bluuebunny/crossref_metadata_2025_split huggingface dataset:
Restriction on realization of single‐amplifier impedance‐converter and its application to LCR filter simulation
Authors: Fujii Nobuo , Yanagisawa Takeshi 
Published: July 2007

AbstractThis paper proposes a method to realize single‐amplifier impedance converters (SAICs), while a conventional general impedance converter (GIC) for an active RC filter requires two operational amplifiers. First, various restrictions required for realizing an SAIC were investigated. Then a method of realizing an SAIC was described which utilizes the property of a grounded active RC two‐port circuit having an impedance‐conversion effect; and the SAIC was applied to the simulation of an LCR filter. an SAIC can approximately simulate an arbitrary circuit with some restricted circuit constants, although the range of circuits which can be simulated exactly is limited. As examples, an asymmetrical sixth‐order BPF and a symmetrical sixth‐order BPF were constructed and their experimental results agreed well with the theoretical values.
Max‐flow min‐cut theorem for the multicommodity flows in certain planar directed networks
Authors: Nagamochi Hiroshi , Ibaraki Toshihide 
Published: July 2007

AbstractClasses CB, CS and CU have been found as the classes of networks, for which there exist efficient graph‐theoretical algorithms for the multicommodity flow problem on the directed network. This paper shows that the max‐flow min‐cut theorem holds for classes CB and CS. In other words, in those networks, if the multicommodity flow is not feasible, there always exists a cut for a source‐and‐sink pair, indicating the unfeasibility of the flow. For class CU, an example has already been presented where the max‐flow min‐cut theorem does not hold.
Synthesis of a flow network with a given centrality on each vertex
Authors: Takeuchi Masashi , Kishimoto Wataru , Kishi Genya 
Published: July 2007

AbstractThis paper considers the problem of constructing a flow network when the centrality at each vertex is given. Here, attention is given to a centrality function such that the centrality at a vertex under consideration is the sum of maximum flow values between it and all other vertices. It is a representative centrality function among those representing the centrality of each vertex in an undirected flow network in which edges have capacity. First, we introduce a necessary and sufficient condition wherein a given sequence is a centrality sequence for a flow network. Next, we present a procedure for determining a terminal capacity matrix of a flow network with a given centrality sequence. Furthermore, we determine a terminal capacity matrix of a flow network with a given centrality sequence such that the sum of edge capacity is minimum. From these discussions we can see that the problem of constructing a flow network when a centrality is given, can be reduced to the previously known problem of constructing a flow network when a terminal capacity matrix is given.
A new structure of prefilters for FIR digital filters
Authors: Aikawa Naoyuki , Sato Masamitu 
Published: July 2007

AbstractThe cascade of a prefilter and an equalizer is effective for realizing an FIR digital filter with few multipliers. Generally, the number of multipliers in the entire filter decreases as the prefilter response becomes closer to a given specification. This paper proposes a new structure of pre‐filters which can produce a response closer to the specification than a conventional structure. Furthermore, a new method of cascading the prefilters, and which improves both the passband and stopband characteristics, is described. In this method, a pre‐filter cascaded with the preceding ones is designed so as to compensate their approximate characteristics. the small passband ripple and large stopband attenuation can be realized by using the proposed prefilters and an equalizer with only few multipliers.
A new DSP‐oriented FFT algorithm
Authors: Mikami Naoki , Matsuoka Masahiro 
Published: July 2007

AbstractIn the traditional fast‐Fourier trans form (FFT) algorithm, the design for high‐ speed operation is made based on the number of operations, especially the number of multiplications, which is time‐consuming in the general‐purpose processor. This is not necessarily reasonable in the digital signal processor (DSP), where the multiplication can be performed in the time equal to that of other operations. From such a viewpoint, this paper proposes an algorithm which matches the feature of DSP, where the pro cessing is fast when the multiply‐accumulate operations are continued in execution. the algorithm is applicable when the number of data IV can be factorized into mutually prime factors. DFT for smaller number (factor) of data is modified into the cyclic convolution, and is executed at a high speed (small‐ N algorithm). the proposed algorithm is composed of the small‐N algorithms and DFT for a large number of data (multidimensional algorithm), which combines small‐N algorithms into a multidimensional DFT. the number of operations is not necessarily reduced in the proposed algorithm, compared with the traditional FFT, but it was verified by an actual run of the program on DSP that the execution time is reduced.
Pitch synchronous Mel inverse LSP analysis‐synthesis technique of LPC voiced residual
Authors: Mano Atsushi , Ozawa Shinji 
Published: July 2007

AbstractThis paper proposes the pitch synchro nous mel inverse LSP analysis‐synthesis technique as a new coding method for LPC voiced residual. the pitch synchronous amplitude spectrum of the voiced residual has characteristic zeros. By applying the inverse LSP analysis in mel domain to this spectrum, the spectrum can be encoded with a high efficiency with order of analysis as small as 12. Then mel LSP inverse filter is proposed which can synthesize directly the voiced residual in the time domain from the mel LSP coefficients. It is shown that the voiced residual can easily be synthesized. For the voiced residual obtained from the forementioned analysis‐synthesis system, the order of the model formel inverse LSP analysis, quantization characteristics and the interpolation performance were investigated, with the mean spectrum distortion of mel pitch synchronous amplitude spectrum as the evaluation measure. the optimization of each parameter was discussed. A pitch synchronous mel inverse LSP analysis‐synthesis system was constructed based on the obtained result. A subjective quality evaluation was made for the synthesized speech, and it was verified that a high‐quality speech can be synthesized at 4.8 kbit/s. These results indicate that the proposed method is applicable with merits to the low bit‐rate vocoder.
Synthesis of LDI digital filters based on symmetrical lattice network simulation
Authors: Iwakura Hiroshi , Niwa Hiroki 
Published: July 2007

AbstractIt is important to realize a low‐sensitivity digital filter, from the view point of economy such as hardware complexity, for which numerous studies have been made. From those studies it is known that the digital filter simulating the analog network, such as leapfrog and wave digital filters, exhibit low coefficient‐sensitivity characteristics. This paper is based on the symmetrical lattice reactance network. It is shown that the required transfer function is given as the output difference between the doubly resistively terminated IX ladder networks constructed with two kinds of one‐port LC reactance elements as the series elements. A method to construct a digital filter is proposed by the leapfrog structure using LDI. the digital filter constructed by the proposed method requires a fewer number of elements. the operating transfer function and the return‐ loss function are realized simultaneously. It is shown by examples that compared with the traditional method based on the simulation of the analog network, almost comparable passband characteristics can be realized.
Use of generalized cepstral distance measure in isolated word recognition
Authors: Kobayashi Takao , Kondo Shun‐Ichi , Imai Satoshi 
Published: July 2007

AbstractThis paper proposes a generalized cepstral discance measure based on minimum phase spictral models and shows the performance of the distance measure for isolated word recognition. the generalized cepstral distance measure approximates the L2 norm between two spectra on a fractional power magnitude scale. the measure is equal to the cepstral distance measure for the special case. Calculation of the distance can be performed efficiently from cepstral coefficients or linear predictor coefficients. Isolated word recognition experiments using a vocabulary of 20 highly confusable Japanese city names indicate that the generalized cepstral distance measure gives higher recognition accuracy than conventional distance measures. Three analysis parameter sets, which are FFT cepstral coefficients, linear predictor coefficients, and improved cepstral coefficients, are also compared in terms of their effects on the recognition performance.
A highly reliable trellis‐coded modulation using binary convolutional codes
Authors: Yamaguchi Kazuhiko , Imai Hideki 
Published: July 2007

AbstractThis paper proposes a coding scheme for multilevel channel coding, aiming at the high‐speed and highly reliable information transmission. the coding scheme is realized by combining binary convolutional codes. the scheme is based essentially on the idea of the coding by Imai‐Hirakawa's block code, and is implemented easily. Since the convolutional code and Viterbi decoding are em ployed, a high coding gain is realized. As an example, 2M level PSK with coding rate (M ‐ 1)/N and 2'M‐level QAM are presented. the result is compared with Ungerboeck's scheme, which is considered at present for practical application. It is shown that a larger coding gain is realized for the same har‐Iware scale. In the convolutional code, the coding scheme minimizing the decoding error probability for the given constraint length and the coding rate is called the optimal scheme. the proposed scheme is not the optimal in this sense. Consequently, the deviation from the optimal scheme is evaluated using the error exponent. it is shown that when the decoding error probability is set as constant, the implementation of the decoder is easier by the proposed method.
A chaotic circuit family including one diode
Authors: Saito Toshimichi 
Published: July 2007

AbstractThis paper considers the chaotic phenomena in a family of three‐dimensional autonomous circuits composed of R, L, C, ‐R (linear negative resistor) and one diode. First, a canonical equation is proposed which describes such a family of circuits. It is a piecewise‐linear equation containing one large parameter α. When a is sufficiently large, an idealization applies where the diode is regarded as a switch. This idealiza tion is valid theoretically. the equation is simplified to a system where the two‐ and three‐dimensional linear equations alternate. the one‐dimensional Poincaré mapping can be defined. the one‐dimensional mapping is formulated explicitly. the generation condition for the chaos is presented in the sense of Li‐Yorke. the condition is verified by an experiment corresponding to a circuit example for chaos generation.
Lyapunov stability of two‐dimensional digital filters
Authors: Hinamoto Takao , Hamanaka Takashi , Maekawa Sadao 
Published: July 2007

AbstractThis paper discusses first the Lyapunov stability based on the Fornasini‐Marchesini model, and derives the two‐dimensional Lyapunov stability condition. the derived condition is general in the sense that it includes as a special case, the traditional two‐dimensional Lyapunov stability based on the Roesser model. Then the necessary and sufficient condition for the forementioned condition is derived. A sufficient condition is shown for the limit‐cycle not to be generated by the overflow in addition, due to the finite word length. Finally, the filter design based on the derived stability condition is discussed.
Realization of leapfrog‐type switched capacitor filters using time‐shared operational amplifier
Authors: Iwakura Hiroshi 
Published: July 2007

AbstractSwitched capacitor filters (SCFs) have recently been used widely since they are suitable for integrated circuits. Most conventional SCFs require the same number of operational amplifiers (OPAs) as the orders of a filter circuit, and each OPA requires a considerable area of a chip and power dissipation compared with other components. Therefore, it is desirable to realize a filter circuit with fewer OPAs. A known method which satisfies these conditions is the leapfrogtype SCF which reduces OPAs to about half.This paper proposes an SCF which can reduce OPAs further, by using a signal flow graph (SFC) with no delay‐free loop which is the necessary condition for realization of a digital filter. In the proposed method, an SFG is divided into several blocks so that each of them can be calculated independently. Then the blocks are calculated sequentially in the order which makes the calculation possible. In other words, a single OPA is time‐shared with multiphase clocks. This paper also proposes another method in which the same DFGs are divided into blocks so that a parallel operation is possible.
Variable threshold soft decision decoding
Authors: Yamaguchi Kazuhiko , Iizuka Hiroshi , Nomura Eiji , Imai Hideki 
Published: July 2007

AbstractSoft decision decoding is a decoding method which tries to improve the error‐correcting capability by utilizing the information concerning the reliability of the received symbol. One of the soft decision decoding methods for practically useful block code, is APP (a posteriori probability) decoding applicable to the majority‐logic de‐codable code. In APP decoding, the error is decided for each symbol as in majority‐logic decoding. This paper proposes a new decoding method (variable threshold APP decoding), in which the error‐correcting capability is improved by controlling the order of error decisions in APP decoding. the performance is evaluated by simulation. As a result, it is shown that the variable threshold APP decoding exhibits better characteristics than APP decoding and approximate APP decoding both in bit error rate and block error rate. When, for example, the proposed decoding is applied to (73, 45) difference set cyclic code, a coding gain of 1.0 dB is obtained compared with the approximate APP decoding. In the proposed decoding method, the time required for decoding is increased, while the hardware scale is almost the same as that of the approximate APP decoding.
Second‐order switched capacitor filter block with two clock frequencies and its design
Authors: Takeda Yasuyoshi , Sato Chikara 
Published: July 2007

AbstractA second‐order variable SC filter block with two clock frequencies has been studied. This block consists of two first‐order variable SC filters feedback connected. By changing the two clock frequencies independently, both the resonant frequency and the Q value can be adjusted. It is possible to fabricate a higher‐order variable SC filter with degrees of freedom by cascading several of the blocks presented here. Derivation of the difference equation, transfer function and design method of this second‐order variable SC filter block is the objective of this paper.
Robust model matching design method which ensures stability
Authors: Zhong Yi‐Sheng , Tagawa Ryozaburo , Eisaka Toshio 
Published: July 2007

AbstractThe 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.
Experimental approximate solutions of nonlinear discrete oscillation with parametric excitation
Authors: Tanaka Toshiyuki 
Published: July 2007

AbstractIn the analysis of a discrete nonlinear system, one of the important problems is to derive a steady solution for the nonlinear difference equation characterizing the system. However, the study for the solution method of the difference system has not been sufficient, and examples of the methods being applied to the parametric oscillation are few. This paper presents a method which derives the steady solution for the quasi‐linear difference equation containing a periodic parametric excitation in its weakly nonlinear function with small parameter e. In this method the harmonic balance and the averaging methods used in the solution of the continuous nonlinear differential equation are applied to the discrete system. Since the nonlinear term in the equation is bounded by ε the discrete waveform in the steady state is almost sinusoidal. This paper presents numerical examples for the solutions by the two methods, assuming the solution as the discrete oscillation corresponding to the sinusoid of the continuous system. As a result, the same solution is obtained by either method. Comparing the result with the output waveform obtained directly by the numerical calculation for the fundamental equation, the validity of the approximate solution by the proposed method was verified.
Cycle breaking and longest path in channel routing problem
Authors: Shimamoto Takashi , Sakamoto Akio 
Published: July 2007

AbstractChannel routing is one of the automatic routing techniques in LSI design. In the channel routing by trunk‐branch scheme, if a cycle exists in the relational graph between subnets, the trunk of a net must be divided to break the cycle. Reference [11] presented a method, which divides the trunk of a net at a terminal on the graph representing the routing requirement. the effectiveness and pseudo‐effectiveness of the method are defined. Through the discussion of the necessary and sufficient condition for the realizability of the division, a condition is derived for the routing requirement to be realized. This paper discusses the degree of freedom remaining in the trunk division (i.e., the connection of free arcs in the modification of the requirement graph). the effect of the degree of freedom on the final result, especially on the longest path in the requirement graph, is discussed. As a result, it is shown that if the trunk division satisfies a certain condition, it does not increase much the longest path length.
A block placement procedure using a force model
Authors: Onodera Hidetoshi , Tamaru Keikichi 
Published: July 2007

AbstractThe paper treats the problem of automatic placement of building blocks of different sizes. This problem arises frequently in automating the layout design of LSIs. In this paper, a method is described for finding a global placement taking into account the connectivity between blocks as well as dimensions of blocks and placing regions. A method is proposed to establish a force model such that attractive forces caused by the connectivity between blocks and repulsive forces caused by the block overlaps are exerted on the blocks, and to derive a placement result as a system equilibrium state. the placement procedure is as follows. We start with the initial placement obtained only by applying attractive forces caused by connectivity, and then add gradually repulsive forces which are calculated based on the actual overlapping area between blocks. As a result, block overlap is removed and thus we can obtain a global placement considering local shape fitness while taking connectivity between blocks in a global way. In this paper, we explain first an outline of the proposed algorithm and then describe details of the methods for realizing it, e.g., functional forms of repulsive force against overlapping area, increasing ratio of repulsive force, and calculation method for the block equilibrium position and block orientation. Finally, the experimental results using examples with 17 and 33 blocks to demonstrate the effectiveness of the proposed algorithm are explained.
Effects of harmonic waves on output characteristics of electromagnetic thickness‐sensor
Authors: Yamada Takayuki , Suzuki Isao 
Published: July 2007

AbstractAn electromagnetic sensor to measure the thickness of metallic objects (including rusty objects) was proposed previously by the authors. the effects of higher‐harmonic waves of the sensor output due possibly to a magnetic nonlinearity of an object, however, was not taken into account. This paper examines this effect in relation to the thickness of the object, and obtains the optimum measuring frequency and current considering a heat‐loss in the sensor at a very low frequency; these are about 0.7 Hz and about 60 mA, respectively. With the improved method, the measurable ranges of thickness of objects are 0.5 to 4.5 mm for steel plates, and 0.5 to 7.0 mm for casting plates, an improvement by a factor of 2.5.
Spatial correlation and probabilistic evaluation of fluctuations at two observation points in a diffuse sound field
Authors: Ohta Mitsuo , Nakasako Noboru 
Published: July 2007

AbstractFluctuations at two observation points in a diffused sound field vary with correlations. In actual acoustic measurements, a sound level meter with a time constant is often used for a white noise input with a frequency bandwidth. Hence, fluctuations of the sound intensity must be considered first. Numerous studies have been reported on the probability density function and the spatial correlation in a diffuse sound field. How‐ever, the former is limited to a single variate distribution while the latter is restricted to the linear correlation of fluctuations. In this paper, the joint probability distribution of the fluctuations of the sound intensity is derived first so that various linear and nonlinear correlation effects between the fluctuations at two observation points are found in connection with the well‐known T‐type probability distribution at in dividual observation point. the resultant distribution expression is a two‐variate joint T distribution for which the arbitrary frequency bandwidth of the input and the effects of square function and arbitrary time constant of the measurement device are reflected structurally in each statistical parameter. Further, to distinguish all correlations between fluctuations at the two observation points A and B, the conditional probability density function on the fluctuation at the point A is derived in the case where a conditioned level is set at the point B. Finally, the present method is applied to the fluctuation data of sound measured in a reverberant room so that the actual usefulness is confirmed.