bluuebunny/crossref_metadata_2025 huggingface dataset:
An SAW contiguous filter bank with an m‐derived ladder
Authors: Misu Koichiro , Nagatsuka Tsutomu , Kimura Tomonori , Wadaka Shusou , Takeda Fumio , Hashimoto Tsutomu
Published: July 2007

This paper proposed anm‐derived ladder for combining surface acoustic wave (SAW) filters in a contiguous filter bank and described the fabricated and analytical results for this filter bank. In the conventional filter bank with a constant‐kladder, spurious responses caused by the SAW filter in the stopband are not sufficiently suppressed. In the proposed SAW contiguous filter bank with anm‐derived ladder, a parallel resonant circuit is used. It consists of an inductor and a capacitor as the series element instead of an inductor, which is used as the series element of a constant‐kladder. By setting the attenuation pole caused by the resonance of the series element to the spurious frequency of the SAW filter, the spurious response of the SAW filter with them‐derived ladder are suppressed more than with the constant‐kladder. Then a 10‐channel SAW filter bank was constructed and the effectiveness of the proposed SAW contiguous filter bank with anm‐derived ladder was verified.
Algorithms for finding noncrossing paths with minimum total length in plane graphs
Authors: Takahashi Jun‐Ya , Suzuki Hitoshi , Nishizeki Takao
Published: July 2007

Assume that G is an undirected planar graph and the edge length of G is a nonnegative real number. When k terminal pairs are specified on two specified face boundaries, this paper gives an algorithm that derives the “noncrossing paths” with the minimum sum of lengths that connects the respective terminal pairs. By the noncrossing paths is meant the paths which do not cross on the plane, although the point or the edge may be shared. the computation time of the proposed algorithm is O(n log n), where n is the number of points on the planar graph G; k need not be a constant.
A signal‐processing method for evaluating a continuous‐level probability function based on few sampled data with roughly digitized level—an application of Z‐transform and numerical laplace transform
Authors: Fujita Yoshifumi , Ohta Mitsuo , Ezumi Hiromichi
Published: July 2007

In the evaluation of the total structure of the non‐Gaussian random noise observed in the real environment, it often happens that not only the mean and the variance of the fluctuation but also the whole probability distribution should be estimated as reasonably as possible, based on a few number of sampled data with a rough quantization level.

This paper notes the following property. the mean, the variance, and the higher‐order moments once evaluated by the averaging operation of the sampled data are more stable and reliable than the individual frequency curve derived directly from the sampled data. Further‐more, by utilizing the saturation property of the cumulative probability distribution, a new method is presented to estimate the probability distribution of the original continuous level. More precisely, the probability distribution for the quantized level is represented by a nonparametric Z‐transform suited to the theoretical evaluation of the statistical moments.

The inverse Laplace transform is applied to the first‐order hold interpolation of the obtained Z‐transform representation and the continuous level distribution is estimated. More precisely, the series expansion form is terminated as an approximation and the result is smoothed. Then the numerical inverse Laplace transform is applied using FILT.

Finally, the proposed method is applied to the data with exponential or Erlang distribution generated by a simulation using the random variable, and the validity of the principle is verified. the method also is applied to the actual environmental noise, and the practical usefulness is verified.
Nonlinear noise reduction based on zero‐crossing representations of a wavelet transform
Authors: Cheong Cha Keon , Aizawa Kiyoharu , Hatori Mitsutoshi , Saito Takahiro
Published: July 2007

A novel nonlinear noise reduction method that effectively reduces both impulsive and Gaussian noise is proposed and investigated. In the proposed method, a zero‐crossing representation that is a kind of compressed description of the coefficients of a multiscale wavelet transform is obtained first and then, using characteristics of the transform coefficients, any noise that is thought to have entered the zero‐crossing representation is suppressed. Noise reduction is achieved by reconstructing the image from the modified zero‐crossing representation. It is shown that, compared to nonlinear filtering methods, the proposed method is more effective for simultaneously reducing both impulsive and Gaussian noise for one‐ dimensional signals and two‐dimensional image signals.
Haar wavelet transform with interband prediction and its application to image coding
Authors: Kuroki Nobutaka , Numa Masahiro , Nomura Takanori , Kameda Masashi , Hirano Kotaro
Published: July 2007

Although the discrete cosine transform (DCT) and wavelet transform have been used as effective techniques for reducing the redundancy of image waveforms, they have the problems of high complexity and causing block distortion. In this paper, a Haar wavelet transform with interband prediction is proposed that permits high‐speed processing and which is suitable for coding as a subband decomposition technique since it generates little block distortion in the decoded images. the interband prediction method of this proposed transform method uses the derivative of the low‐band waveform to predict the high‐band waveform each time a band is split into two sub‐bands. This results in a transform process that gives a prediction residual signal with lower entropy. During the inverse transform, the prediction coefficients obtained from the forward transform are used to predict the high‐frequency waveforms and then, by adding the prediction residual, the original high frequency waveforms are reconstructed. Because the high‐frequency waveforms are generated that smoothly interpolate the up‐sampled, low‐frequency waveforms, any block effects from this prediction technique are difficult to see in the decoded image. A detailed analysis of the proposed interband prediction process is performed by interpreting the proposed transform as a subband decomposition process that is based on the use of a symmetric short‐kernel filter (SSKF) filter bank; then experiments are conducted that demonstrate its performance as a subband decomposition for coding of real image data and the possibility of progressive build‐up of the decoded images. Results show that this transform method is effective for coding image data.
On a method of cryptanalysis for cryptosystem with involution
Authors: Tsunoo Yukiyasu , Okamoto Eiji , Uematsu Tomohiko
Published: July 2007

Cryptology provides the technological basis for guaranteed network security. In the field of business, to guarantee the security of information, everyone needs easy‐to‐use cryptology. Starting with the announcement by the U.S. National Bureau of Standards of the DES cryptosystem and the proposal by NTT of the FEAL cryptosystem, secret‐key cryptographic algorithms have been published, involution‐type block cryptosystems which use secret key to encipher 64 bits of plain text as 64 bits of cipher text [2, 3, 9, 10, 12]. NTT has pro posed the one‐way function MAP, which is also an invo lution‐type transform. In this paper, we study a new cryptanalytic method for involution‐type cryptosystems and show by example that it can be applied practically in a ciphertext‐only attack on the one‐way function MAP. For an involution‐type cryptosystem, by increasing the amount of construction in the data‐randomization phase and by increasing the size of the secret key, we can guarantee the computational security by preventing the estimation of the secret key. However, if we study the properties of the function used in the design of the data‐ randomization phase, by specifying input/output values of the function which are independent of the secret key, it may be possible to use the inverse function [11] to calculate the secret key. This method is called the inter Mediate cipher method. the intermediate cipher method differs from previous methods in the small quantity of data required for breaking.
High‐performance VLSI architecture suitable for control systems for state‐space digital filters using distributed arithmetic
Authors: Tsunekawa Yoshitaka , Miura Mamoru
Published: July 2007

This paper proposes and evaluates high‐performance VLSI architecture for real‐time state‐space digital filters used in digital signal processing and digital control. A VLSI‐oriented highly parallel architecture for state‐space digital filters with high sampling rate and small latency has already been proposed by the authors. To speed up and reduce hardware complexity, the distributed arithmetic (of which processing time depends on only, word length) is applied to this architecture, making good use of highly accurate state‐space digital filters. The very high sampling rate can be implemented independently of filter order and the number of the input and output.
A test theory using probabilistic network
Authors: Ueno Maomi , Ohnishi Hitoshi , Shigemasu Kazuo
Published: July 2007

This paper aims at estimation of the state of the knowledge (understanding) of the individual learner by executing a test composed of a small number of items.

The following approach is proposed: (1) a teacher model is introduced, and the inference process for the understanding state of the learner is represented using the Bayesian network; (2) as the network information of the items based on the teacher model, EVINI (expected value of item network information) is proposed. As the test information, EVTIN (expected value of test information with network) is proposed; and (3) a strategy for item selection based on EVINI is proposed.

The following advantages are realized in practice: (1) only a small amount of data is required to construct the model; (2) the goal of inference is to know the understanding state of the individual learner, and the result provides the feedback from an educational viewpoint; (3) the knowledge of the teacher about the domain can be utilized in the inference process, which helps to improve the prediction efficiency; and (4) it is easy to add or modify the knowledge of the teacher model.

In this paper, the proposed technique is applied to the real data, as the inference example of the method. the strategy for presenting the items is compared to item response theory, and discussions are made from the theoretical viewpoint.
An arithmetic coding scheme for gray‐scale images with minimum description length criterion
Authors: Zhang Qi , Kohno Ryuji , Imai Hideki
Published: July 2007

In the aritbmetic coding of the multilevel gray‐scale image (128 levels) up to now, the fixed model for the gray‐scale image is applied.

This paper proposes a method to improve the compression ratio by introducing the minimum description length (MDL) into the encoder to describe more precisely the local property of the image and by adaptively selecting the probabilistic model. In the course of this development, a practical form of MDL criterion is presented to be adaptively applied to the encoding as an approximation to the MDL criterion proposed by Rissanen.

A problem in the arithmetic coding of the multilevel gray‐scale image using a probabilistic model with a large number of parameters is the deterioration of the compression ratio. to solve this problem, this paper proposes a convergence acceleration method for the average code length for the arithmetic encoding of the high‐level gray‐scale image. Then, based on the proposed acceleration method, the MDL criterion is introduced into the arithmetic coding of the multilevel gray‐scale image. Finally, the effectiveness of the proposed coding method is demonstrated by computer simulation.
Constrained transition vector quantization method for noisy channel speech coding
Authors: Hanawa Masanori , Hasegawa Takaaki
Published: July 2007

A novel coding method is considered here that combines source coding of speech and channel coding. A vector quantization method using a constrained transition codebook is proposed. Encoding with the proposed method gives codes an error‐correction ability without addition of explicit redundant bits by error‐correcting coding.

To compare the proposed and the ordinary source‐ channel tandem coding methods, we carried out computer simulations, transmitting LPC cepstrum vectors of speech. In the presence of channel errors, cepstrum distortion of the decoded cepstrum vector by the proposed method is significantly decreased compared with one by the ordinary vector quantization. Moreover, from comparisons with tandem methods, it is shown that the proposed method is more robust than the ordinary tandem methods. Moreover, its error‐correction ability is almost equal to rate 1/2 convolutional coding.
Digital signal processor supporting two types of instruction sets
Authors: Kiuchi Atsushi , Nakagawa Tetsuya , Kaneko Kenji , Baji Toru
Published: July 2007

An architecture for a general purpose digital signal processor has been developed that enables the specification of peripherals and the core part to be easily changed for specific applications. an evaluation chip LSI providing 24‐bit fixed‐point arithmetic was created in order to verify the performance. This digital signal processor (DSP) basically uses RISC‐type architecture in order to simplify the hardware structure and provides two types of instruction sets, called RISC‐type and DSP‐type instructions, to maintain the benefit of the original DSP. the RISC‐type instruction set consists of several types of single‐task operating commands: arithmetic, logical, data transfer, branch, etc. It provides wide operand flexibility. On the other hand, the DSP‐type instruction set consists of parallel operating commands as provided for conventional DSP which execute several data processing and data transfer operations. However, operand flexibility is restricted by providing these two types of instruction sets in one LSI and a higher performance can be achieved than with conventional DSPs which have the same bit length and only one instruction set.
Parameter‐controlled chaos neural networks
Authors: Kasahara Tsuyoshi , Nakagawa Masahiro
Published: July 2007

A chaos neural network with externally controlled parameters is proposed and a few applications to an association model and a combinatorial optimization problem are given. It is found that a relatively higher success rate for the memory retrieval can be achieved by using the association model instead of the conventional model (burdened by chaotic wandering) because unacceptable trapping at a spurious state is avoided and the traveling salesman problem can be solved using parameter control reminiscent of the annealing in the Boltzmann machine.
A generalization of lin divergence and the derivation of a new information divergence
Authors: Shioya Hiroyuki , Da‐Te Tsutomu
Published: July 2007

The f‐divergence is a general class of divergences which includes various divergences used in representing the difference between two probability densities. It was introduced by Csiszár. Lin, on the other hand, in 1991 introduced a divergence based on a new idea. the inequalities relating the divergence to the existing two divergences are shown, and the results are applied to the problems in information theory.

This paper discusses the relation between the inequality concerning Lin's divergence and the inequality concerning the f‐divergence. Based on Lin's method of constructing the divergence, a new divergence called Hermite‐Hadamard divergence is introduced. the property of the proposed divergence, as well as its relation to Lin's inequality, are discussed. Using the proposed divergence, an inequality which is improved further compared to the inequality relating the Kullback diver gence and Hellinger distance is derived. Those results are derived in this paper.
Asynchronous excitation phenomena in oscillatory circuit containing periodically operating analog switch
Authors: Matsuki Michitaka , Mori Shinsaku
Published: July 2007

In an oscillatory circuit containing a nonlinear negative resistor whose v‐i relationship is i =‐g1v + g3v3, asynchronous excitation by external force cannot exist under certain conditions. However, it can be confirmed experimentally and theoretically that excitation can exist when the resistor is biased and periodically switched by an analog switch at a much higher frequency than both resonant and external frequencies. The oscillation modes obtained change in the order of trivial, hard, soft, hard and trivial as external force increases. This mode variation has been found to be the same as that in i = g, 1v ‐ g3v3 + g5v5 except for the hard oscillation.
Chaos and fundamental bifurcation phenomena from a relaxation oscillator with periodic thresholds
Authors: Kohari Ken'Ichi , Saito Toshimichi , Kawakami Hiroshi
Published: July 2007

A relaxation oscillator which includes a hysteresis element with periodic variation of the threshold is analyzed. Deriving a one‐dimensional return map, the parameter regions are clarified where the system exhibits a periodic solution or quasi‐periodic solutions and where the response characteristics of the system having periodic solutions becomes devil's stair and where the attractors may coexist. When the system has chaotic response, then coexistence of chaotic and periodic attractors is confirmed. Various results are confirmed from the laboratory experiments.
A synthesis of an optimal fuzzy filter based on local statistics
Authors: Takashima Hironori , Taguchi Akin , Murata Yutaka
Published: July 2007

This paper presents a design method of data‐dependent filters that uses simplified fuzzy inference. Since the antecedents of fuzzy inference can comprise several local characteristics (i.e., observation values), it is possible for the fuzzy filter to adjust its weights to adapt to local image data. the tuning of membership functions and fuzzy rules of the proposed filter results in a least mean square (LMS)‐like algorithm. Thus, local characteristics can be increased for the proposed fuzzy fiter optimally. This paper, introduces a new observation value (calculated from local statistics) into the proposed filter. the proposed filter changes filter behavior according to the local properties of signals and provides good noise attenuation in all regions of image, including detail regions, while still preserving the details.
Nonlinear signal processing based upon a reconfigurable knowledge database
Authors: Iiguni Youji
Published: July 2007

One of the most important problems in signal processing is to estimate the output for a known input, here referred to as a query from the input/output (I/O) data seen to this point. A nonlinear adaptive estimation method that uses a k‐d trie (k‐dimensional binary digital search tree) as an alternative to the k‐d tree is presented here. The observed data are stored in the k‐d trie, and a local model to answer each query is built by using thennearest neighbors of the query. The database contents are updated as each new data point is available such that the latestNdata are contained. This enables nonlinear adaptive signal processing in nonstationary environments.

A storage procedure for the k‐d trie that allows efficient update is presented for further reduction in computation time and storage requirement. A recursive least square algorithm, used for a recursive production of the local model, improves computational efficiency. Finally, a method of removing outliers is presented for improving estimation accuracy.
A partial realization algorithm for finite‐size two‐dimensional arrays
Authors: Matsuo Tsuyoshi , Hasegawa Yasumichi , Okada Yoshikuni
Published: July 2007

The following figure realization problem has already been posed. “For arbitiarily given figure (two‐dimensional array), find the (mathematical) model that realizes (describes) the figure. If there exist two or more models, show that they are isomorpric.” As a solution to the problem, the figure realization theorem is obtained as follows. “For any given figure, there always exists a canonical commutative linear representation system that realizes the figure, which is unique except for the isomorphism.” There also are studies to analyze the finite‐dimensional canonical commutative linear representation system.

Based on those results, this paper discusses the partial realization (faithful description) problem for the finite‐size two‐dimensional array. an algorithm is given which derives the minimum‐dimensional commutative linear representation system that partially realizes (faithfully describes) the given finite‐size two‐dimensional array. As an application example, the dyeing and weaving patterns are considered and the usefulness of the proposed algorithm is shown. the proposed method is a processing to compress the information of the finite‐size two‐dimensional array by maximally utilizing the regularity of the finite‐size two‐dimensional array considering the vertical and horizontal positions.
Optimum linear transformation for a quadratic form of signals
Authors: Sakamoto Hiroyasu
Published: July 2007

This paper proposes the optimum skew linear transformation of signals which gives the minimum mean error of a positive definite quadratic form of the signals. the relation of this problem with hybrid coding is emphasized. Based on conventional orthogonal KLT, the condition for the optimum transformation is given in a generalized eigenvalue problem. We show the geometrical meaning and a solving method of the condition, and discuss the existence and the uniqueness of its solution. Numerical performance comparisons show superiority of this transformation to others such as KLT and DCT. Because this transformation is a generalization of KLT to a general quadratic form, it is as significant as KLT is.
Prefiltering for blind equalization
Authors: Itoh Katsuko , Shimamura Tetsuya , Suzuki Jouji
Published: July 2007

A major problem with blind equalization algorithms based on the distribution matching principle is that they need a long time to accomplish their convergence. What affects convergence has not been discovered to date. the similarity between the least mean‐square (LMS) algorithm (widely used for adaptive equalization) and the blind algorithms is considered here. It is expected that the convergence of the blind algorithms is dependent on the condition number of the correlation matrix of the input sequence. Prefiltering methods, including coefficient‐fixed type and coefficient‐adaptive type, are derived for blind equalization. In these methods, the prefilters are realized by a prediction error filter, which has the ability to compensate for amplitude distortion induced by a channel. Since the prefilters have such ability, the blind equalizers to be cascaded with them are required only to compensate for phase distortion. As a result, the burden imposed on the blind equalizers is reduced. the prefilters output a near‐white sequence and lead to an improvement in the convergence of the blind algorithms, being degraded as the condition number is increased. the effective ness of the proposed methods is validated by computer simulations.
Design of noise generator using chaotic circuit
Authors: Suzuki Katsunori , Mori Shinsaku , Nishio Yoshifumi
Published: July 2007

This paper proposes an approach for a noise generator based on the chaotic circuit. In the proposed circuit, the diode is assumed to be an ideal switch. By means of this simplification, it is possible heuristically to understand the generation of several oscillation frequencies in the circuit. Also, it is possible to confirm the switching of the frequency from the attractor. The frequency depends strongly on the memory device in the circuit. More frequencies are generated by increasing the dimensions of the circuit since the spectral characteristics can be controlled easily by the variation of the circuit elements and by the increase of the dimensions of the circuit. Hence, the structure suggests an application to a noise generator. Further, the validity of the analysis results are confirmed by the circuit experiment.
Design of a cerebellar model arithmetic computer with adaptive resolution and its application to nonlinear signal processing
Authors: Kawamoto Isao , Iiguni Youji , Adachi Norihiko
Published: July 2007

A cerebellar model arithmetic computer (CMAC) with adaptive input space quantization is designed. the quantization intervals are compressed by the segmentation of weights in the region where the function to be learned changes rapidly, whereas those are extended by the integration in the region where the function changes slowly. the adaptive resolution technique reduces required memory space and improves learning speed. Results of simulation studies on nonlinear signal processing problems are reported. A storage procedure for the weights and new criteria for the integration and segmentation are also reported.
Mathematical programming formulation for neural combinatorial optimization algorithms
Authors: Urahama Kiichi
Published: July 2007

This paper considers the anlog neural solution of the combinatorial optimization problem. the solution method is analyzed based on the Lagrange multiplier method for the continuous relaxation problem of 0‐1 integer programming. It is shown that the solution process can be interpreted as the solution by the gradient method for the saddle point of the Lagrange function. an improved Hopfiled net is derived from the formulation as the pure integer programming. the elastic net and the generalized deformable model are derived from the mixed integer programming problem. Based on those results, an interpretation of the deterministic annealing is derived from the viewpoint of mathematical programming. It is shown that the Lagrange function can work as the Lyapunov function for the solution process and the convergence property of those neural methods of solution is analyzed.
Vias‐per‐net in multilayer multiterminal topological routing
Authors: Fujiyoshi Kunihiro , Kajitani Yoji
Published: July 2007

The proble of finding routing and layer assignment, which minimizes the number of vias ignoring the line width and the separation of lines, is NP‐hard even if the nets are restricted to two‐terminal nets. Numerous heuristic algorithms have been developed in an attempt to improve efficiency. However, as for properties of solutions, it has only been proven that the number of vias for t‐terminal nets in an optimum solution (min vias) is at most t ‐ 1, if the number of layers is restricted to 2. It is also known that properties of vias and terminals can be defined in a number of different ways for the multilayer topological via minimization (TVM) problem using unlimited number of layers. the problem becomes complex. In this paper, the forementaioned theroem is generalized and it is proven that a t‐terminal net uses at most t ‐ 1 adjacent‐layer connecting vias in any optimum solution for the general case without any restriction on the number of layers. Further, a class of problems is shown to exist which require that many vias. Similar results are also shown for all‐layer connecting vias.
A method of measuring moving characteristics of precision mechanical‐translation stages using ultrasonic plane waves and applying it to a line‐focus‐beam acoustic microscopy system
Authors: Ono Yuu , Kushibiki Jun‐Ichi , Chubachi Noriyoshi
Published: July 2007

A method of measuring moving characteristics of precision mechanical‐translation stages using ultrasonic plane waves is presented. The moving characteristics are obtained by analyzing the phase variation of ultrasonic waves reflected from a reflector set on the translation stage. The resolution is 4 nm at an ultrasonic frequency of 225 MHz; higher resolution is expected at higher frequencies. The method is demonstrated for a line‐focus‐beam (LFB) acoustic microscopy system. Measurement errors of the phase velocity of leaky surface acoustic waves, caused by the positioning errors of thez‐stage, are investigated both theoretically and experimentally. The linear positioning and straightness errors of thez‐stage are estimated at 225 MHz and these effects on velocity measurements are successfully removed using the measured moving characteristics.
Speaker identification using multipulse excited linear prediction analysis (LPC) vocal tract model
Authors: Hangai Seiichiro , Miyauchi Kazuhiro , Urabe Nobuyuki
Published: July 2007

The authors devised a method to identify a speaker where feature parameters are derived by normalizing the position and the magnitude of the subpulses in the multipulse train extracted from the vowel and are combined with 14th‐order linear prediction analysis (LPC) cepstrum. In other words, 10 code vectors are constructed for each registered speaker in the 16‐dimensional space. the speaker is identified by determining the code with the smallest VQ distortion for the vector of the unknown speaker.

A speaker identification experiment is executed using five sustained vowels of 100 speakers for the same time period. A 100‐percent identification rate is realized by combining the feature parameters of the multipulse source composing the feature vector and the feature parameters of the LPC cepstrum by the weights of 3:2. the speaker identification also is attempted for 10 subjects after 10 days by the same method, and the identification rate of 69 percent is obtained. the result is an improvement of 12 percent, compared to the case where only the LPC cepstrum is used. the identification rate also is improved for the speech one and two months later.
A new type circuit of switched‐capacitor transversal filter using transfer function coefficient ratios
Authors: Kawakami Izumi , Mori Toshihiro , Takahashi Nobuaki , Takebe Tsuyoshi
Published: July 2007

The switched‐capacitor (SC) filter is considered to be a very practical one for integrated circuit (IC) realization due to its ease of high‐accuracy realization because the coefficients of the transfer function can be realized with the capacitance ratio of the capacitors. However, when the coefficients of the transfer function are widely spread, the spread of the capacitance is also large, Hence, the total capacitance is eventually increased so that IC realization is difficult.

In this paper, a new SC filter configuration is proposed in which the coefficients of the transfer function of the SC transversal filter is rearranged in the order of magnitude and the ratio of the neighboring coefficients is used as the normalized capacitance (ratio to the capacitance of the operational amplifier for coefficient multiplication) of the coefficient capacitors. According to the proposed configuration, the spreading of the capacitance ratio and the total capacitance can be reduced significantly. Since the coefficient ratio of the transfer function is used, there is concern that the characteristic degradation by the element deviations might be larger than the one in the conventional SC transversal filter.

It is shown by the element sensitivity analysis and the numerical example that the characteristic variation of the cascaded SC filter by the present configuration is smaller than the one in the conventional cascaded SC filter in which the basic block is made of SC transversal filters and, therefore, there is no problem in practice. Finally, a test result is presented for an SC transversal filter using a third‐order transfer function coefficient ratio.
Design and implementation of linear‐phase two‐channel perfect reconstruction finite impulse response filter banks with equiripple stopband
Authors: Nagai Takayuki , Ikehara Masaaki
Published: July 2007

This paper presents a design and implementation method for the linear phase two‐channel perfect reconstruction finite impulse response (FIR) filter banks with arbitrary filter lengths. This method is labeled a weighted Lagrange‐Newton method; it is obtained by introducing the least squares weighting function into the Lagrange‐Newton method. Filter banks with good stopband attenuation (equiripple stopband) can be designed using this method. Furthermore, a new method is presented that makes the filter banks with a lattice structure using the property of a polyphase component matrix. the filter banks can achieve perfect reconstruction in spite of coefficient quantization.
Tracking system for spread spectrum communication on power line
Authors: Kimura Naoki , Miyashita Teruo , Sato Chikara , Suzuki Rie
Published: July 2007

In the power line data transmission using the direct sequence/spread spectrum (DS/SS) communication system, the frequency band is limited to 10 to 450 kHz. Then, it is useful to apply the Manchester coding so that the spectrum of the pseudonoise (PN) waveform contains a larer power in the given frequency band. As the delay‐lock loop (DLL) for the synchronization tracking of the Manchester‐coded PN waeform, M2/3Δ and M1.5Δ types are known. the M2/3Δ type has the feature that the tracking error is small, and the M1.5Δ type has the feature that the tacking range is wide.

This paper proposes the M2/3Δ improved DLL as an improved DLL which has both the excellent trakcing performance of the M2/3Δ type and the wide tracking range of the M1.5Δ type. It is a DLL combining the M2/3Δ and M1.5Δ types the basic equation and the tracking trajectory by a computer simulation are presented. Then the tracking performance is analyzed theoreitcally in the linear interval of the phase comparator characteristics, and the result is compared to the performance of the M2/3Δ type.
Burst error correcting 2k‐ary cyclic an codes generated by natural number A1·p
Authors: Murakami Ryutaro , Shimada Ryosaku , Ohkura Yoshiteru
Published: July 2007

Cyclic an codes are considered to be suitable for error detection and error correction in arithmetic operations. Cyclic an codes using radix 2k representation are called 2k‐ary cyclic an codes. This paper is concerned with 2k‐ary cyclic an codes for burst error correction. After describing the structure of these codes and arithmetic burst errors, the smallest code generator A1 that provides a 2k‐ary cyclic an code with the burst error correcting ability b is presented. We also suggest burst error correcting 2k‐ary cyclic an codes generated by A1.p. Furthermore, error‐trapping decoding for 2k‐ary cyclic an codes is shown.
Transient analysis of multisection transmission lines using a modified central difference method
Authors: Murakami Kazuhito , Ishii Junya
Published: July 2007

In designing high‐speed pulse integrated circuits, it is necessary to clarify that the propagating waveform is affected by the multiple reflection caused by discontinuity and mismatching of the interconnected lines. In this paper, we present the modified central difference method with simple boundary treatments for solving Telegrapher's equations only in the time domain in which both voltage and current solutions are derived at the same time. As this algorithm has a compact structure and memory independent of time as compared with other algorithms, the transient phenomena including multiple reflection along the whole line can be represented dynamically. It is also available for the steady state. It is shown by simulation studies that inhigh‐speed pulse transmission on microwave analog circuits and digital VLSIs, this technique is simple and useful in analyzing any type of propagating signals on uniform and various nonuniform transmission lines.
Realization of one‐dimensional nonuniform‐band maximally decimated filter banks using two‐dimensional filters
Authors: Fukuoka Takashi , Ozawa Hiroyuki , Ikehara Masaaki , Takahashi Shin‐Ichi
Published: July 2007

In the design of nonuniform‐band filter banks (NFB), each filter must be designed to satisfy the perfect reconstruction (PR) condition corresponding to division ratio for the system. A simple design method for nonuniform‐band maximally decimated filter banks using two‐dimensional (2‐D) filter to realize two‐channel system is presented here. A 2‐D filter is designed using a McCl‐ellan transform and one axis is fixed to arbitrary frequency in a 2‐D plane. Then the frequency response that turns out to be a cross sectional response of 2‐D filter, 1‐D variable filter that has fixed frequency as a parameter can be realized. Using 2‐D filters with complementary power, the lowpass filter and high‐pass filter always satisfies power complementarily. the 2‐D filter has been designed so that the filter pair satisfies the perfect reconstruction condition and it is adopted as the group of the necessary filters for NFB. In this method, the PR condition is not satisfied strictly but system simplification is achieved. Design examples, amplitude responses of them, and simulation results of the reconstruction of the signal are shown.
Tracking capability and floating‐point error analysis in multirate complex recursive weighted least squares algorithm
Authors: Shimizu Jun'ya , Miyanaga Yoshikazu , Tochinai Koji
Published: July 2007

This paper presents an analysis of a multirate complex recursive weighted least squares (MC‐RLS) algorithm based on an analytic signal. Conventional adaptive filters for multiple sinusoid extraction have been based on lattice filter structures or gradient methods because of their low computational cost and/or low sensitivity to quantization errors. On the other hand, the RLS algorithm is easy to introduce assuming time‐variant quantities in the algorithm when sinusoid frequencies have the time‐varying property. However, the relationship between the tracking capability and the quantization error of the least‐squares (LS) algorithm in the transversal structure has not been reported. In addition, an improvement algorithm for these errors have not been reported. In this paper, we shall describe a new RLS algorithm in a transversal filter structure with a superior transient property, reduced sensitivity to quantization errors, and low computational cost. First, an analytic signal‐based autoregressive model is introduced and the MC‐RLS algorithm is shown. Then, using an excess mean‐square error of the MC‐RLS algorithm, the tracking capability shown to be unaffected by the analytic transform and the decimation. In addition, the floatingpoint error and the computational cost of the MC‐RLS algorithm are analyzed. It is shown that both floatingpoint error and computational cost are smaller than those of conventional RLS algorithms that use real signals.
Probabilistic analysis of the time complexity of quicksort
Authors: Mizoi Tadashi , Osaki Shunji
Published: July 2007

Quicksort is a well‐known sorting algorithm based on the divided control. the array to be sorted is divided into two sets as follows. an element in the array is specified, and the set of values larger than the value of that element and the set of values smaller than that value are constructed. Each of those two sets are sorted independently. the procedure is iterated for the divided sets. In other words, the algorithm has a recursive structure. the average time‐complexity of the quicksort (the average number of comparisons) is O(n log n). Depending on the data to be sorted, however, the performance may be deteriorated drastically. In the worst case, the time complexity is O(n2).

In this paper, the generating function based on the time complexity of the quicksort is introduced and the mean and the variance of the time complexity is determined analytically using the generating function. Based on the derived mean and the variance, the chi‐square test is applied as to whether or not the distribution of the time complexity can be approximated by the normal distribution. Then, a method is proposed in which the probability R(x) that the time complexity exceeds a certain value x is determined accurately by approximating the distribution of the time complexity of the quicksort by a three‐parameter Weibull distribution. Finally, the selection of the sorting algorithm is discussed.
Automatic learning in chaotic neural networks
Authors: Watanabe Masataka , Aihara Kazuyuki , Kondo Shunsuke
Published: July 2007

A fully local algorithm which can automatically detect and learn an unknown pattern is proposed for a mutually connected recurrent neural network, and its fundamental properties are numerically analyzed. the algorithm is applied to chaotic neural networks composed of neuron models with spatiotemporal inputs and refractoriness and to conventional mutually connected neural networks. It is shown that the former could learn more patterns with greater robustness than the latter.
Use of static/dynamic parameters in automatic phonemic segmentation system for english continuous speech
Authors: Furuichi Chieko , Aizawa Katsura , Imai Satoshi
Published: July 2007

In this paper, a phonemic segmentation system is constructed which will be useful in practice as a preprocessing for English continuous speech recognition. the usefulness of the system is demonstrated. English continuous speech contains many weakly voiced vowels with ambiguous utterances called schwa, making stable detection of the phonemic boundary difficult. In fact, there has been almost no detailed proposal for a segmentation system that can be used in practice as a preprocessing to the recognition.

In the proposed method, the mel cepstrum is extracted from the speech signal, using the unbiased estimation of the log spectrum. This process is known to be stable, i.e., less affected by the fine spectral structure. Then the dynamic segmentation parameters are determined from the mel cepstrum such that the boundary of the schwa phonemes can be detected using the pseudo‐differentiation filter. By combining the dynamic parameters with the static parameters, the phonemewise segmentation is executed in a hierarchical form.

The proposed system can segment English continuous speech containing diversified phonemic environments into phonemes on the time axis. the system operates based only on the acoustic knowledge of phonemes, which are common to speakers with different qualities and utterances, without using the speaker‐dependent complex boundary detection rules. Also, the proposed system is evaluated by experiment using English continuous phoneme‐balanced speech uttered by one each native English‐speaking male and female for 350 s. For the total number of phonemes 3024, the detect ratio for the phoneme boundary is 97.1 percent, the boundary delete ratio is 2.9 percent, and the boundary add ratio is 24.2 percent.
Two‐dimensional spectral estimation method with data extension and its improvement
Authors: Shimamura Tetsuya , Miao Weiguo , Suzuki Jouji
Published: July 2007

This paper considers the application of the two‐dimensional (2‐D) spectral estimation for estimating the frequency of the 2‐D sinusoidal signal from a small number of data arrays. Comparing the 2‐D autocorrelation method, which is a typical 2‐D method of linear prediction, and the data extension method proposed by Frost et al., it is shown that the data extension method gives a better result of spectral estimation. It is noted that the estimation accuracy is deteriorated when an additive noise is mixed.

From such a viewpoint, this paper considers first the noise‐compensation method, where the noise power is subtracted from the diagonal elements of the correlation matrix in the Yule‐Walker equation in the one‐dimensional (1‐D) processing. Then, a simple and efficient noise‐compensated autoregression (AR) parameter estimation method is derived, which can solve at the same time the problems of the accurate estimation of the noise power and the deterioration of the AR parameter estimation accuracy due to the oversubtraction of the estimated noise power, whch have been considered unsolved.

Applying the idea to the data extension method, an improved data extension method is derived. It is shown that by this improvement, the accuracy of the data prediction is improved and the estimation of the 2‐D spectrum is improved greatly.
An identification method for nonminimum‐phase autoregressive moving average system using phase‐equivalent moving average system
Authors: Miyazaki Nozomi , Hamada Nozomu
Published: July 2007

One of the most important issues of signal processing is to identify the model that can describe the generation process for an observed signal through the statistical analysis of the random signal. Recently, as an identification method for such process, a method is considered in which the phase information of the signal using higher‐order cumulants is paid close attention. This paper proposes a method in which the observed one‐dimensinal signal is modeled as the output signal of a linear time‐invariant system generated with the non‐Gaussian white process. the generation systm for the signal (not constrained to be minimum phase) is identified using the third‐order cumulant of the observed signal. Because the system characteristics are decomposed into their components in the cepstral domain, the proposed method does not require the factorization procedure. In other words, the feature of the method is that it is extendable to the multidimensional system identification, for which the factorization generally is impossible.
On a decoding algorithm by superposition for binary cyclic codes for certain burst errors
Authors: Ogino Atsushi , Kohnosu Toshiyuki , Hirasawa Shigeichi , Nishijima Toshihisa
Published: July 2007

Cyclic codes are often used for the purpose of correcting burst errors in digital transmission and recording systems. Since the implementations of the encoding and syndrome calculations of these codes are simple, these codes have superior properties from a practical perspective. In this paper, an effective decoding method is given for certain burst errors (multiple solid‐burst errors) that are contained within the detected errors resulting from a binary cyclic “predecoding” proces, where the precoding process for burst error correction is terminated at the detection stage. Such burst errors can be expressed as random errors by superimposing a copy of itself that has been cyclically shifted by one bit. So, by performing random error decoding on a version of the received sequence that has been superimposed with a one‐bit cyclically shifted copy of itself, burst errors can be accurately corrected. This decoding method completely preserves the correction capacity of the precode and is applicable to any binary cyclic code with a minimum distance of at least 5. the correctable multiple solid‐burst errors depend on the minimum distance of the code. Furthermore, by applying this technique to interleaved random error‐correction codes for the purpose of correcting burst errors, it is shown that it is possible to extend the length of correctable burst errors.
Double‐talk detection method with detecting echo path fluctuation
Authors: Fujii Kensaku , Ohga Juro
Published: July 2007

In the echo canceller, where the normalized least mean square (NLMS) algorithm is used as the adaptive algorithm, the double‐talk disturbs the coefficients of the adaptive filter and increases the probability of howling generation. to cope with this phenomenon, the echo canceller usually is provided with a detector to detect the generation of the double‐talk. the echo canceller is designed to minimize the disturbance by stopping the update of the adaptive filter coefficients when the double‐talk is detected. In the detection of the double‐talk, it is effective to monitor the increase of the residual echo since the double‐talk component masked by the echo can be detected. In this method, however, the residual echo also is increased by the fluctuatio of the echo path and the fluctuation of the echo path should also be detected.

This paper proposes a method to differentiate between the double‐talk and the echo path fluctuation, based on the magnitude of the correlation between the input‐containing echo and the echo replica. In other words, the property is utilized that the correlation between the echo replica and the echo is large for the double‐talk and is small for the echo path fluctuation. A double‐talk detection method also is proposed where the power ratio of the residual echo and the echo replica is used as the parameter. It is shown that the double‐talk detection without deteriorating at all the echo elimination can be realized by a structure with a delay preceding the coefficient update operation.