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This paper describes a method of configuration of a complex RiCR filter with coupled Chebyshev characteristics. The frequency shift method is a well-known method of obtaining a complex filter from a real filter, in which the transfer characteristic of the real filter is shifted along the frequency axis. The complex coupled Chebyshev filter obtained by the frequency shifting of a coupled Chebyshev filter has an excellent attenuation characteristic, but the number of constituent elements for active realization is large and the circuit is complex. In the proposed method, the complex coupled Chebyshev filter is first transformed to a complex RiCR filter so that the number of constituent elements for active realization is reduced and the circuit is simplified. In this paper, a method is presented for transforming the complex coupled Chebyshev filter to a complex RiCR filter. The attenuation characteristics and the numbers of constituent elements of the complex filter designed by the proposed method, the complex coupled Chebyshev filter, and the conventional complex RiCR filter, are compared for confirmation of the effectiveness of the proposed method. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 89(5): 12–20, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20212 [ABSTRACT FROM AUTHOR]

This paper discusses codes, and a decoding method, that can correct compound errors containing a mix of random and burst errors. Specifically, the paper provides a minimum compound distance that represents a compound error correcting capacity for iterated codes, proposes a compound error correcting decoding algorithm for the code, and demonstrates that the decoding algorithm is capable of correcting compound errors up to half the minimum compound distance of the code. The proposed decoding algorithm is a generalized decoding method in which the decoding method for iterated codes indicated by Reddy and Robinson is a particular case. By this generalization, the proposed decoding algorithm can correct compound errors that cannot be corrected by conventional decoding methods. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 89(1): 60–72, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20035 [ABSTRACT FROM AUTHOR]

This paper proposes a relaxation algorithm for an inverse filter that is adaptive to environmental change in a multichannel sound reproduction system. For a conventional multichannel sound reproduction system, a design method has been proposed for an inverse filter by means of the Moore–Penrose-type generalized inverse matrix. However, in the inverse filter in the conventional method, background noise and line noise may be enhanced when the room transfer characteristics vary. In order to resolve this problem, an adaptive relaxation method based on the truncated singular value decomposition (TSVD) has been introduced. In the present paper, a microphone for monitoring the noise is placed at a location far from both ears of the listener. Relaxation of the inverse filter is carried out by using the reproduced sound observed by this microphone. From the results of computer simulation, it is found that the proposed method prevents amplification of the noise. Also, as a result of a subjective evaluation experiment on sound reproduction in a real acoustic environment, the sound quality was found to be improved significantly without damaging the sound localization of position. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(7): 15–26, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10159 [ABSTRACT FROM AUTHOR]

A novel design algorithm is proposed for two-dimensional linear-phase FIR digital filter approximating constant amplitude over the entire passband. The design technique proposed in this paper is an analytical method in which the filter coefficients can be obtained with noniterative numerical calculation. This design technique is based on the idea that the specified filter characteristics of the filters to be realized can be made out using the frequency response of a one-dimensional window function with well-known characteristics. In this paper, we present a number of design examples to illustrate the flexibility and effectiveness of the proposed technique as a comprehensive design technique. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(2): 43–58, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10033 [ABSTRACT FROM AUTHOR]

It is known that an IIR filter can provide an excellent amplitude characteristic that is of lower order than an FIR filter. Further, the IIR filter using all-pass circuits can realize an identical characteristic with fewer multipliers than in the conventional IIR filter. In this paper, a configuration and design method are given for a two-channel filter bank consisting of IIR filters with all-pass circuits. Lossless transforms and lossy transforms using these filter banks are proposed. One lossless transform is the S transform based on the Haar transform, but there is a problem that correlation remains in the output signals. In this paper, in place of the Haar transform in the S transform, a lossless transform based on two-channel integer IIR filter banks consisting of delay lines and all-pass circuits is proposed. On the other hand, the Haar transform is the only lossy transform based on the two-channel FIR filter bank that offers linear phase and orthogonality. Also, there are no design methods and no proof of effectiveness in applications of the two-channel IIR filter bank with both characteristics for image coding. In this paper, a two-channel orthogonal linear-phase IIR filter bank with a parallel structure of the all-pass circuits is proposed in order to maximize the coding gain, with the order of regularities as the constraint. Finally, the proposed methods are applied to stationary image coding and the effectiveness of the present method is shown. © 2002 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 85(7): 1–9, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10001 [ABSTRACT FROM AUTHOR]

In this paper we discuss a method for reducing the error propagation based on a combination of the error correcting code and the runlength limited code. When the runlength limited code and the error correcting code are used together, the runlength limited encoding is usually applied after the data have been encoded by the encoder of the error correcting code. However, error propagation generated in the data transmission channel can be substantially mitigated during the decoding of the runlength limited code if the runlength encoding is implemented first with subsequent implementation of the error correcting code. The method proposed in this paper consists of a periodic insertion of check symbols in the data sequence encoded first by the runlength limited code and then by the error correcting code. We also proposed a new runlength limited code suitable for the proposed combined method and evaluated its performance. © 2001 Scripta Technica, Electron Comm Jpn Pt 3, 84(11): 45–51, 2001 [ABSTRACT FROM AUTHOR]

Multicompensator configurations where multiple compensators are simultaneously in operation for a given plant have been widely used for performance or reliability of the total control system. The idea of cooperation among compensators in such control has not as yet been clarified. This paper presents a conception along with its mathematical model. Passive redundancy is a theory for achieving reliability so that the total control system is still stable even if some compensators fail. Based on this theory, this paper introduces a new idea of advanced cooperative order in a compensator set from the viewpoint of sharing loads with each other, where all compensators must contribute to reduction of any other's load. That is, each compensator has a positive definition of membership in the compensator set. It is shown that this advanced cooperative order can be achieved. © 1997 Scripta Technica, Inc. Electron Comm Jpn Pt 3, 80(10): 9–18, 1997 [ABSTRACT FROM AUTHOR]

The adaptive line embracer eliminates noise from a sinusoidal signal with the unknown frequency buried in the noise and extracts the sinusoidal signal. The performance of the adaptive line embracer is represented by the signal-to-noise improvement figure. The signal-to-noise improvement figure increases as the bandwidth of the adaptive line figure increases is made narrower. The convention adaptive line enhancer, however, contains an incompatible problem in that the convergence of the adaptive coefficient is deteriorated when the signal-to-noise improvement figure is by narrowing the bandwidth. This paper proposes the adaptive line enhancer with the discriminated structure to solve this problem. The adaptive line enhancer with the discriminated structure is composed of two parts. One is the (frequency) estimator that estimates the frequency of the unknown sinusoidal signal, and the other is the (sinusoidal signal) discriminator that discriminates the unknown sinusoidal signal from the noise. The estimator is the conventional adaptive line enhancer, and the bandwidth is determined so that the convergence of the adaptive coefficient is the fastest. The discriminator is the narrowband bandpass filter where the center frequency is controlled by the adaptation coefficient. This paper the optimal design method for each of those paths, and the effectiveness of the method us verified by a computer simulation. [ABSTRACT FROM AUTHOR]

This paper proposes a narrowband filter, which is a finite impulse response (FIR) filter where the phase response exhibits a constant phase-linear characteristic, independently of the center frequency, and the execution format is a sum of stable recursive filters. The optimal design of the differentiator (differentiation filter) using that narrowband filter is discussed. To realize a sharp narrowband characteristics, higher-order FIR filtersare needed The narrowband filter proposed in this paper has the feature that it is the sum of the recursive filters, and the operation can be executed with the number of operations which is independent of the order. The filter considered is obtained as a linear combination of a number of narrowband filters with distributed center frequencies. By connecting the first-order sum or difference filters in series to the above filter, a filter can be constructed which has an arbitrary amplitude response and approximately constant linear-phase response. Especially, the optimal design procedure is given for the set of bandpass differentiator filters and application examples are shown for the estimation of the transfer function of the linear system. [ABSTRACT FROM AUTHOR]

This paper considers the complex coefficient transfer function with transmission zeros on the imaginary axis and presents a method to realize such a transfer function by a reactance network including imaginary resistors with double resistor termination. Using the obtained network and the reference network for the complex-wave digital filter, the complex digital filter with transmission zeros on the imaginary axis can easily be extended, which is important in practice. In this paper, Brune's method modified by Belevitch as well as Fujisawa's method extended by Morimoto et. al. are improved and the realizability condition is relaxed. Then, several construction examples are shown an the features of construction methods are compared and discussed. [ABSTRACT FROM AUTHOR]

This paper proposes the ADPCM speech coding system using mel-cepstrum as the parameter for the short-term prediction. In the proposed coding system, the backward type adaptive prediction is applied based on the adaptive mel-cepstrum analysis. Operations such as noise shaping and postfiltering are executed through the mel-cepstrum. Consequently, it is expected that the noise shaping and the postfiltering have the effect which is matched to the human auditory characteristics. This paper presents the configuration of the coding system with some improvements such as the addition of the pitch predictor, and the quality of the coded speech is evaluated. It is shown as a result that the proposed coding system realizes the same speech quality at 16 kit/s as that of the 32-kbit/s ADPCM, even though there is no algorithm delay. [ABSTRACT FROM AUTHOR]

It is known that the quadrature mirror filter (QMF) bank used for subband coding cannot avoid the aliasing effect for adaptive signal processing or spectral estimation, etc. A new class of filter banks is desired for these applications where the effect of aliasing also be considered. In this paper, a new class of filter bank is reported which has alias-free characteristics at equally spaced frequency points. This filter bank is called frequency sampling filter bank, and it consists of nonrecursive FIR filters. This paper also reports design methods for two types of the filter banks, keeping in mind the reduction in amount of calculation for practical implementation. In one method, all channel filters of the filter bank are determined by complex exponential modulation of a prototype filter. The other method determines the channel filters by cosine or sine modulation of a prototype filter. Both designs have the forementioned aliasing characteristics, and an efficient structure based on polyphase configuration is possible. [ABSTRACT FROM AUTHOR]

The information extraction techniques which transform the description at the logic circuit level in the LSI design process into the description at the functional level are important. The authors have been studying the automatic transformation technique. By combining that technique with the technique for automatic logic synthesis which is now being developed, it is expected that the descriptions at the logic circuit level and the functional description can be transformed to each other. This will help to construct the integrated CAD system. The feature of the method proposed in this paper is that the arithmetic operation functions and the functions at the RT level can be extracted from the tremendous amount of complex circuit descriptions and the results are represented in a simple and clear form. Consequently, the method is especially suited to the behavior verification of the circuits, such as CPUs. In this paper, the object of consideration is limited to the circuits in the CPU system and the application of the functional information extraction technique to the design verification is discussed. Since the behavior description for a CPU is given in a simple and clear form, using the information code table or state transition table, the design verification is easy by a visual comparison of the extracted functional description. The complex logic functions in the control circuit are compared using BDD. An actual example of verification is shown in this paper for the 8-bit microprocessor, and it is demonstrated that a practical processing is realized from the viewpoints of required time and memory capacity. [ABSTRACT FROM AUTHOR]

A new design method for two-dimensional allpass filters to approximate the desired phase characteristic is proposed in this paper. Considering the denominator polynomial of the allpass transfer function, the design method via Hilbert transform is expanded to a two-dimensional ease with complex coefficients. When the Hilbert transform is used, a problem arises in that sometimes truncating coefficients cause the filter to be unstable. In this paper, in an unstable ease, the resultant frequency response is approximated as a low-order polynomial by using the Lagrange interpolation technique and the same procedures are repeated. Therefore, this proposed method ensures the stability of the designed filter because the coefficients of order higher than the desired order are small enough to be almost zero. [ABSTRACT FROM AUTHOR]

This paper considers digital filters with hypercomplex coefficients. Filters of this type can reduce the order of filters to one-fourth the one with real coefficients. It is reported that the first-order digital filter with hypercomplex coefficients can realize the fourth-order digital filter with real coefficients which can be expressed by a second-order all-pass filter with complex coefficients. This paper considers also generalization including the foregoing special case and shows that the first-order digital filter with hypercomplex coefficients can realize an arbitrary-order digital filter with real coefficients less than 4. Arbitrary higher-order digital filter with real coefficients can be realized by parallel connection or cascade connection of first-order digital filters with hypercomplex coefficients. This paper considers the following three different realizations: (a) realization by parallel connection: (b) realization by cascade connection of real part; and (c) realization by cascade connection. We find that case (b) is the most low sensitive. Finally, stability criteria of hypercomplex digital filters of first-order are suggested. [ABSTRACT FROM AUTHOR]

This paper defines anew the maximum two-route flow as a measure to represent the relation between two vertices in the flow network. The maximum two-route flow corresponds to the maximum number of communication channels that are composed of two-route paths between two terminals in a communication network. The F-rings that can be composed simultaneously, containing the considered two vertices, is defined as the maximum two-route flow between those two vertices. It is shown in this paper that a theorem exists for the maximum two-route flow similar to the max-flow min-cut theorem for the ordinary flow. [ABSTRACT FROM AUTHOR]

Slepian and Wolf considered a coding system (called SW system) for coding the multiterminal (information) source. This paper extends the SW system and defines anew a system (called SWL system) where encoders are mutually links. The problem of determining the optimal error exponent in the admissible rate region has not been answered completely for the SW system. It is shown in this paper that this problem can completely be solved for the SWL system. It is shown also that the optimal exponent can be achieved by the universal code. It is shown further that even if the encoders are linked, the admissible rate region is not enlarged and the reliability of the information transmission is not improved when the transmission rate is outside the admissible rate region. It is interesting to note that those results correspond in a sense to those for the channel with feedback. [ABSTRACT FROM AUTHOR]

Suppose that a biconnected planar graph G = (V, E) is given such that each vertex and edge are weighted. This paper proposes an algorithm for finding a planar embedding such that the total sum of weights of vertices and edges on the exterior window is maximized (among all possible planar embeddings of G). Among algorithms for determining planarity of a graph there is one that uses a data structure called a PQ tree. The algorithm presented in this paper is the one which extends it and uses the PQ tree as its data structure as well. However, it can find a required planar embedding in O(¦V¦) time by assigning special weights to each node of each PQ tree generated in the algorithm. [ABSTRACT FROM AUTHOR]

This paper proposes a method for precisely estimating sinusoidal frequency using five samples. The estimation procedure is implemented by interpolating the impulse location between DFT results. Five samples are used because two complex components are available in the DFT results. However, an estimation method which uses four samples and is called Prony's approximation method is known in the literature. To derive interpolation formulas for six or more samples is more complex. In this paper, strict estimation formulas and their simplified versions arc deduced from the DFTs of five samples. The frequency estimation error is also discussed with a computer-simulated experiment which shows a I percent r.m.s. error at a signal-to-noise ratio of 30 dB. This method may be applicable for representing frequency in a direct binary form which can be processed by simple operations such as addition, exclusive-OR, and sign extractions form the calculated value. [ABSTRACT FROM AUTHOR]

This paper proposes a method of realizing an adaptive normalized ARMA lattice filter. This filter was proposed in a previous paper and can be realized with an arbitrary ARM A order and with all the lattice coefficients less than one. Furthermore, these coefficients are equal, so the structure is simplified. The adaptive realization method proposed in this paper retains these features. In addition, we perform spectral estimation of a time-varying signal with frequency weighting and model identification of an unknown input system. In our paper we describe an experiment with a real speech signal to prove that our algorithm can be used for frequency-weighted spectral estimation. The use of real data to perform spectral estimation has not been attempted in other references. [ABSTRACT FROM AUTHOR]

This paper considers the problem of an optimum linear filter on the receiving side of the PPM transmission system in which both the additive noise and the signal-dependent noise exist. The optimum filter for the PPM transmission derived in this paper is the one in which the mean-square value (MSE) of the timing error is minimized at the location of the pulse detection. In contrast to the optimum filter in the PAM transmission which maximizes the signal-to-noise ratio in the amplitude direction, the present optimum filter is of a different type which maximizes this ratio in the time-axis direction. In this paper, the general equation which the optimum filter must satisfy is derived as an equation containing the previous research results. Subsequently, the timing error, optimum pulse detection level and the configurations of the filter on the receiving side and of the pulse location detector are presented. It is pointed out that, by the use of the optimum filter derived in this paper, baseband multiplexed transmission systems using pulses can be realized easily by combining the PAM and PPM systems. The method for multiplexing and the receiver configuration are shown. [ABSTRACT FROM AUTHOR]

So far, three periodically time-varying models have been proposed in the research on periodically time-varying digital filters. They are the coefficients time-varying model, the output sampling polyphase model, and the input sampling polyphase model. Analyses of these models have also been completed by the authors. This paper proposes new models to solve a problem which the three conventional models cannot solve. It is known that a periodically time-varying digital filter can be used to implement a spectrum scrambler. However, when these models are to be used to implement a spectrum scrambler, time-invariant filters in these models are restricted to FIR filters. This paper proposes and analyzes an output sampling polyphase model and an input sampling polyphase model, both of which contain a discrete Fourier transformer. Relationships between the newly proposed models and the conventional coefficients time-varying model are established. BY making use of the newly proposed periodically time-varying models, we can design a spectrum scrambler employing not only FIR time-invariant filters but IIR time-invariant filters as well. Furthermore, compared to the conventional models, the newly proposed periodically time-varying models are shown to have an advantage of less amount of computation when they are used to implement a spectrum scrambler. [ABSTRACT FROM AUTHOR]

This paper proposes a technique of functional Information extraction which transforms design description at the logic circuit level in the LSI design process to a description at functional level. This technology is considered to be a powerful means of logic design verification similar to the circuit extraction technology already in practical use for layout design verification. Moreover, by automatically generating function description language from given circuit description, it can be used in functional and fault simulations and is useful in reusing circuits. An extraction technique of logic and arithmetic functions of combinational circuits is investigated herein. As internal representation of logic functions, binary decision diagrams are used for efficient treatment. For extraction of arithmetic functions, a technique of first extracting logic functions, and then representing them by arithmetic expressions is used. This technique has a characteristic that the sane function description is obtained from circuits with the same functions even though their circuit structures are different. Therefore, functional information extraction independent of circuit structure can be done. This paper describes first the extraction of logic functions, which is the basic technique, and then explains the arithmetic function extraction technique. Finally, some concrete examples are evaluated. [ABSTRACT FROM AUTHOR]

This paper considers the 2-fl separable-denominator digital filter, and discusses its real-time and parallel-processing-oriented realization. The aim of this paper is to employ a large number th processors concurrently, and to realize a real-time processing minimizing the data throughput delay by applying pipeline processing. More precisely, the case where the poles of the 2-D separable- denominator transfer function are composed of real and complex poles is considered, and two new methods of realization are presented based on the Roesser-type local state-space model, which is suited to the real-time processing. It is shown that the data throughput delay in those methods is equal to the time for two multiplications and one addition, or the time for one multiplication and two additions. The processor factor and the efficiency in each realization are investigated, and the effectiveness of the realization methods are verified from the viewpoints of real-time processing and paralle processing. [ABSTRACT FROM AUTHOR]

This paper proposes a new design method for the filter bank. The filter bank is a system which can reconstruct a signal via band-splitting, decimation, interpolation and restoration. It is employed widely to solve various problems in the communication system, and various methods have been presented for its specification, implementation and design. This paper discusses the approximate realization of the filter bank, which is called the M-channel uniform-splitting FIR perfect-reconstruction filter bank, and its design principle is as follows. First, the analysis filter is designed so that each channel has the characteristic which is the translation of the given LPF on the frequency axis. Then the synthesis filter is designed in the time-domain so that the unit impulse response of the whole system is equivalent to the unit pulse, i.e., that the response of the linear-phase all-pass transmission characteristic. The Hilbert transformer plays the important role in the translation of the characteristic. Compared with the conventional design, the proposed method has the merit that no constraint is imposed on the band splitting and the band is partitioned uniformly with an arbitrary number of divisions. Design examples are shown to indicate the usefulness of the proposed design. [ABSTRACT FROM AUTHOR]

If the signal space matched to the representation is selected according to the properties of the signal to be handled, the signal processing can be made efficient. this paper is a step toward the structuralization of the signal spaces for systematic selection of the signal space. As a particular problem, the relation is analyzed between the signal spaces composed of spline functions, which have the feature that the smoothness as the continuous differentiability varies according to the degree, and the band-limited signal spaces, which is fundamental in the signal processing. As a result of analysis, it. is shown that the signal spaces composed of spline functions agree with the band-limited signal spaces in the limit of infinite order. This implies that the signal spaces composed of spline functions and the band-limited signal spaces can be structured as a sequence with the degree as the parameter. The result in this paper provides a view that the continuous differentiability can be used as an index in the systematic selection of the signal spaces. [ABSTRACT FROM AUTHOR]

Recently, for signal processing such as echo cancellation and equalization it is required to develop IIR-adaptive digital filters. This paper proposes a new-type IIR- adaptive algorithm based on the orthogonalization method, considers the convergence of the algorithm, and verifies its effectiveness by simulation. White's algorithm is one of IIR-adaptive algorithms reported thus far and Feintuch's algorithm is known as its approximation algorithm. This paper also derives an IIR-adaptive algorithm using both Feintuch's algorithm and the orthogonalization method, First, it can be shown that a generalized algorithm (IIR-LI) of IIR-learning identification method based on previous heuristic methods can be derived using the idea of Feintuch's algorithm and the conjugate gradient method. Next, a new IIR-adaptive algorithm (IIR-AACG) based on the orthogonalization principle of the conjugate gradient method is presented. Further, by applying the orthogonalization method to a past input vector sequence we derive an IIR-Affine projection algorithm (1IR-AP). Finally, these algorithms are compared by computer simulation to verify the effectiveness of the orthogonalization method. [ABSTRACT FROM AUTHOR]

This paper aims for high reliability as well as reduction of hardware size, cost and development period for the time-division multiplex access (TDMA) equipment, and proposes a new TDMA equipment architecture and modular construction. The fundamental approach employed in this paper is the digital realization of the analog circuit and the LSI implementation of the digital circuit. Then to realize TDMA equipment with a simple architecture and to realize a widely applicable LSI, TDMA equipment is reconstructed into a hardware-oriented architecture. Based on the new architecture and the optimal function assignment to the modules, the following LSI and IC were developed. Six types for synchronization control, 11 types of LSI and IC for modulation-demodulation (3 LSI, 4 MIC and 4 HIC), and 1 type of terrestrial network interface. By this development of LSI and IC, the hardware scale of TDMA equipment was reduced to one-fifth, and the development period is reduced to one-half compared with previous equipment. The maintenance-free property is also achieved. The developed LSI and IC can be applied to various TDMA equipment in general with different TDMA frame and rate. For TDMA equipment to be developed in the future, the hardware-scale and the development period can be reduced as in this case. [ABSTRACT FROM AUTHOR]

This paper proposes a new construction method for 2-D IIR digital filters with all-pass subfilters as components. Recently, numerous studies have been made to construct digital filters by combining all-pass characteristics. This paper is based on the construction by a parallel connection of an all- pass subfilter and delay element, which can realize approximately the linear phase characteristic. First, the basic design for 1-D digital filters as well as its properties are described. Then the theory for the 1-D case is extended to the 2-D case, and a new construction method for 2-D digital filters with low-pass characteristics with circular symmetry is shown. The proposed method features the following. The phase characteristic is approximately linear in the passband. The filter can be realized with a smaller number of multipliers than in the traditional method. The coefficient sensitivity is low. [ABSTRACT FROM AUTHOR]

The design method entitled "robust model matching" was recently proposed for the design of practical robust control systems. By this method, if the overall system is stable, any desired robustness in the steady-state and transient characteristics can be guaranteed in spite of parameter variations for minimum phase plants. However, when using this method it is always necessary to employ a trial-and-error process to ascertain stability. in this paper a robust model matching design method which guarantees stability is proposed for the case when the parameter variations are such that the plant remains a minimum-phase plant without a change in the sign of the gain. Moreover, only the controlled variable and the operating quantity are measurable. By this approach) any desired robustness in the steady-state and transient characteristics can be achieved while guaranteeing robust stability without the necessity of a trial and error process. In this paper, first the necessary conditions a robust filter must fulfill to ensure stability are presented. Next, the degree relationships that certain polynomials concerning the model matching controlled must fulfill to design a robust filter are clarified. Further, it is proven that control system stability is ensured by the design of an appropriate robust filter without narrowing the class of realizable reference models. Finally, the proposed design method is set forth and an example design problem is given. [ABSTRACT FROM AUTHOR]

In the ultrasonic pulses-echo measurement system, the deconvolution signal processing enables maximum information to be extracted from the echo signal under the physical restriction of the acoustic propagation loss, thereby realizing significant improvement of the resolution capability. This paper presents the deconvolution filtering technique which satisfies the practical requirements for the mitigation of noise amplification compatible with a simple fast procedure. Specifically, the time domain deconvolution filter is determined in such a way that the second-order moment of the desired response is minimized under the band-limited condition, thereby optimizing the filter coefficient subject to the constraints of the small processing noise. In addition, the envelope detection is incorporated for the removal of the distortion due to the band-limited nature of the signal. Furthermore, the theoretical formula for the range resolutions achievable with this technique is derived. At the end of this paper, results of the evaluation experiments are shown to demonstrate that high-resolution measurement comparable to the theoretical limit is obtained with small processing noise. [ABSTRACT FROM AUTHOR]

This paper proposes a new method of constructing the multirate FIR digital filter with arbitrary passband, using real-coefficient filters for the decimator and interpolator composing the multirate filter. It is already known that the bandpass filter can easily be realized using the complex-coefficient filter. In such a construction, however, the processing of the analytic signal is required using Hubert transform, which makes the whole system somewhat complex. From such a viewpoint, this paper presents a method of constructing the multirate bandpass digital filter using the real-coefficient filter. Another method is also presented, where the multirate lowpass filter with arbitrary passband can be constructed by combining the complementary response and a lowpass filter. The multirate method is effective in realizing a narrowband response, but may not be useful in realizing a relatively wideband response. By the proposed method, the latter can easily be realized drastically reducing the number of multiplications. Similar methods can be applied to highpass and bandstop filters. [ABSTRACT FROM AUTHOR]

This paper studies the analysis and minimization of roundoff noise in separable denominator multidimensional (SD M-D) digital filters on the basis of state-space representations. First, the roundoff noise in SD M-D digital filters is analyzed and the !2-norm dynamic range constraint is discussed. Using the results, a synthesis method of optimal realizations (with respect to roundoff noise) is proposed. Further, the simple relation between balanced realizations and optimal realizations of SD M-D digital filters is obtained and an efficient method for synthesizing optimal realizations from balanced realizations is proposed. In addition, this paper also proves that optimal realizations are free of overflow oscillations under zero input conditions. Finally, the synthesis procedure is illustrated by a numerical example. [ABSTRACT FROM AUTHOR]

It is known that a leapfrog-type digital filter obtained by simulating an LC ladder filter terminated with resistors on both ends is a digital filter with a low coefficient sensitivity. In this paper, an analog circuit to be simulated is chosen as a distributed circuit in which the real frequency and the discrete variable correspond rigorously. By means of an equivalent transformation, a leapfrog-type digital filter is constructed in which no delay free loop is generated. In this method, an LC ladder circuit expressed in terms of the Richard variable p is transformed to the circuit expressed in terms of the WI variable y. The digital filter is configured by stimulation of the relationship between the node voltage and series branch current in each element of LC ladder circuit. However, if the configuration is based on general distributed LC ladder circuits, a delay-free loop is always generated and configuration becomes unstable. As a solution to this problem, unit elements are introduced in this paper. It is shown that a realizable signal-flow graph can always be obtained if stub circuits are used as the original circuit. The basic section necessary to circuit configuration is presented. Finally, as a practical example, the present method is used for design of a fourth-order bandpass filter. The results are described and compared with those by other configurations. [ABSTRACT FROM AUTHOR]

The modeling of computation using logic circuits occupies an important position in the fundamentals of complexity theory including quantum complexity theory. Consequently, research into methods for computing logic functions on quantum circuits as well as for minimizing and simplifying such circuits has become extremely important. In this paper we explicitly formulate the depth minimization problem for quantum logic circuits and show that this problem is closely related to a geometric approach to deriving a lower bound on the size of a quantum logic circuit. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(10): 9–17, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20298 [ABSTRACT FROM AUTHOR]

In H.264/MPEG-4 AVC a wide range of different prediction block forms and prediction signal generation methods are available. By selecting an optimal combination of coding modes from the multiple possible combinations of such modes, improvements in coding efficiency can be achieved. However, when using the rate-distortion optimized coding mode decision method based on the Lagrange multipliers, at the same time as seeing a significant improvement in coding efficiency, we are faced with the problem that the mode decision procedure becomes extremely demanding computationally. The H.264/MPEG-4 AVC high profile introduces adaptive block size transformations thereby making the number of combinations of coding mode that can be selected even larger than under the main profile. In this paper we investigate a method for hierarchically and adaptively reducing the number of mode combinations. Specifically we propose a method for quickly deciding the coding mode while limiting the reductions in coding efficiency by the correlation information between two mode decision cost functions in accordance with a quantization parameter. The results of experiments confirm that by using the proposed method, the encoding time excluding the motion search can be reduced by up to 4 times for the main profile and by up to 7 times for the high profile as compared to rate-distortion optimized coding mode decision. © 2007 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(9): 41– 55, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20342 [ABSTRACT FROM AUTHOR]

The construction of an infinite number of nonstabilizer codes of Clifford type was given for the first time by the authors. However, the method of construction was restricted to encoding from odd prime p-level quantum states to 2p-level states. In this paper, the result is extended to the construction of nonstabilizer-type quantum codes which encode quantum states of arbitrary odd 2j+1-level to quantum states of 2h(2j+1)-level, where h are mutually prime 2j+1. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 90(4): 63 – 68, 2007; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20222 [ABSTRACT FROM AUTHOR]

This paper describes a wideband speech coding method that is robust against packet loss, and a packet transmission control method. The coding is based on 4-kHz-band CELP speech coding and consists mainly of multilevel coding using MDCT transform encoding, with a speech quality enhancement codec for the nonspeech portion and a higher subband enhancement codec for coding the higher subband at or above 4 kHz. This method can provide high-quality coding of nonspeech signals such as background music as well as wideband speech. If this coding method and packet transmission control are combined, speech transmission can continue without much deterioration even when the network becomes congested and packet loss occurs. Subjective and objective testing confirmed that, unlike conventional coding methods, high-quality speech transmission was possible even with 10% random packet loss when this method is used for speech transmission on an IP network. © 2006 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 89(12): 20–30, 2006; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20291 [ABSTRACT FROM AUTHOR]

Recently, approximation problems for FIR digital filters in the complex domain have been studied extensively. This class of problems minimizes the absolute value of the complex error of the filter studied and the desired response. Hence, the problem is to approximate amplitude and phase. However, in the waveform transmission, the group delay to indicate the delay of the wave is more important than the phase characteristic. To date, the phase approximation has been used as an indirect approximation of the group delay. Hence, the designed group delay characteristic is substantially degraded from the desired group delay. In the present paper, the group delay error is included in the performance function as a performance parameter of the least squares method. By deriving the least square error with repeated transformation of the desired response, a direct approximation of the group delay is realized and both the amplitude and the group delay are optimized. Finally, by numerical simulation, the method is compared with the conventional ones and the effectiveness of the present method is demonstrated. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(11): 20–27, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20169 [ABSTRACT FROM AUTHOR]

This paper considers chirp pulse microwave CT (CP-MCT), in which the chirp pulse and signal processing technology are used to estimate the internal structure and temperature variation of biological objects. A method is developed in which the CT image to be acquired by actual measurement can be derived by numerical computation. The aftereffect function between the transmitting and receiving antennas is calculated by a Gaussian pulse and the FDTD method. After convolution of the function with the input chirp pulse signal on the time axis, the measured signal at a point is constructed by the same signal processing as in real measurement. This computation is repeated for each point on the translation-scanning axis. The projection data are acquired and the CT image is generated. The validity of this computation procedure, which is called the aftereffect function method, is demonstrated by comparing the resolution and the measured temperature variation in actual measurements with the results of a simulation computation. Using the aftereffect function method, the point-spread function of an imaging system which is difficult to measure, can be derived. An analytical model for the human head is constructed. Simulations are performed for the attenuation distribution and the temperature variation distribution. Based on the results, the feasibility of the biological imaging by CP-MCT is discussed. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(9): 53–63, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20190 [ABSTRACT FROM AUTHOR]

In this paper, we employ a technique that is used to recognize a material from the reflected ultrasonic waves and propose a category recognition system using two ultrasonic sensors. In this system, the reflected waveforms are handled as two-dimensional images, and which category an object belongs to is instantaneously recognized by pattern matching to reference data by a combinatorial logic circuit. In this system, combinations of information such as the material, angle, and distance of the object are set as the recognition categories. The pattern matching is carried out by a combinatorial logic circuit created directly from the reference data. The recognition experiments with measured waveform data performed according to the proposed method demonstrated the features of the proposed system and its effectiveness when applied to recognition under specific conditions. With five materials set as the categories, the results show that a recognition rate of nearly 100% was obtained, and high recognition results were obtained for almost all of the materials when the categories were extended to combinations of the material, angle, and distance. © 2005 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 88(7): 33–42, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.20147 [ABSTRACT FROM AUTHOR]

This paper describes a systematic numerical analysis method for hybrid networks consisting of lumped and distributed elements. The lumped element circuit portion consists of linear and nonlinear elements while the distributed circuit portion consists of transmission lines with loss terms that have frequency dependence. The shape of the transmission line is not only uniform but also may be nonuniform due to limitations in implementation. The power supply is not only the independent source but may also be a distributed source induced by an external incident electromagnetic field. The transmission line section is formulated in the complex frequency domain (s domain) by means of the Laplace transform. If the lumped element circuit section consists only of linear time-invariant elements, then the formulation is by the state equation derived from the telegrapher's equation with the voltage and current vectors on the transmission line as the variables. For the analysis of the entire circuit, the numerical inverse Laplace transform is applied to the results obtained in the s domain. If the lumped element circuit section contains nonlinear elements, then the transmission line section is formulated by transforming the results based on the scattering matrix with the incident voltage and reflected voltage vectors at both ends of the transmission line in the s domain into the time domain. The analysis of the entire circuit is carried out in the time domain by discretization. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(7): 1–14, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10160 [ABSTRACT FROM AUTHOR]

The authors propose a method to reproduce surface textures by mapping images used in shape restoration as textures when generating a three-dimensional computer graphic image after restoring a three-dimensional shape from multi-viewpoint data of a real-world model. In the past, an image used as a texture had to be captured under the same lighting conditions and at the same spatial resolution as its source. In this paper, the authors consider a method to select and correct an optimal image from among images captured using free camerawork, and then use it for mapping. In other words, the authors take into consideration the distance between the patch surface and points of capture, while at the same time comparing the orientation of the patch surfaces which comprise a three-dimensional object and the orientation of the camera at the point from which it takes pictures. As a result, optimal pictures can be selected even when the distance to the subject or the position of the camera changes. In addition, by matching the pixel values distributed along the shared boundaries among adjacent patch surfaces, errors that may exist in the mapping between the patch surface and texture image can be corrected. Using their method, the authors perform verification experiments on a three-dimensional shape obtained using the factorization method and achieve good results. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(5): 70–81, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10173 [ABSTRACT FROM AUTHOR]

The authors describe a technique for selectively sharpening only edges that were blurred due to various causes without making interference factors such as random noise more conspicuous. The selective sharpening technique described in this paper is implemented as an iterative updating nonlinear algorithm obtained by discretizing a type of nonlinear reaction-diffusion time-evolution equations. The update terms of this algorithm consist of a quadratic nonlinear smoothing term, reaction term, and overshooting term. By performing quantitative evaluation experiments, the authors showed that the proposed technique has significantly better selective sharpness enhancement capabilities than existing sharpness enhancement techniques such as the peaking method. In addition, the authors introduced a processing parameter adaptive control method in the proposed technique and applied it to the suppression of breathing distortions in video sequences. Breathing distortions, which are video sequence distortions that often appear in old films, are time-varying blurring distortions that occur due to temporal variations of the focus. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 87(5): 33–47, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10086 [ABSTRACT FROM AUTHOR]

With recent advances in digital technology, it is now easy to copy digital content. As a result, unauthorized copying of content is widespread and copyright protection has become a serious issue. One solution to this problem is electronic watermarking, for which various techniques have been proposed. However, since no method of evaluation of the robustness of electronic watermarking methods has been established, it is difficult to compare various methods. This paper presents an analysis of the correlation-based electronic watermarking method in which a pseudo-random sequence is embedded as a watermark into the transform coefficients of the image and watermarking is detected by correlation. A method of evaluation is also proposed. Based on the proposed method, an electronic watermarking method using discrete wavelet transforms is proposed. Finally, based on the results, methods for improving the performance of the watermarking method are discussed. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(11): 1–12, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10093 [ABSTRACT FROM AUTHOR]

If there is a frequency offset in the local oscillator circuit in QPSK demodulation, the BER is degraded. Previously, methods for estimating this frequency offset have been proposed. However, few techniques that have been studied can estimate this offset in an environment with a large frequency offset or noise. A digital adaptive IIR notch filter is effective for estimation of a large frequency offset in an environment with noise. However, the noise is colored in the receiver due to the IF band BPF inserted in order to suppress out-of-band noise. Therefore, the converged value has a bias error with respect to the target value. The method proposed in this paper is based on the fact that the phase of the digital IIR-BPF is zero at the center frequency of the amplitude characteristic. This characteristic is used for control by applying a negative feedback to the digital IIR-BPF coefficients. Then, no bias error is generated because the noise in the in-phase channel and that of the quadrature channel in QPSK are uncorrelated. Further, the proposed method does not drift but maintains the present status due to the uncorrelated character described above, even in the case of channel disconnection such as shadowing. Therefore, this is a frequency offset estimation method suitable for mobile communications. As a result of evaluation of the proposed method in terms of the BER in a white Gaussian noise channel, it is confirmed that the proposed method provides a BER lower than that of the conventional digital adaptive IIR notch filter. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(9): 1–12, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10089 [ABSTRACT FROM AUTHOR]

Electronic watermarking has drawn attention as a way to prevent or limit unauthorized copying of digital contents and has been studied and improved extensively. In this paper, as an improved version of the electronic watermarking method proposed by Cox, a method is proposed in which a pseudorandom number sequence is embedded as watermarking information into the wavelet coefficients of the image. In this method, two conventional schemes known as improvements of Cox's method are included. With regard to this electronic watermarking scheme, the detection accuracy of the watermarking information and the quality of the image with watermarking are analyzed. It is shown that the detection accuracy of the watermarking information is better than in the conventional scheme. This conclusion is verified by numerical experiments. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(8): 31–40, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10036 [ABSTRACT FROM AUTHOR]

Enlargement of digital images is equivalent to narrowing the sampling spacing. Hence, it is necessary for enlargement of the digital images to estimate the high-frequency components lost in the sampling by some means. Recently, an enlargement method accompanied by estimation of lost high-frequency components has been proposed. The present authors have also noted the Laplacian pyramid layer expression, which is a multiresolution representation of digital images, and have proposed an enlargement method to estimate lost high-frequency components. Based on the Laplacian pyramid layered representation, estimation of unknown high-frequency components is equivalent to estimation of unknown high-resolution Laplacian components. Since the layered expression of the Laplacian pyramid has band division encoding as the objective, the image size between the layers is 2 and the enlargement method based on this expression is also limited to 2. However, it is possible to obtain a layered expression with an arbitrary size ratio by varying the passband width of the Gaussian filter and eliminating the pixels corresponding to the bandwidth, in place of continuous use of the half-band Gaussian filter and downsampling of the usual Laplacian pyramid layered image. In this paper, based on this concept, an enlargement method with high-frequency component estimation that allows setting of an arbitrary enlargement rate is proposed. The effectiveness of the proposed method is clarified from abundant application examples. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(8): 41–51, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10018 [ABSTRACT FROM AUTHOR]

In this paper, a design method is proposed for a two-dimensional separable-denominator state-space digital filter based on the projection method. First, the square error function is expressed in terms of the coefficient matrix of the local state-space model. Next, the given FIR filter is realized by a local state-space model. An equivalent transform based on the eigenvalue–eigenvector decomposition of the impulse response Grammian is applied to this filter. The obtained model is approximated at lower order by orthogonal projection. The projection parameters are optimized by an iterative algorithm based on the error function, deriving a lower dimensional model with a higher design accuracy. The conventional lower dimensional approximation algorithm based on rounding down of the coefficient matrix is used to find the initial values in the present method. The stability of the filter obtained by the present method is always guaranteed. The effectiveness of the present method is confirmed by numerical examples. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(6): 10–18, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10066 [ABSTRACT FROM AUTHOR]

Electronic watermarking technology has attracted substantial attention recently as a way to provide protection of copyright and prevention of unauthorized copying of digital content. This is realized by embedding copyright information within the content in a form not perceivable by humans. In this paper, an electronic watermarking method is proposed in which the watermarking information is embedded in binary patterns such as characters and figures by making use of morphological signal processing. This method is applied to alphabet characters to show the robustness of the proposed method to noise and printing. It is found possible to realize electronic watermarking by the proposed method for sentences in Western languages. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 3, 86(6): 67–75, 2003; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjc.10007 [ABSTRACT FROM AUTHOR]