We frequently use a polynomial to approximate a complex function. This study shows a method which determines the optimum coefficients and the number of terms of the polynomial, and the error of the polynomial is estimated.

In the data path circuits of asynchronous systems, logical faults may first manifest as undetectable, transient wrong codewords, in spite of encoding the inputs and the outputs and proper organization which enables the faults to be propagated to the primary outputs in the form of non-codewords. Due to this, the conventional methods of concurrent error detection (CED) using the logic (voltage) monitoring is not effective. In this paper, we suggest a mixed-signal approach to achieve CED for a class of asynchronous circuits, known as self-timed circuits. First, we show that it is impossible to guarantee the CED using logic monitoring of the primary outputs in spite of proper encoding and organization of self-timed circuits. Then, we discuss different manifestations of single stuck-at faults occurring during normal operation in these circuits. Finally, we present the feasibility of achieving CED using a built-in current sensor (BICS) along with encoding techniques.

In the conventional sound field reproduction system with control of the transfer functions from the source to both ears of a listener, a slight shift of the ears caused by movement of the listener inevitably results in sound localization being different from that expected. In this paper, a method for reproducing a sound field by controlling the transfer function from the source to multiple points (called the "method of multiple-points control" hereafter) is applied to a sound reproduction system with the aim of expanding the area which can be controlled. The system is controlled so that the transfer functions from the input of the system to the multiple points adjacent to the original receiving points have the same desired transfer function. By placing the control points at appropriate intervals, a "zone of equalization" is formed. Based on a computer simulation, the intervals between control points is discussed. The configuration of the loundspeakers for sound reproduction is also discussed.

Heavy-ion-induced soft errors (single event upset) in submicron silicon-on-insulator (SOI) MOSFETs under space environmental conditions are studied over the temperature range of 100-400 K using three-dimensional device simulator with full-temperature models. The temperature dependence of the drain collected charge is examined in detail when a heavy-ion strikes the gate center perpendicularly. At very low temperatures, SOI MOSFETs have very high immunity to the heavy-ion-induced soft errors. In particular, alpha-particle-induced soft errors hardly occur at temperatures below 200 K. As the temperature increases, the collected charge shows a marked rate of increase. The problem of single event upset in SOI MOSFETs becomes more serious with increasing working temperature. This is because the induced bipolar mechanism is a main factor to cause charge collection in SOI MOSFETs and the bipolar current increases exponentially with increasing temperature. At room and high temperatures, the drain collected charge is strongly dependent on channel length and SOI film thickness.

In this paper, we analyze the multiple access interference of a variable processing gain DS/CDMA system and define discrete partial crosscorrelation functions. We also evaluate the bit error rate of the system using Gaussian approximation and bounding technique. Three kinds of spreading codes (long, short, and random codes) are considered in the analysis of the system. It is shown that the bit error rate of a user is not relevant to the processing gain of interfering users: it is relevant only to the processing gain of the user, transmitted powers, PN sequences, and spreading codes. The performance of short codes turns out to be better than that of long and random codes as in other systems.

This paper evaluate the performance of a PN (pseudonoise) code acquisition for a direct-sequence/code-division-multiple-access (DS/CDMA) system in a mobile satellite environment. The acquisition scheme considered consists of a parallel matched-filter and a FFT processor. The uplink of mobile satellite channel is modeled as shadowed Rayleigh fading channel. The effects of power control error and shadowing are considered in the analysis of acquisition performance. It is shown that the power control error causes acquisition to be slower than the case of perfect power control, and for high SNR/chip, the effect of power control error becomes less significant. It is also shown that the case with heavy shadowing takes longer time to achieve acquisition than that with light shadowing. For the subinterval-based PN code search, the parallel MF scheme is thought to be more appropriate than the serial MF scheme at the cost of complexity. The analysis in the paper can be applied to the uplink of a DS/CDMA system for packet-type services in a mobile satellite channel.

Roundoff error due to iterative computation with finite wordlength degrades the quality of decoded images in fractal image coding that employs a deterministic iterated function system. This paper presents a state-space approach to roundoff error analysis of fractal image coding for grey-scale images. The output noise variance matrix and the noise matrix are derived for the measures of error and the output noise variance is newly defined as the pixel mean of diagonal elements of the output noise matrix. A quantitative comparison of experimental roundoff error with analytical result is made for the output noise variance. The result shows that our analysis method is valid for the fractal image coding. Our analysis method is useful to design a real-time and low-cost decoding hardware with finite wordlength for fractal image coding.

In this paper, we evaluate the throughput performance of CDMA Slotted ALOHA systems. To improve the throughput performance, we employ the Quasi-synchronous sequences and the Modified Channel Load Sensing Protocol as an access control procedure. As a result, we found a good throughput by the QS-sequences. By employing MCLSP, we can keep the maximum throughput even in high offered load and in the presence of a long access timing delay, which is one of the issue in satellite packet communication systems.

Hankel's asymptotic expansions are frequently used for numerical calculation of cylindrical functions of complex order. We beforehand need to estimate the precisions of the cylindrical functions calculated with Hankel's asymptotic expansions in order to use these expansions. This letter presents comparatively simple expressions for rough estimations of the errors of the cylindrical functions calculated with the asymptotic expansions, and features of the errors are discussed.

In this paper stochastic aradient adaptive filters using the Sign or Sign-Sign Algorithm are analyzed based upon general assumptions on the reference signal, additive noise and particularly jointly distributed tap errors. A set of difference equations for calculating the convergence process of the mean and covariance of the tap errors is derived with integrals involving characteristic function and its derivative of the tap error distribution. Examples of echo canceller convergence with jointly Gaussian distributed tap errors show an excellent agreement between the empirical results and the theory.

In this paper, the bit error rate (BER) considering tracking performance is evaluated, by theoretical analysis and computer simulation, for a bi-orthogonal system using a synchronizing pseudo-noise (PN) sequence and co-channel interference cancellers. A system that improves on Tachikawa's system is proposed. It is found that the optimum ratio of the information signal energy to the synchronizing signal energy varies with Eb/No, and the canceller is better for small L than for large L (L = length of the sequence). Moreover, it is found that the BER considering synchronization performance improvse as the equivalent noise bandwidth Bn decreases.

The Time-Slicing paradigm is a newly developed method for the training of neural networks for speech recognition. The neural net is trained to spot the syllables in a continuous stream of speech. It generates a transcription of the utterance, be it a word, a phrase, etc. Combined with a simple error recovery method the desired units (words or phrases) can be retrieved. This paradigm uses a recurrent neural network trained in a modular fashion with natural connectionist glue. It processes the input signal sequentially regardless of the input's length and immediately extracts the syllables spotted in the speech stream. As an example, this character string is then compared to a set of possible words, picking out the five closest candidates. In this paper we describe the time-slicing paradigm and the training of the recurrent neural network together with details about the training samples. It also introduces the concept of natural connectionist glue and the recurrent neural network's architecture used for this purpose. Additionally we explain the errors found in the output and the process to reduce them and recover the correct words. The recognition rates of the network and the recovery rates for the words are also shown. The presented examples and recognition rates demonstrate the potential of the time-slicing method for continuous speech recognition.

This paper describes an adaptive neural vector quantization algorithm with a deleting approach of weight (reference) vectors. We call the algorithm an adaptive learning and self-deleting algorithm. At the beginning, we introduce an improved topological neighborhood and an adaptive vector quantization algorithm with little depending on initial values of weight vectors. Then we present the adaptive learning and self-deleting algorithm. The algorithm is represented as the following descriptions: At first, many weight vectors are prepared, and the algorithm is processed with Kohonen's self-organizing feature map. Next, weight vectors are deleted sequentially to the fixed number of them, and the algorithm processed with competitive learning. At the end, we discuss algorithms with neighborhood relations compared with the proposed one. The proposed algorithm is also good in the case of a poor initialization of weight vectors. Experimental results are given to show the effectiveness of the proposed algorithm.

Optical Frequency Division Multiplexing (OFDM) is an attractive multiplexing approach for exploiting optical communication technology. Although considerable progress has been made in this approach, it still suffers from numerous potential impairments, stemming from several phenomena. (i.e., laser unstability, residual temperature variations, linear and nonlinear cross talk.). Conventional serial coding technique is not practical in lightwave systems, as it changes the system's bit rate that is not desirable. In this paper a new Parallel Coded Optical Multicarrier Frequency Division Multiplexing (PCOM-FDM) technique has been investigated. The strategy of multicarriers, together with Parallel Forward Error Control (PFEC) coding, is a potentially novel approach as in this approach we have, 1) Investigated optical multicarrier communication that is effective in combating dispersion and increasing throughput, 2) Proposed PFEC coding which is different from conventional serial coding in respect that it does not change the system bit rate per carrier and prevents the effects of channel wandering. It is highly desirable in lightwave systems and thus holds a vital importance in practical high speed optical communication systems. Theoretical treatment shows that the proposed approach is promising and practical.

The performance improvement of the partial response maximum-likelihood (PRML) system for (1, 7) run-length limited (RLL) code is studied. As a new PRML system, PR (1, 1, 0, 1, 1) system called modified E2PR4 (ME2PR4 ) followed by Viterbi detector for (1, 7) RLL code is proposed. At first, a determination method of the tap weights in transversal filter to equalize to PR (1, 1, 0, 1, 1) characteristic taking account of a noise correlation is described. And the equalization characteristics of the transversal filter are evaluated. Then, a Viterbi detector for ME2PR4 utilizing the constraint of run-length of (1, 7) RLL code is presented. Finally, the bit-error rate is obtained by computer simulation and the performance is compared with that of the conventional PRML systems called PR4, EPR4 and E2PR4 systems with Viterbi detector. The results show that among these systems our system exhibits the best performance and the SNR improvement increases with the increase in the linear density.

Moving-picture transmission through narrow band and high bit error rate communication channels, such as a mobile communication channel, requires improved compression rate and enhanced error resilience. Variable-length codes are one of the essential techniques of compressing digital video information. This technique is used in various video coding schemes although a bit error in the channel impairs the synchronization of variable-length codewords, resulting in propagation of the error. With a hybrid video coding method in particular, which combines motion-compensation and transform coding, once an error is detected in the coded data, subsequent data cannot be decoded. Consequently, even an error-free portion of any data received must be discarded. To minimize the influence of an error in a channel on coded video data, this paper proposes a new video coding syntax which makes the best use of the self synchronizing characteristic of variable-length Huffman codes. Owing to the Huffman code's characteristic, the proposed coding syntax enables a decoder to decode the data portion that cannot be decoded, due to an error, by the conventional syntax without adding any redundancy. Computer simulation has verified the effectiveness of this proposed syntax in video coding with a very low bitrate and erroneous communication channel.

It is thought that we have generally succeeded in establishing learning algorithms for neural networks, such as the back-propagation algorithm. However two major issues remain to be solved. First, there are possibilities of being trapped at a local minimum in learning. Second, the convergence rate is too slow. Chang and Ghaffar proposed to add a new hidden node, whenever stopping at a local minimum, and restart to train the new net until the error converges to zero. Their method designs newly generated weights so that the new net after introducing a new hidden node has less error than that at the original local minimum. In this paper, we propose a new method that improves their convergence rate. Our proposed method is expected to give a lower system error and a larger error gradient magnitude than their method at a starting point of the new net, which leads to a faster convergence rate. Actually, it is shown through numerical examples that the proposed method gives a much better performance than the conventional Chang and Ghaffar's method.

In 1989, Ahlswede and Dueck introduced a new formulation of Shannon theory called identification via channels. This paper presents a simple construction of codes for identification via channels when the probability of false identification is measured by its average. The proposed code achieves the identification capacity, and its construction does not require any knowledge of coding theory.

An algorithm for finding the unequal error protection (UEP) capability of a q-ary image of a low-rate qm-ary cyclic code is presented by combining its concatenated structure with the UEP capability of concatenated codes. The results are independent of a choice of a basis to be used for expanding an element over GF (qm) into GF (q). A table of the UEP capability of binary images of low-rate Reed-Solomon codes over GF (26) is given. It is shown that the encoding and decoding algorithms for superimposed concatenated codes can be applied to a q-ary image as a linear UEP code.

The performance of APP (a posteriori probability) decoding algorithm which is well known as a soft decision decoding algorithm for majority logic decodable codes is further improved by iterating the algorithm one or more times. This letter shows that there exists the optimal non-zero threshold value of the decision function that minimizes the decoded error rate in two-pass APP decoding though the optimal threshold value in one-pass APP decoding is zero.