational transconductance amplifiers (OTA) which consist of the proposed low distortion cross-coupled pair are applied to the realization of OTA-Capacitor filters. The SPICE simulation shows that the THD of the filter is 0.0047% and the power dissipation is 22.6 mW.

This paper proposes a novel method to realize highly linear MOS circuits using MOSFETs in the nonsaturation region. The proposed method is based on the cancellation of nonlinearity of two MOSFETs by using a current inversiontype negative impedance converter. First, grounded and floating resistor realizations are discussed. Next, by exploiting the MOS resistor circuits, gyrators and inductors are realized. As an application example, a third-order doubly-terminated LC filter is simulated. SPICE analysis shows low total harmonic distortions, excellent controllability and small gain error in the passband.

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.

It is often desired to change the cutoff frequencies of digital filters in some applications like digital electronic instruments. This paper proposes a design of variable lowpass digital filters with wider ranges of cutoff frequencies than conventional designs. Wave digital filters are used for the prototypes of variable filters. The proposed design is based on the frequency scaling in the s-domain, while the conventional ones are based on the z-domain lowpass-to-lowpass transformations. The first-order approximation by the Taylor series expansion is used to make multiplier coefficients in a wave digital filters obtained from a frequency-scaled LC filter become linear functions of the scaling parameter, which is similar to the conventional design. Furthermore this paper discusses the reduction of the approximation error. The curvature is introduced as the figure of the quality of the first-order approximation. The use of the second-order approximation to large-curvature multiplier coefficients instead of the first-order one is proposed.

For the effective control of microwaves in the frequency domain, we propose a new method utilizing current distributions of standing waves on the terminated microstrip line. We analized a short ended microstrip line using the (FD)2TD method to indicate the effectiveness of our proposal. Further we proposed an optically controlled microstrip filter as an application of this method.

This paper presents an equation capable of briefly evaluating the length of white noise sequence to be sent as a training signal. The equation is formulated by utilizing the formula describing the convergence property, which has been derived from the IIR filter expression of the NLMS algorithm. The result revealed that the length is directly proportional to I/[K(2-K)] where K is a step gain and I is the number of the adaptive filter taps.

A two-cascaded image processing approach to enhance the subtle differences in X-ray CT image is proposed. In the method, an asymmetrical non-linear subfilter is introduced to reduce the noise inherent in the image while preserving local edges and directional structural information. Then, a subfilter is used to compress the global dynamic range of the image and emphasize the details in the homogeneous regions by performing a modular transformation on local image den-sities. The modular transformation is based on a dynamically defined contrast fator and the histogram distributions of the image. The local contrast factor is described in accordance with Weber's fraction by a two-layer neighborhood system where the relative variances of the medians for eight directions are computed. This method is suitable for low contrast images with wide dynamic ranges. Experiments on X-ray CT images of the head show the validity of the method.

This letter proposes a microstrip band elimination filter having an optically controlled small gap on a resonant section for the shift of the eliminated frequency range using the semiconductor plasma. The basic characteristics of this filter are analized theoretically utilizing the (FD)2TD method.

Multiplication-accumulation is the basic computation required for image filtering operations. For real-time image filtering, very high throughput computation is essential. This work proposes a hardware algorithm for an application-specific VLSI architecture which realizes an area-efficient high throughput multiplier-accumulator. The proposed algorithm utilizes a priori knowledge of filter mask coefficients and optimizes number of basic hardware components (e.g., full adders, pipeline latches, etc.). This results in the minimum area VLSI architecture under certain input/output constraints.

A method for detecting multiple rigid motions in images from an optical flow field obtained with multi-scale, multi-orientation filters is proposed. Convolving consecutive gray scale images with a set of eight orientation-selective spatial Gaussian filters yields eight gradient constraint equations for the two components of a flow vector at every location. The flow vector and an uncertainty measure are obtained from these equations. In the neighborhood of motion boundary, the uncertainty of the flow vectors increase. By using multiple sets of filters of different scales, multiple flow vectors are obtained at every location, from which the one with minimal uncertainty measure is selected. The obtained flow field is then segmented in order to solve the aperture problem and to remove noise without blurring discontinuity in the flow field. Discontinuities are first detected as those locations where flow vectors have relatively larger uncertainty measures. Then similar flow vectors are gouped into regions. By modeling flow vectors, regions are merged to form segments each of which belongs to a planar patch of a rigid object in the scene.

The transient scattering of a half sine pulse wave by a conducting rectangular cylinder with an open sidewall is rigorously analyzed by using the point matching method (taking into account the edge condition exactly) combined with the fast inversion of Laplace transform. Numerical results are presented for back scattered and forward scattered responses of the far fields when a half sine pulse is incident on the open side and the closed side of the cylinder. The physical meaning of the transient responses is discussed in detail. The comparison of the responses with those by a perfect conducting rectangular cylinder is presented.

A systematic theory of the optimum sub-band interpolation using parallel wavelet filter banks presented with respect to a family of n-dimensional signals which are not necessarily band-limited. It is assumed that the Fourier spectrums of these signals have weighted L2 norms smaller than a given positive number. In this paper, we establish a theory that the presented optimum interpolation functions satisfy the generalized discrete orthogonality and minimize the wide variety of measures of error simultaneously. In the following discussion, we assume initially that the corresponding approximation formula uses the infinite number of interpolation functions having limited supports and functional forms different from each other. However, it should be noted that the resultant optimum interpolation functions can be realized as the parallel shift of the finite number of space-limited functions. Some remarks to the problem of distinction of images is presented relating to the generalized discrete orthogonality and the reciprocal property for the proposed approximation.

The size and weight of marine pulse radar systems must be limited in order to mount them on board boats. However, the azimuthal resolution of a marine radar with a small antenna is degraded by the antenna beam width. It is desirable to use signal processing techniques to increase both the azimuthal resolution and the range resolution of such systems without changing their external configuration. This paper introduces a resolution enhancement method based on deconvolution, which is a kind of inversion. The frequency domain deconvolution method is described first. The effectiveness of the proposed method is shown by simulation. Then, an example of resolution enhancement processing is applied to a pulse radar. The results of practical experiments show that this method is a promising way of upgrading radars by simply processing the received signals.

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.

Most of actual environmental systems show a complicated fluctuation pattern of non-Gaussian type, owing to various kinds of factors. In the actual measurement, the fluctuation of random signal is usually contaminated by an external noise. Furthermore, it is very often that the reliable observation value can be obtained only within a definite fluctuating amplitude domain, because many of measuring equipments have their proper dynamic range and original random wave form is unreliable at the end of amplitude fluctuation. It becomes very important to establish a new signal detection method applicable to such an actual situation. This paper newly describes a dynamical state estimation algorithm for a successive observation contaminated by the external noise of an arbitrary distribution type, when the observation value is measured through a finite dynamic range of measurement. On the basis of the Bayes' theorem, this method is derived in the form of a wide sense digital filter, which is applicable to the non-Gaussian properties of the fluctuations, the actual observation in a finite amplitude domain and the existence of external noise. Differing from the well-known Kalman's filter and its improvement, the proposed state estimation method is newly derived especially by paying our attention to the statistical information on the observation value behind the saturation function instead of that on the resultant noisy observation. Finally, the proposed method is experimentally confirmed too by applying it to the actual problem for a reverberation time measurement from saturated noisy observations in room acoustics.

A narrowband interference in direct sequence spread spectrum communication systems also affects the characteristics of a delay lock loop. In this paper, the delay errors of a baseband delay lock loop (DLL) in the presence of the interference which consists of a narrowband Gaussian noise and several tones are examined, and when a filter is used to reject the interference, the characteristics of the DLL are analyzed using the Fourier method. Furthermore, from the calculation results of the delay error in case where a prediction error filter with two-sided taps is used as the rejection filter, it is shown that the filter is necessary to keep the DLL in the lock-on state.

In this paper, we attempt the comparison of the image/signal restoration between Projection Filter, which is regarded as one of the linear optimal filters, and the non-linear filter based on MEM. From the simulation, we show the advantage of MEM restoration filter in restoring noisy degraded images.

An adaptive signal processing using Acoustic Charge Transport device, which has great potential for processing very wide band signals in real time, is investigated. It shows that adaptive system for signals of bandwidth from dc up to 500 MHz can be implemented in real time.