This paper describes the waveform relaxation (WR) algorithm with the under relaxation method based on the virtual state formulation (VSF) technique and the effect of multirate behavior in this algorithm. First, we present the virtual state relaxation method using VSF technique. Next, we introduce the VSF method into WR algorithm in order to exploit the multirate behavior. Furthermore, we construct the relaxation-based circuit simulator DESIRE2 and apply this simulator to the transient analysis of MOS circuits. Finally, we show that the present technique enables to use efficiently the multirate integration method in VSR and reduce the total simulation time without losing the waveform accuracy.

Despite its potential to realize image communication at extremely low rates, model-based coding (analysis-synthesis coding) still has problems to be solved for any practical use. The main problems are the difficulty in modeling unknown objects and the presence of analysis errors. To cope with these difficulties, we incorporate waveform coding into model-based coding (model-based/waveform hybrid coding). The incorporated waveform coder can code unmodeled objects and cancel the artifacts caused by the analysis errors. From a different point of view, the performance of the practically used waveform coder can be improved by the incorporation of model-based coding. Since the model-based coder codes the modeled part of the image at extremely low rates, more bits can be allocated for the coding of the unmodeled region. In this paper, we present the basic concept of model-based/waveform hybrid coding. We develop a model-based/MC-DCT hybrid coding system designed to improve the performance of the practically used MC-DCT coder. Simulation results of the system show that this coding method is effective at very low transmission rates such as 16kb/s. Image transmission at such low rates is quite difficult for an MC-DCT coder without the contribution of the model-based coder.

Quantization of the impulse response coefficients due to finite word length causes the moments to deviate from their ideal values. This deviation is found to have a linear variation with the output roundoff noise of the filter realized in direct form. Since the zeros and poles of a given filter also move away from their designed locations due to quantization, we show a relation between the zeros and poles and the moments of the impulse response.

In recent years, the digitalization of transmission links, such as optical fibre cables, satellite links, and terrestrial microwave links, has been progressed rapidly in many countries. In addition, many types of digital studio equipment have been developed and TV programs can be produced or edited without any picture quality degradation by using such equipment, for example, digital VTR. A high-efficiency bit-reduction coding system is the most promising and effective means for this situation in terms of reducing the cost of digital transmission of TV programs with high picture quality. Considering this background, a new digital coding system has been developed, which makes it possible to transmit up to 4 NTSC TV programs simultaneously over a single DS3 45Mbps link including two high quality sound channels and one 64kbps ancillary data channel for each TV program. The principal bit-reduction technique employed is 2 dimensional intraframe WHT (Walsh Hadamard Transform) coding, which gives higher coding performance for composite TV signals than DCT (Discrete Cosine Transform) coding. In order to attain high picture quality at around 8Mbps for 4 channel transmission, a 3 dimensional adaptive quantization cube which reflects human visual perception sufficiently is employed in the intraframe WHT coding scheme. The hardware has been made compact like a home use VTR. In this paper, first, the algorithm of the coding scheme developed for the coding system is presented, and then the system configuration and its basic coding performance are described.

High-speed digital land mobile communications suffer from frequency-selective fading due to a long delay difference. Several techniques have been proposed to overcome the multipath propagation problem. Among them, an adaptive array antenna is suitable for very high-speed transmission because it can suppress the multipath signal of a long delay difference significantly. This paper describes the LMS adaptive array antenna for frequency-selective fading reduction and a new diversity technique. First, we propose a method to generate a reference signal in the LMS adaptive array. At the beginning of communication, we use training codes for the reference signal, which are known at a receiver. After the training period, we use detected codes for the reference signal. We can generate the reference signal modulating a carrier at the receiver by those codes. The carrier is oscillated independently of the incident signal. Then, the carrier frequency of the reference signal is in general different from that of the incident signal. However, the LMS adaptive array works in such a way that the carrier frequency of the array output coincides with that of the reference signal. Namely, the frequency difference does not affect the performance of the LMS adaptive array. Computer simulations show the proper behavior of the LMS adaptive array with the above reference signal generator. Moreover, we present a new multipath diversity technique using the LMS adaptive array. The LMS adaptive array reduces the frequency-selective fading by suppressing the multipath components. This means that the transmitted power is not used sufficiently. We propose a multiple beam antenna with the LMS adaptive array. Each antenna pattern receives one of the multipath components, and we combine them adjusting the timing. Then, we realize the multipath diversity. In addition to the multipath fading reduction, we can improve a signal-to-noise ratio by the diversity technique.

Threre are two types of nonlinear associative silicon retinas. One is a sparse Hopfield type neural network which is called a H-type retina and the other is its dual network which is called a DH-type retina. The input information sequences of H-type and HD-type retinas are given by nodes and links as voltages and currents respectively. The error correcting capacity (minimum basin of attraction) of H-type and DH-type retinas is decided by the minimum numbers of links of cutset and loop respectively. The operation principle of the regeneration is based on the voltage or current distribution of the neural field. The most important nonlinear operation in the retinas is a dynamic quantization to decide the binary value of each neuron output from the neighbor value. Also, the edge is emphasized by a line-process. The rates of compression of H-type and DH-type retinas used in the simulation are 1/8 and (2/3) (1/8) respectively, where 2/3 and 1/8 mean rates of the structural and binarizational compression respectively. We could have interesting and significant simulation results enough to make a chip.

A new error rate formula for narrowband Differential Quaternary Phase Shift Keyed system in a Rayleigh fading channel is obtained in closed-form. The formula predicts a non-zero error probability for noiseless reception. As predicted, the computed error rates approach some constant or floor values as the signal-to-noise ratio is increased beyond a certain limit. In the presence of various Doppler frequency shifts, an IF filter bandwidth of about one times the symbol rate is found to lead to a minimum error probability prior to the appearence of the error rate floor.

A new set of self-consistent linear equations is presented for the analysis of the startup characteristics of gyrotron oscillators with an open cavity consisting of weakly irregular waveguides. Numerical results on frequency detuning and oscillation starting current for a whispering-gallery-mode gyrotron are described in which these equations were utilized. Experiments for making a check on the effectiveness of the derived equations showed that they well express the operation of gyrotrons in comparison with the linear theory using an empty cavity field as the wave field.

Very high resolution images with more than 2,000*2.000 pels will play a very important role in a wide variety of applications of future multimedia communications ranging from electronic publishing to broadcasting. To make communication of very high resolution images practicable, we need to develop image coding techniques that can compress very high resolution images efficiently. Taking the channel capacity limitation of the future communication into consideration, the requisite compression ratio will be estimated to be at least 1/10 to 1/20 for color signals. Among existing image coding techniques, the sub-band coding technique is one of the most suitable techniques. With its applications to high-fidelity compression of very high resolution images, one of the major problem is how to encode high frequency sub-band signals. High frequency sub-band signals are well modeled as having approximately memoryless probability distribution, and hence the best way to solve this problem is to improve the quantization of high frequency sub-band signals. From the standpoint stated above, the work herein first compares three different scalor quantization schemes and improved permutation codes, which the authors have previously developed extending the concept of permutation codes, from the aspect of quantization performance for a memoryless probability distribution that well approximates the real statistical properties of high frequency sub-band signals, and thus demonstrates that at low coding rates improved permutation codes outperform the other scalor quatization schemes and that its superiority decreases as its coding rate increases. Moreover, from the results stated above, the work herein, develops a rate-adaptive quantization technique where the number of bits assigned to each subblock is determined according to the signal variance within the subblock and the proper quantization scheme is chosen from among different types of quantization schemes according to the allocated number of bits, and applies it to the high-fidelity encoding of sub-band signals of very high resolution images to demonstrate its usefulness.

This paper proposes a Mean-Separated and Normalized Vector Quantizer with edge-Adaptive Feedback estimation and variable bit rates (AFMSN-VQ). The basic idea of the AFMSN-VQ is to estimate the statistical parameters of each coding block from its previous coded blocks and then use the estimated parameters to normalize the coding block prior to vector quantization. The edge-adaptive feedback estimator utilizes the interblock correlations of edge connectivity and gray level continuity to accurately estimate the mean and standard deviation of the coding block. The rate-variable VQ is to diminish distortion nonuniformity among image blocks of different activities and to improve the reconstruction quality of edges and contours to which the human vision is sensitive. Simulation results show that up to 2.7dB SNR gain of the AFMSN-VQ over the non-adaptive FMSN-VQ and up to 2.2dB over the 1616 ADCT can be achieved at 0.2-1.0 bit/pixel. Furthermore, the AFMSN-VQ shows a comparable coding performance to ADCT-VQ and A-PE-VQ.

The outputs of all gates in a circuit are assumed to be observable unber the highly observable condition, which is mainly based on the use of E-beam testers. When using the E-beam tester, it is desirable that the test set for a circuit is small and the test vectors in the test set can be applied in a successive and repetitive manner. For a combinational circuit, these requirements can be satisfied by modifying the circuit into a k-UCP circuit, which needs only a small number of tests for diagnosis. For a sequential circuit, however, even if the combinational portion has been modified into a k-UCP circuit, it is impossible that the test vectors for the combinational portion can always be applied in a successive and repetitive manner because of the existence of feedback loops. To solve this problem, the concept of k-UCP scan circuits is proposed in this paper. It is shown that the test vectors for the combinational portion in a k-UCP scan circuit can be applied in a successive and repetitive manner through a specially constructed scan-path. An efficient method of modifying a sequential circuit into a k-UCP scan circuit is also presented.

The encoding of high quality and super high definition images requires new approaches to the coding problem. The nature of such images and the applications in which they are used prohibits the introduction of perceptible degradation by the coding process. In this paper, we discuss techniques for the perceptually transparent coding of images. Although technically lossy methods, images encoded and reconstructed using these techniques appear identical to the original images. The reconstructed images can be postprocessed (e.g., enhanced via anisotropic filtering), due to the absence of structured errors, commonly introduced by conventional lossy methods. The compression, ratios obtained are substantially higher than those achieved using lossless means.

A frame-installed lightwave synthesizer is constructed for optical frequency-division-multiplexing (FDM) communication. The synthesizer consists of two DFB diode lasers, electrical feedback loops, and an HCN gas cell used as a frequency reference at v0=192,843GHz (1.55459µm in wave-length). Output frequency can be stabilized at anywhere within v0(220) GHz. The beat note observed between the synthesizer and another HCN-stabilized DFB laser is constant within 2MHz over 100 hours. Frequency stability better than 410-10 (80kHz, without normalization) is obtained for an averaging time of 200s.

It is known that the uniquely decodable code pairs (C1, C2) for the two-user binary adder channel relates to the maximum independent set of a graph associated with a binary code. This paper formulates the independence number of a class of graphs associated with binary linear codes, and presents an algorithm of the maximum independent set for those graphs. Uniquely decodable code pairs (C1, C2)'s are produced, where C1 is a linear code and C2 is a maximum independent set of the graph associated with C1. For the given C1, the transmission rate of C2 is higher than that by Khachatrian, which is known as the best result as so far. This is not rather surprising because the code C2 is a maximum independent set in this paper but not be Khachatrian's.

A closure test MCT (meet-in-the-middle closure test) has been introduced to analyze the algebraic properties of cryptosystems. Since MCT needs a large amount of memory, it is hard to implement with an ordinary meet-in-the-middle method. As a feasible version of MCT, this paper presents a switching closure test SCT based on a new memoryless meet-in-the-middle method. To achieve the memoryless method, appropriate techniques, such as expansion of cycling detection methods for one function into a method for two functions and an efficient intersection search method that uses only a small amount of memory, are effectively used.

This paper examines the key technologies and applications of optical frequency division multiplexing (OFDM) systems. It is clarified that a 100-channel OFDM system is feasible as a result of multichannel frequency stabilization, common optical amplification and channel selection utilizing a tunable optical filter. Transmission limitation due to fiber four-wave mixing is also described. Major functions and applications of the OFDM are summarized and the applicability of OFDM add/drop multiplexing is examined.

It is sometimes required to change the frequency characteristics of a digital filter during its operation. In this paper a new synthesis of variable even-order IIR digital filters is proposed. The cut-off frequency of the filter can be changed by a single parameter. The fundamental filter structure is a cascade of second-order sections. The multiplier coefficients of each section are determined by using the Taylor series expansion of the lowpass to lowpass frequency transformation. For this method any second-order section can be used as a prototype, but here in this paper only the direct form and the lattice form are described. Unlike the conventional method, any transfer functions can be used for the proposed method. Finally a designed example shows that the proposed filter has wider tuning range than the conventional filter, and the advantage of the proposed filters is confirmed.

LOTOS is a language developed within ISO for the formal description of communication protocols and distributed systems. In LOTOS, requirements for a distributed system are called a "service specification". Each node exchanges synchronization messages to ensure the temporal ordering for the execution of events in a service specification. The actions of each node are described as a "protocol specification". This paper gives a survey for a method to derive protocol specifications from a service specification written in a LOTOS based language. In order to derive the protocol specifications, we make the syntax tree of a given service specification and give some attributes for each node in the tree. The protocol specifications are derived automatically by evaluating these attributes. The derived protocol specifications satisfy the given service specification. We also explain a LOTOS simulator for the execution of derived protocol specifications. The related works are also summarized.

It has been recognized that there exist some disparities between properties of continuous control systems and those of discrete ones which are obtained from their continuous counterparts by use of a sampler and zero order hold. This still remains true even if the sampling rate becomes fast enough and sometimes causes unfavorable effects in control systems design. To reconcile with this conflict, use of delta operator has been proposed in place of z-operator recently. This note formulates a delta domain Lyapunov matrix equation and shows that the equation actually mediates the discrete Lyapunov equation and its continuous counterpart.

This paper proposes a designing algorithm for quadrilateral recursive filters which consist of four quarter-plane filters in the four quadrants. This can realize a perfect zero-phase filtering which is essential for image processing. Furthermore, several parallel processing algorithms capable of performing under very high parallel efficiency are developed on line-connected and mesh-connected processor arrays. By these proposals, the advantage of two-dimensional non-causal zero-phase recursive digital filters is made clear.