This paper proposes a fast convergence algorithm for adaptive FIR filters with sparse taps. Coefficient values and positions are simultaneously controlled. The proposed algorithm consists of two stages： flat-delay estimation and tapposition control with a constraint. The flat-delay estimation is carried out by estimating the significant dispersive region of the impulse response. The constrained tap-position control is achieved by imposing a limit on the new-tap-position search. Simulation results show that the proposed algorithm reduces the convergence speed by up to 85% over the conventional algorithms for a white signal input. For a colored signal, it also converges in 40% of the convergence time by the conventional algorithms. The proposed algorithm is applicable to adaptive FIR filters which are to model a path with long flat delay, such as echo cancelers for satellite-link communications.

Optical amplifier structures suitable for a 622Mbit/s repeater in an optical communication system containing one in-line amplifier have been investigated. Two wavelengths of 1.533µm and 1.549µm are considered for two cases, i.e., single-channel transmission and two-channel wavelength division multiplexing transmission. The basic amplifier structure is of a two-stage type where forward pumped and backward pumped erbium-doped fibers are connected with each other through intermediate optical filters and an optical isolator. First, the effect of the intermediate optical filters was clarified in optical gain and bit error rate characteristics. Then, the erbium-doped fiber length was optimized on the basis of the allowable optical loss of the optical system which was operated at a bit error rate of 10-9. As a result, the appropriate length of the forward pumped erbium-doped fiber was found to be about 20m for both cases of single-channel and two-channel wavelength multiplexing amplifiers. With the designed amplifier used in the system, the calculated allowable optical line loss was more than 90dB for both the cases.

The timing jitter reductions with differently shaped optical bandpass filters are discussed and the transmission distance achievable against the timing jitter is evaluated using optical bandpass filters in several tens of Gb/s soliton transmission. Experimental confirmation of timing jitter reduction with optical bandpass filters is demonstrated in 10Gb/s optical soliton recirculating loop experiments by measuring the timing jitter and the bit error rates.

This paper investigates the properties of synchronized alternating one-way multicounter machines (lsamcm's) which operate in real time (lsamcm-real's) and whose leaf-sizes are bounded by a constant or some function of the length of an input. Leaf-size reflects the number of processors which run in parallel in scanning a given input. We first consider the hieracrchies of lsamcm-real's based on the number of counters and constant leaf-sizes. We next show that lsamcm-real's are less powerful than lsamcm's which operate in linear time when the leaf-sizes of these machines are bounded by a function L(n) such that limn[L(n) log n/n]0 and L(n)2.

The scope of this paper is the realization of FIR digital filters with an emphasis on linear phase and maximally flat cases. The transfer functions of FIR digital filters are polynomials and polynomial evaluation algorithms can be utilized as realization schemes of these filters. In this paper we investigate the application of a class of polynomial evaluation algorithms called "recursive triangles" to the realization of FIR digital filters. The realization of an arbitrary transfer function using De Casteljau algorithm, a member of the recursive triangles used for evaluating Bernstein polynomials, is studied and it is shown that in some special and important cases it yields efficient modular structures. Realization of two dimensional filters based on Bernstein approximation is also considered. We also introduce recursive triangles for evaluating the power basis representation of polynomials and give a new multiplier-less maximally flat structure based on them. Finally, we generalize the structure further and show that Chebyshev polynomials can also be evaluated by the triangles. This is the triangular counterpart of the well-known Chebyshev structure. In general,the triangular structures yield highly modular digital filters that can be mapped to an array of concurrent processors resulting in high speed and effcient filtering specially for maximally flat transfer functions.

This paper proposes new algorithms for adaptive FIR filters. The proposed algorithms provide both fast convergence and small final misadjustment with an adaptive step size even under an interference to the error. The basic algorithm pays special attention to the interference which contaminates the error. To enhance robustness to the interference, it imposes a special limit on the increment/decrement of the step-size. The limit itself is also varied according to the step-size. The basic algorithm is extended for application to nonstationary signals. Simulation results with white signals show that the final misadjustment is reduced by up to 22 dB under severe observation noise at a negligible expense of the convergence speed. An echo canceler simulation with a real speech signal exhibits its potential for a nonstationary signal.

Although the perception of gloss is based on human visual perception, some methods for extracting glossiness, in contrast to human ability, have been proposed involving curved surfaces. Glossiness defined in these methods, however, does not correspond with psychological glossiness perceived by the human eye over the wide range from relatively low gloss to high gloss. In addition, the obtained glossiness in these methods changes remarkably when the curvature radius of the high-gloss object becomes larger than 10mm. In reality, psychological glossiness does not change. These methods, furthermore, are available only for spherical objects. A new method for extracting glossiness is proposed in this study. For the new definition of glossiness, a spatial filter which simulates human retina function is utilized. The light intensity distribution of the curved object is convoluted with the spatial filter. The maximum value Hmax of the convoluted distribution has a high correlation with psychological glossiness Gph. From the relationship between Gph and Hmax, new glossiness Gf is defined. The gloss-extraction equipment consists of a light source, TV camera, an image processor and a personal computer. Cylinders with the curvature radii of 3-30 mm are used as the specimens in addition to spherical balls. In all specimens, a strong correlation, with a correlation coefficient of more than 0.97, has been observed between Gf and Gph over a wide range. New glossiness Gf conforms to Gph even if the curvature radius in more than 10 mm. Based on these findings, it is found that this method for extracting glossiness is useful for the extraction of glossiness of spherical and cylindrical objects over a wide range from relatively low gloss to high gloss.

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.

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.

For IF direct sampling phase detection method (IFSM) which realizes the arithmetical operations with digital filters by direct A/D (Analog to Digital) conversion of IF (Intermediate Frequency) signal, the method to eliminate DC offset is proposed and developed by using the gate array. A principle of the proposed method and the results of the measurement are shown.

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.

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.

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.

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.

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.

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 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.

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.

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.