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This paper discusses a routing scheme for an autonomous distributed wireless network (mobile ad-hoc networking) using an ultra-wideband (UWB) system. The UWB system is a kind of spread-spectrum communication system which has the function of measuring the distance between the transmitter and the receiver with high accuracy. Using this distance measurement function, the received energy per bit is kept constant by adjusting the spreading factor according to the transmission distance. This paper considers a packet-forwarding scheme based on the above feature of the UWB system, where the spreading factor is adjusted according to the distance between the transmitter and the receiver. The objective is to realize higher throughput than in single-hop transmission by using multihop transmission which controls the maximum transmission distance. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(2): 22–30, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20082 [ABSTRACT FROM AUTHOR]

Checkerboard distortion is the kind generated in the sampling rate converter using a digital filter. In this paper, such distortion is studied in the continuous-time domain. The distortion is newly defined in the problem of interpolation by a continuous interpolation function and conditions for avoiding the checkerboard distortion are derived. Further, it is shown that a digital filter that does not generate checkerboard distortion can be created by an appropriate sampling to the continuous interpolation function satisfying the conditions in this paper. The relationship between the continuous interpolation function with time scale transform and the checkerboard distortion is studied. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(12): 59–68, 2001 [ABSTRACT FROM AUTHOR]

This paper considers the set of waveforms for which the weighted power-of-p (1 < p < +∞) integral of the Fourier spectrum is bounded. Since different weight functions can be used in different frequency bands, as long as they are positive, the restriction of the waveform by the power-of-p integral is sufficiently general. It is assumed that the waveform belonging to such a set is added to several given linear time-invariant filters. A theory of the optimum approximation of the original waveform is established using the sampling sequences of the corresponding outputs. The measure of the original waveform is established using the sampling sequences of the corresponding outputs. The measure of the approximation error is defined as the upper limit of the absolute value of the difference between the original and the approximating waveforms when the original waveforms are changed one by one over the whole set of waveforms. Considering the sampling points, which substantially affect the approximation at a fixed time t, a theorem is presented which indicates that there exists a waveform which has zero value at all such sampling points and has the absolute value of the approximation error which reaches the above upper limit. It is shown also that a similar result can be established for the multidimensional waveform. In this paper, it is shown that the proposed method of linear approximation is better than any other approximation using the same sample values, including the nonlinear signal processing, as long as the forementioned set of waveforms and the measure for the approximation error are considered. [ABSTRACT FROM AUTHOR]

This paper considers approximation formulas of the n-dimensional wave f(X) expressed by the sum of sampled values of f(X) multiplied by certain n-dimensional waves called interpolation functions. A systematic discussion is made to determine whether or not the approximate function converges to the original wave f(X). In this paper it is assumed that the interpolation functions satisfy the causality in regard to a certain axis in the n-dimensional space Rn. Usually, that axis is considered to be the tine axis. These discussions are important from the theoretical viewpoint, since it is closely related to the problem as to whether or not interpolation functions can be realized as impulse responses of physically realizable time-in- variant linear circuits. The set of sampling points is defined as a set of vertices obtained by sampling periodically all the vertices of the n-dimensional parallel polyhedra periodically arranged in the space Rn. This definition of the set of sampling points Includes most of the important arrangements of the sampling points, such as the hexagon and octagon lattices in the two-dimensional plane. In this paper, it is assumed that the interpolation functions have supports, which are parallel translations of each other. The functional forms of the interpolation functions may be different. The set of f(X) considered in this paper is defined as the set of n-dimensional waves whose Fourier spectra have the weighted power-of-p norm (p > 1) smaller than the specified positive value. It is shown first that, under the condition that the squared-sum of the interpolation functions is bounded, the considered approximate expression does not in general converge to the original wave. It is shown that if the I ore-going condition is relaxed, an approximate expression can be found with the impulse response of a causal time-invariant linear network as the interpolation function, which converges to the original wave. [ABSTRACT FROM AUTHOR]

This paper presents a unified theory for the nonuniform sampling theorem for the low-pass band-limited waveform. In most practical cases, the mean interval of the sampling points is in the so-called oversampling state. One can also assume that there exists a certain period in the sampling point sequence. Those properties of the sampling point sequence are assumed in this paper. Yen has derived a sampling theorem for the case where the sampling-point sequence has a period. In his theorem, however, the interpolation function has a spectrum which is a stepwise discontinuous function, not suited to the realization of the interpolation function by a digital or other similar filters. One of the greatest features of the sampling theorem proposed in this paper is that the interpolation function has a continuous spectrum. The problem has been discussed by one of the authors for the case where the sampling frequency and the band-end frequency of the waveform are in an integer ratio. In this paper, the sampling theorem is derived in a unified and integrated form without imposing such a condition. The variance of the approximation error is also derived for the case where a statistically independent additive error is superposed on the sampled value, and it is shown that the proposed sampling theorem is better in that respect. The discussion is extended to the multidimensional band-limited waveform, and a sampling theorem is shown which has an interpolation function with a continuous spectrum. [ABSTRACT FROM AUTHOR]

The YUV 4:2:0 format is a representative format for color images in international standards. In this format, both color difference components U and V reduce the number of pixels in the horizontal and vertical directions by one-half. Consequently, this reduction degrades the color difference components. At a relatively high bit rate, the degradation produced by reducing the number of pixels is often worse than the degradation that depends on the coding. Thus, processing to compensate for the degradation due to the format transform becomes necessary. The pixels of the color difference components reduced once on the encoding side are determined by interpolation on the decoding side. The typical interpolation methods are bilinear interpolation, bicubic interpolation, and interpolation by a finite impulse response (FIR) filter. When these interpolation methods were applied to interpolate the color difference components in the YUV 4:2:0 format, blur and lower saturation were produced in the colors in the color difference components returned to the original number of pixels. In this paper, we propose a method that uses information in the luminance component not having a reduced number of pixels to interpolate the color difference components in the YUV 4:2:0 format. Consequently, color blurring is prevented in the edges of the image, and an objective is to improve the saturation in the entire image. We based the proposed method on bilinear interpolation and bicubic interpolation, and formulated an interpolation method using the luminance component as an extended form. The proposed method and the bicubic interpolation are compared in both objective and subjective evaluations, and demonstrate the effectiveness of the proposed method. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 89(9): 1–10, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20218 [ABSTRACT FROM AUTHOR]

Generally, in image segmentation or object extraction, the edge image of the target object has a substantial effect on the extraction accuracy. In this paper, we focus on the characteristics of the currents flowing in an electrical circuit and propose a technique to interpolate and enhance the edges of a still image by simulating this circuit behavior and apply this technique to image segmentation. The proposed technique constructs a resistor circuit by mapping the edge intensity of each point in the edge image to a resistance, applies a suitable voltage to this circuit, and determines the distribution of the flowing current. Based on the property that the current concentrates in the path that minimizes the sum of the resistances, the current distribution that was obtained is inversely transformed again into the edge intensities in order to enhance weak edges and interpolate broken edges. In several experiments, the proposed technique produced better results than other edge interpolation and enhancement techniques. And when used with the watershed algorithm, highly accurate segmentation was obtained. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(9): 22–33, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10085 [ABSTRACT FROM AUTHOR]

The IDO method, known as a numerical method suitable for partial differential equations containing an advection term, is improved in terms of its accuracy. In the method proposed in this paper, the order of Hermite interpolation is increased with a parameter h in order to deal with the higher order differential coefficients as independent variables. (In the conventional IDO method, h = 3 for calculation of the advection term and h = 5 for calculation of the diffusion term.) When this method is applied to a one-dimensional advection-diffusion equation, it is found that the numerical error is reduced to about 1/5 of that in the conventional IDO method if h = 7 is used. By using two mutually overlapping grid systems, the scheme becomes stable. Further stability is achieved by applying the time-splitting method. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(8): 11–18, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10124 [ABSTRACT FROM AUTHOR]

Enlargement processing of digital images is usually carried out by means of an interpolation function obtained by simplification of the sinc function. However, in such an interpolation method, restoration of the high-frequency components lost in digitizing the images is not carried out at all. Hence, the enlargement causes blurring for images with many edges and detailed signals. For one-dimensional signals, Carrato and colleagues have proposed an interpolation method in which the discontinuities of the step edge signals are preserved while other signals are linearly interpolated. In this method, no blurring occurs in the expansion processing of strong frames of the images. However, Carrato's method has the following problems. (1) Only the step edges in the image signals are taken into account so that consideration of the detailed signals is insufficient. (2) Extension of the one-dimensional interpolation method given by Carrato to a two-dimensional interpolation method has a problem in interpolation at the center of the lattice. In this paper, a method is proposed to accomplish improvement of Carrato's method. With regard to problem (1), the interpolation not only is effective for the step edge signals but also preserves the apex of the peak and valley signals. With regard to problem (2), improvement is planned by directly deriving the interpolation value from the original signals. The proposed interpolation processing is accomplished by describing the processing sequence by a rule. The proposed method consists of two sets of fuzzy reasoning. The proposed method and the conventional interpolation method are compared from the numerical and visual perspectives and the effectiveness of the proposed method is clarified. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(3): 57–66, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.1140 [ABSTRACT FROM AUTHOR]

In secret-sharing schemes, a secret s is encoded as V1 Vn, and Vi is distributed to member Pi. By the access sets, we mean exactly those subsets of {Pi} which can recover the secret s. Previous secret-sharing have utilized the methods of polynomial interpolation [20] and projective geometry [6], among others. In this paper we show how to obtain secret-sharing schemes from combinatorial designs, including orthogonal arrays (QAs) and t - (v, k, 1) designs. [ABSTRACT FROM AUTHOR]

High-frequency responses of switched-capacitor filters deteriorated mostly as a result of the finite gain bandwidth product of an operational amplifier. The authors previously proposed a type of FIR and IIR switched-capacitor filter using a parallel cyclic-type circuit in order to improve high-frequency performance. In the parallel cyclic-type circuit, an operational amplifier is not used as a delay element. Instead, both delay and weighting operations are implemented with switches and a coefficient capacitor simultaneously. Only one operational amplifier is used for a summing operation. Consequently, the maximum operational frequency of the filter is extended up to the unity gain frequency of the operational amplifier. This paper proposes a new type of FIR and IIR switched-capacitor interpolation filter that can be operated in a high-frequency region by using the parallel cyclic-type circuit. Effects of both the finite gain band-width of the operational amplifier and parasitic capacitances on fitter performance are analyzed by applying the analysis method of periodically time-varying digital filter. These effects are found to be very few in number below the unity gain frequency of the operational amplifier. Third-order FIR and biquadratic IIR interpolation filters are constructed with discrete components and verified to have good frequency performance that coincides with the analyzed one. [ABSTRACT FROM AUTHOR]

The digital sinusoidal oscillator considered in this paper is one of the signal sources employed in the digital signal processing system. A construction of the oscillator is discussed where the memory capacity is reduced and the increasing rate of the noise is suppressed by interpolating the output of the oscillator of table look-up type or feedback type. A design method for the interpolation filter is presented first where no interpolation error is produced if the coefficient wordlength of the interpolation filter is infinite. Then, considering the case where the wordlength is kept finite, the hold-type sampling expander is proposed where the waveform distortion is kept low compared with the traditional zero-interpolation-type sampling rate expander. The distortion in the proposed method is analyzed where the wordlength of the interpolation filter coefficient is kept finite, and the waveform distortion is shown to be low. The increasing rate of the round-off error is examined by a computer simulation for the oscillator, which is obtained by applying the interpolation to the feedback-type oscillator. The effectiveness of the interpolation is demonstrated. Finally, a construction of the block-digital sinusoidal oscillator is shown with a smaller hardware complexity than in the traditional method, by constructing the block-structured oscillator by realizing the proposed interpolation filter as a polyphase filter. [ABSTRACT FROM AUTHOR]

This paper is aimed at the reduction of the hardware complexity and the computational complexity in the interpolation processing. A construction and design method for the interpolator is shown where only the coefficients of the power of two are employed. The interpolator is composed of the cascade connection of a passband compensatory filter, a zero-interpolator, and a stopband forming filter. Since no multiplier is used in the proposed construction, zero coefficient sensitivity as well as a high-speed operation are realized. FIR or IIR construction is used for the passband compensatory filter, and the stopband forming filter is constructed by a cascade connection of RRS. The design procedure for the interpolation is as follows. The linear programming is applied first to design the stopband forming filter. Then the passband compensatory filter is designed by the indirect exhaustive method or the least-square method. It is shown through design examples that the proposed interpolator is useful in reducing the hardware complexity and the computational complexity. [ABSTRACT FROM AUTHOR]

The spline interpolation using the sampling bases has a feature of real-time computation which is not observed in other methods since the result of interpolation is derived directly from the sampled value sequence. Then it is necessary that the sampling function should be truncated in a finite interval. This paper aims to clarify the relation in this method between the error in the result of interpolation due to the truncation and the length of the truncation interval. The result is summarizes in a correspondence table for the truncation interval length and the error. In the proposed interpolation method, the truncation interval length to ensure the given tolerable range of error can be determined using the correspondence table. [ABSTRACT FROM AUTHOR]

Assume that signal f(t) is impressed on N parallel time-invariant liner networks Hm(ω) (m = 1∼N), and consider the uniformly sampled values gm(nT) (m=1∼N; n=0, ± 1, ±2; T > 0) of the output signal gm(t) (m=1∼N). This paper discusses the following problem from a unified viewpoint. The response g(t), when the signal f(t) is passed through the given filter H(ω), is to be approximated by and expression y (t) which is the sum of the forementioned sample value multiplied by time functions ø mn(t) (m-1 ∼ N; n=0, ± 1, ± 2,). It is assumed that the signal f(t) belongs to the set 1 of the signals for which the weighted square integral of the Fourier spectrum F(ω) is not greater than a positive number A. In the foregoing, ømn(t) is called the interpolation function. First, it is shown that given Hm(ω)(m=1∼N), the time-limited interpolation function, which minimizes the upper limit emax(t) of the error e(t) =g(t) - y(t) over all f(t) belonging to T, is obtained by shifting the impulse responses øm(t) (m=1∼N) of certain N linear time-invariant interpolation filters øm (ω) (m=1∼N) and øm(ω) (m=1∼N) are optimized so that the measure emax(t) of the error is minimized, the upper and the lower bounds of the optimal emax(t) are shown. Considering an application where N is large, the optimization is discussed in the sense that the equivalent multiplicity is reduced at the sacrifice of the approximation error. [ABSTRACT FROM AUTHOR]

This paper shows that by taking every M th impulse response from the impulse respons- es of FIR low-pass digital filters and constructing a new digital filter based on them. a filter is obtained with low-pass characteristic such that the passband width is N times that of the original filter. Next, taking advantage of this characteristic, a method is proposed for designing a low-pass digital filters with variable cutoff frequency based on the same circuit by first designing a basic digital filter and varying 14, that is, changing over taps. This principle is applicable not only to FIR filters but similarly also to HR filters if some restrictions are allowed for denominator polynomials. Further, it is shown that by applying this theory to IFIR (Interpolated FIR) digital filters and multi rate digital filters, filters can be realized which are more effective with respect to their cutoff characteristic and operational points. [ABSTRACT FROM AUTHOR]

It is an important realization of digital filters to reduce the amount of hardware, notably multipliers. This paper proposes a cascade connection of IFIR (Interpolated FIR) filters by which a steep characteristic can be obtained with a small number of tmultipliers in most cases. The reduction of the multipliers is achieved by the determination of the transfer function of the overall cascade connection (not each element) by using the L- norm approximation. This method also has an advantage in that its central frequency can be determined arbitrarily. The cascade connection of several elements reduces the degrading of its characteristic caused by the quantized coefficients as shown in examples, since the conditions required for each element are relaxed in this structure. [ABSTRACT FROM AUTHOR]