bluuebunny/crossref_metadata_2025_split huggingface dataset:
Parallel‐form realization of 2‐D adaptive separable‐denominator state‐space filters
Authors: Muneyasu Mitsuji , Hinamoto Takao 
Published: July 2007

AbstractThis paper treats the parallel‐form realization of two‐dimensional (2‐D) adaptive state‐space filters whose transfer functions are separable in the denominator. the adaptive filters are described by the Roesser local statespace (LSS) model, and the least mean squares (LMS) method is used here to adjust the filter parameters.First, the Roesser LSS model with separable denominator is realized in a parallel form. Then the stability condition of the filter is shown. Next, the LMS method is employed to adjust the filter parameters in the parallel form. In addition, the computational time and the number of processors required for the realization are investigated. Finally, the adaptive filter is applied in numerical examples to the 2‐D spatial‐domain filter design and to the noise reduction of an image.
An oversampling subband adaptive digital filter with rational decimation rations
Authors: Kiya Hitoshi , Yamazaki Hirotatsu , Ashihara Koji 
Published: July 2007

AbstractRecent research has been conducted on the subband adaptive filters wherein filter banks are used in the adaptive filters. However, when a complete reconstruction filter bank is used for subband coding, then the effect of aliasing caused by decimation cannot be avoided in subband adaptive digital filter (ADF). One of the solutions to this problem is the oversampling scheme where maximum decimation is not used. In this paper, a filter bank is proposed that allows us to choose arbitrary rational numbers as decimation ratios. As a special case, they include the integral decimation ratio (which may be considered as the general form of filter bank in oversampling scheme). This filter bank simultaneously maintains the advantages of oversampling scheme and makes possible the realization of a subband adaptive system that has a close to maximum decimation ratio with the reduction of filter order. In this paper, the design conditions of the filter bank are first formulated. Based on these design conditions, one method to design the filter banks is proposed. to confirm the effectiveness of this filter bank, the system identification examples are then simulated.
Use of a priori information in generalized cepstral modeling of degraded speech
Authors: Kanno Toshio , Kobayashi Takao , Imai Satoshi 
Published: July 2007

AbstractThis paper proposes an improved form of generalized cepstral modeling for estimating speech parameters in a noisy environment. The original formulation of the technique based on maximum a posteriori (MAP) estimation procedure uses no a priori information of speech signal. In the improved technique, an initial estimate of speech spectrum is given by using a priori signal‐to‐noise ratio (SNR) and then an iterative procedure is performed.Several modifications on the iterative procedure also are discussed to reduce residual noise in enhanced speech and spectral distortion of estimated spectrum. A speech enhancement system based on the proposed technique is developed and several experimental results are shown.
Performance prediction of word recognition using the probability of word occurrence
Authors: Otsuki Takashi , Otomo Teruhiko , Ito Akinori , Makino Shozo 
Published: July 2007

AbstractThe words in natural language have different occurrence probabilities. Consequently, the information obtained from the event, i.e., the occurrence of a word, is larger than in the case of the occurrence with uniform probability. In other words, it will be effective to utilize the occurrence probability of the word in the recognition and it is sensible to examine its error‐correcting ability.This paper considers the situation where the word occurrence probability is used in the word recognition process and presents a method to estimate the relation between the phoneme/character recognition score and the word recognition score. In the past derivation of the evaluation formula for the word recognition score, it is assumed that the word occurrence probability is uniform for whole words and the difference is ignored.From such a viewpoint, this paper derives the evaluation formula considering the word occurrence probability. By comparing the value estimated by the derived evaluation formula and the value obtained by the simulation for the word recognition, it is found that there is a considerable error due to the approximation and the word recognition score is estimated as approximately 10 percent lower for the phoneme recognition score of 80 percent. Then the approximation procedure is modified and an evaluation formula containing a correction factor is derived. the difference between the value estimated by the corrected evaluation formula and the value obtained by simulation is less than 5 percent for the phoneme recognition score of 80 percent. In other words, the precision is improved and the word recognition score, when the word occurrence probability is utilized, can be estimated accurately.
Method of suppressing burst noise interference in images using smearing‐desmearing transform
Authors: Ohue Kenji , Takabayashi Akifumi 
Published: July 2007

AbstractTo realize high‐quality image transmission and storage, it is important to reduce the effects of the impulsive noise and the image data erasure produced in the transmission and storage. This paper proposes a new method for image noise suppression using the smearing‐desmearing transform.The smearing‐desmearing transform retains the power spectrum of the image signal and can be applied to the highly efficient coding. the basic principle for the image noise suppression using the smearing‐desmearing transform is shown and the construction of the suitable smearing‐desmearing transform is presented. Then, it is shown that the noise suppression performance of the image noise suppression method based on the smearing‐desmearing transform depends on the variance of the noise, even if there exist the same number of noises. Based on such a property, a method to suppress the burst noise in the image based on the interleaving‐deinter‐leaving and smearing‐desmearing transform is proposed where the smearing‐desmearing transform is applied after spreading the burst noise to fit to the image noise suppression process. Finally, by a computer experiment, it is shown that the proposed method has an excellent effect in image noise suppression.
Design technique of cellular neural network
Authors: Nakai Koji , Ushida Akio 
Published: July 2007

AbstractThe cellular neural network (CNN) is composed of a planar placement of cells which consists of a (piecewise‐linear) nonlinear element and controlled current sources. It features a simple structure which is close to that of the retina and is expected to be utilized in pattern recognition and image processing.In CNN, each cell is connected to the neighborhood cells by the same pattern, and by adjusting the connection pattern, CNN with various functions can be designed. the connection pattern is called the cloning template. It is very important in the development of new CNN to establish the design method for the cloning template.In this paper, the operational characteristics are assumed so that the output for the normative input satisfies the specified conditions. the constraint is derived from the specification and the template is designed so that the corresponding cost function is minimized. the simplex method is used as the optimization technique, which features a simple algorithm. As application examples, noise‐remover CNN as well as the maze‐tracing CNN are designed and satisfactory results are obtained. the design method is reported in this paper.
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

AbstractThis 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

AbstractAssume 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

AbstractIn 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

AbstractA 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

AbstractAlthough 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

AbstractCryptology 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

AbstractThis 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

AbstractThis 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

AbstractIn 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

AbstractA 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

AbstractAn 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

AbstractA 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

AbstractThe 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.