In recently year, the analysis of power management becomes more important. It is difficult to obtain the maximum power because this is NP-complete. For an n-input circuit, there are 22n different input patterns to be considered. There are two major methods for this problem. First method is to generate input patterns to obtain the maximal power by simulating these generated patterns. This method is called pattern based. The other one uses probability method to estimate the power density of each node of a circuit to calculate the maximal power. In this paper, we use a pattern based method to estimate the maximal power. This method is better than that of probability for the simulation of power activity. In practical applications, these generated patterns can be applied and observe the activity of a circuit. These simulated data can be used to examined the critical paths for performance optimization. A simulated annealing algorithm is proposed to search input patterns for maximum power. Firstly, it transforms this problem into an optimization problem to adapt the simulated annealing method. In this method, there are three strategies for generating the next input patterns, called neighborhood. In the first strategy, it generates the next input pattern by changing the status of all input nodes. In the second strategy, some input nodes are selected and changed randomly.

In this paper, we propose a two-dimensional (2-D) least-squares lattice (LSL) algorithm for the general case of the autoregressive (AR) model with an asymmetric half-plane (AHP) coefficient support. The resulting LSL algorithm gives both order and space recursions for the 2-D deterministic normal equations. The size and shape of the coefficient support region of the proposed lattice filter can be chosen arbitrarily. Furthermore, the ordering of the support signal can be assigned arbitrarily. Finally, computer simulation for modeling a texture image is demonstrated to confirm the proposed model gives rapid convergence.

This paper analyzes mechanisms of radiated emissions from multilayer printed circuit boards (PCBs) and presents a model to describe the characteristics of such radiation. The radiation mechanism from a four-layer PCB, including the internal power and ground planes, is investigated using a time-domain magnetic field measurement near the PCB. Measurement of the waveform indicates that the main source of radiation is in the power distribution planes. To investigate the characteristics of the radiation from the power distribution, the S21s of the board are measured; the board impedance and the transmission characteristics of the power distribution planes are found to be directly related to the S21 between the two points in the board. The results indicate that the power distribution acts as a transmission line at frequencies higher than 100 MHz. A model that can explain well the radiation properties of these planes treats them as a parallel-plate transmission line interconnected by decoupling circuit comprising a decoupling capacitor and interconnect inductance. From the transmission line theory it is deduced that the line resonance gives rise to strong radiated emissions. The interconnect inductance is an important factor in determining the radiation characteristics.

We consider methods to locate sync words in packet or frame transmission over the additive white Gaussian noise channel. Our starting point is the maximization of the probability of correctly locating the sync word. We extend Massey's original result to the specific synchronization problem, where the sync words is prefixed to the data stream and each packet is preceded by idle transmission or additive white Gaussian noise. We give simulation results for several interesting sync words such as Barker sequences of length 7 and 13 and a sync word of length 17 with good cross-correlation properties. One of the conclusions is that the newly derived formula for the probability of correctly locating the sync word enables the reduction of the false sync detection probability.

This paper presents a new upper bound on overall bit error rate (BER) for a concatenated code which consists of an inner convolutional code and an outer interleaved Reed-Solomon code. The upper bound on BER is derived based on a lower bound on the effective minimum distance of the concatenated code. This upper bound can be used for the cases when the interleaver size is small such that the conventional upper bound is not applicable.

Concatenated codes have many remarkable properties from both the theoretical and practical viewpoints. The minimum distance of a concatenated code is at least the product of the minimum distances of an outer code and an inner code. In this paper, we shall examine some cases that the minimum distance of concatenated codes is beyond the lower bound and get the tighter bound or the true minimum distance of concatenated codes by using the complete weight enumerator of the outer code and the Hamming weight enumerator of the inner code. Furthermore we propose a new decoding method based on Reddy-Robinson algorithm by using the decoding method beyond the BCH bound.

Turbo codes have fascinated many coding researchers because of thier near-Shannon-limit error correction performance. In this paper, we discuss multi-dimensional turbo codes which are parallel concatenation of multiple constituent codes. The average upper bound to bit error probability of multidimensional turbo codes is derived. The bound shows that the interleaver gains of this kind of codes are larger than that of conventional two-dimensional turbo codes. Simplified structures of multi-dimensional turbo encoder and decoder are proposed for easier implementation. Simulation results show that for a given interleaver size, by increasing the dimension, great performance improvement can be obtained.

Irreducible components of canonical graphs for second order spectral null constraints at a rational submultiple of the symbol frequency fsk/n are studied where fs is the symbol frequency. We show that if n is prime then a canonical graph consists of disjoint irreducible components. We also show that the number of irreducible components of a canonical graphs is finite if n is prime. For the case n = 2 and p O mod n, all aperiodic irreducible components are identified explicitly where p is a parameter of a canonical graph.

A novel neural network architecture for image texture classification is introduced. The proposed model (Kernel Modifying Neural Network: KM Net) which incorporates the convolution filter kernel and the classifier in one, enables an automated texture feature extraction in multichannel texture classification through the modification of the kernel and the connection weights by the backpropagation-based training rule. The first layer units working as the convolution kernels are constrained to be an array of Gabor filters, which achieves a most efficient texture feature localization. The following layers work as a classifier of the extracted texture feature vectors. The capability of the KM Net and its training rule is verified using a basic problem on a synthetic texture image. In addition, the possibilities of applying the KM Net to natural texture classification and biological tissue classification using an ultrasonic echo image have been tried.

This paper deals with a probabilistic formulation of the diffraction and scattering of a plane wave from a periodic surface randomly deformed by a binary sequence. The scattered wave is shown to have a stochastic Floquet's form, that is a product of a periodic stationary random function and an exponential phase factor. Such a periodic stationary random function is then represented in terms of a harmonic series representation similar to Fourier series, where `Fourier coefficients' are mutually correlated stationary processes rather than constants. The mutually correlated stationary processes are written by binary orthogonal functionals with unknown binary kernels. When the surface deformations are small compared with wavelength, an approximate solution is obtained for low-order binary kernels, from which the scattering cross section, coherently diffracted power and the optical theorem are numerically calculated and are illustrated in figures.

Time-related jitters caused by small noise voltage due to electromagnetic noise induce malfunction of digital equipment. The jitters increase with not only magnitude of the noise but also resonance of digital circuits in the equipment. In this report, we proposes a linear equivalent circuit model of a digital CMOS gate for analyzing circuit resonance and verifies the validity of the model.

In this Letter, we investigate the loss performance of a discrete-time single-server queueing system with periodic vacations, with which we are often confronted in traffic control, such as cell scheduling or priority control schemes, at ATM nodes. Explicit expressions are derived for the cell loss ratio in terms of the distribution of the buffer contents in an infinite capacity queue.

The numerical property of the recursive least squares (RLS) algorithm has been extensively, studied. However, very few investigations are reported concerning the numerical behavior of the predictor-based least squares (PLS) algorithms which provide the same least squares solutions as the RLS algorithm. In Ref. [9], we gave a comparative study on the numerical performances of the RLS and the backward PLS (BPLS) algorithms. It was shown that the numerical property of the BPLS algorithm is much superior to that of the RLS algorithm under a finite-precision arithmetic because several main instability sources encountered in the RLS algorithm do not appear in the BPLS algorithm. This paper theoretically shows the stability of the BPLS algorithm by error propagation analysis. Since the time-variant nature of the BPLS algorithm, we prove the stability of the BPLS algorithm by using the method as shown in Ref. [6]. The expectation of the transition matrix in the BPLS algorithm is analyzed and its eigenvalues are shown to have values within the unit circle. Therefore we can say that the BPLS algorithm is numerically stable.

A family of nonce-based authentication and key distribution protocols based on the trusted third-party model are proposed which are not only efficient on the view points of computation and communication, but also secure against on-line and off-line password guessing attacks. A new concept of implicit or indirect challenge-response authentication which can be used to combine the processes of identify authentication and data integrity assurance during key distribution and to make the entire protocol be more concise and efficient is introduced in this paper. In the proposed family of protocols, specific protocol can be chosen such that the secure session key to be distributed is selected by specific participant in the protocol. Detailed security analyses of every protocols are given.

This paper describes relation between the number of library pairs and error probability to have all the pairs as fixed points for second-order bidirectional associative memory (BAM). To estimate accurate error probability, three methods have been compared; (a) Gaussian approximation, (b) characteristic function method, and (c) Hermite Gaussian approximation (proposed by this paper). Comparison shows that Gaussian approximation is valid for the larger numbers of neurons in both two layers than 1000. While Hermite Gaussian approximation is applicable for the larger number of neurons than 30 when Hermite polynomials up to 8th are considered. Capacity of second-order BAM at the fixed error probability is estimated as the function of the number of neurons.

The redundancy of universal lossy data compression for discrete memoryless sources is considered in terms of type and d-ball covering. It is shown that there exists a universal d-semifaithful code whose rate redundancy is upper bounded by (A-1/2)n-1ln n+o(n-1ln n), where A is the cardinality of source alphabet and n is the block length of the code. This new bound is tighter than known ones, and moreover, it turns out to be the attainable minimum of the universal coding proposed by Davisson.

Recently, the low earth orbit satellite communications has been attracting much attention. These communications have many strong features, however, the communication performances are influenced by carrier frequency offset (CFO) and, particularly, it is hard to acquire the synchronization. A large number of publications have so far been made on the synchronization acquisition of DS/SS systems under CFO and most of them make use of the maximum likelihood decision in finding the maximum values of Fourier transform outputs. However, the implementations of Fourier transforms usually require high cost and large space. In this paper, we propose a new simple acquisition scheme using half-symbol differential decoding technique for DS/SS systems under CFO. This scheme makes use of the addition and subtraction of baseband signals and their delayed versions, (omitting Fourier transforms), together with integrations by recursive integrators, and thus resulting in much simpler implementation. In general, it is shown that the proposed scheme can acquire the code synchronization under carrier frequency offset with much smaller computational complexities and the sacrifice of longer acquisition time.

In this paper, we examine a new initial symbol acquisition method for M-ary spread-spectrum (M-ary/SS) signals that are affected by large carrier frequency offset. By the effect of the carrier frequency offset, preamble signal energy is dispersed to the undersired outputs. The proposed method is based on the collection of such dispersed signal energies by using reference patterns. The reference patterns are constructed by using the characteristic of Hadamard code sequences. The effectiveness of the proposed method is evaluated in terms of mean acquisition time.

Bit-interleaving can enhance performance of a trellis coded modulation system over a fading channel. A combined system with decision feedback equalization is proposed. In the system, TCM decoded symbols are fed back for equalization. To avoid a bad effect of decoding delay, a deinterleaver is utilized effectively. Information sequence is divided into three subsequences and encoded by three encoders. Among the 3 code vectors from the encoders, bits are interleaved and decoding proceeds in parallel. Simulation results show that the proposed system realizes 0.6 dB more coding gain than a symbol interleaved system. A calculation method of a branch metric for decoding is proposed. Performance with the branch metric is shown to be nearly independent from the desired/undesired power ratio of a intersymbol interference channel. An approximate upper bound is analyzed for the proposed system, and the optimum code is searched.

Since spontaneously spoken language expressions appear continuously, the transfer stage of a spoken language machine translation system have to work incrementally. In such the system, the high degree of incrementality is also strongly required rather than that of quality. This paper proposes an incremental machine translation system, which translates English spoken words into Japanese in accordance with the order of appearances of them. The system is composed of three modules: incremental parsing, transfer and generation, which work synchronously. The transfer module utilizes some features and phenomena characterizing Japanese spoken language: flexible wordorder, ellipses, repetitions and so forth. This in influenced by the observational facts that such characteristics frequently appear in Japanese uttered by English-Japanese interpreters. Their frequent utilization is the key to success of the exceedingly incremental translation between English and Japanese, which have different word-order. We have implemented a prototype system Sync/Trans, which parses English dialogues incrementally and generates Japanese immediately. To evaluate Sync/Trans we fave made an experiment with the conversations consisting of 27 dialogues and 218 sentences. 190 of the sentences are correct, providing a success rate of 87.2%. This result shows our incremental method to be a promising technique for spoken language translation with acceptable accuracy and high real-time nature.