This paper describes a super high definition (SHD) image processing system we have developed. The computing engine of this system is a parallel processing system with 128 processing elements called NOVI- HiPIPE. A new pipelined vector processor is introduced as a backend processor of each processing element in order to meet the great computing power required by SHD image processing. This pipelined vector processor can achieve 120 MFLOPS. The 128 pipelined vector processors installed in NOVI- HiPIPE yield a total system peak performance of 15 GFLOPS. The SHD image processing system consists of an SHD image scanner, and SHD image storage node, a full color printer, a film recorder, NOVI- HiPIPE, and a Super Frame Memory. The Super Frame Memory can display a ful color moving image sequence at a rate of 60 fps on a CRT monitor at a resolution of 2048 by 2048 pixels. Workstations, interconnected through an Ethernet, are used to control these units, and SHD image data can be easily transfered among the units. NOVI- HiPIPE has a frame memory which can display SHD still images on a color monitor, therefore, one processed frame can be directly displayed. We are developing SHD image processing algorithms and parallel processing methodologies using this system.

An optical array imaging system is presented with basic experimental results. First, a remote object is illuminated with laser light at an angle and the reflected light is detected with an array sensor after interfering it with the reference light. This process is repeated by changing the illumination angle to collect a set of fringe patterns, which are A/D converted and stored in a harddisk in a computer. Then, the data are processed on a computer, first, to estimate the complex-amplitude object wave fields, second, to derive the eigenvector with the maximum eigenvalue for the correlation of the estimated object fields, and finally, to form an image of the object. The derivation of the eigenvector follows an iterative algorithm, which can be interpreted as the process of repeating backward wave propagation of the field between the two apertures illuminating and detecting laser light. The eigenvector field can be expected to backpropagate to focus at a point on the object with the maximum coefficient of reflection, so that a beam-steering operation is applied to the eigenvector to form an image of the object. The method uses only the information of the array data and the lateral spacings of the receiving array (CCD) elements. Hence, the method can give good images of objects even if the reference light is uncollimated with an unknown distorted wavefront, and even if the illuminating angles are imprecise in three dimensions. Basic experimental results clearly show the usefulness of the method.

Multidimensional signal processing has recently been attracting attention in various fields, and has been studied theoretically. TV receives using 3-D (3-Dimensional: horizontal, vertical and temporal) processing, such as IDTV (ImproveD TV), are already available. In addition, television systems with high quality video and mostly with wide-aspect ratio are being studied worldwide. All the proposed systems adopt 3-D signal processing. 3-D processing can fully utilize the transmitted signal, and can take full advantage of the available bandwidth. This results in improved picture quality. This paper reviews the 3-D signal processing used in IDTV and EDTV (EnhanceD TV) in Japan. Video signals are analyzed in the 3-D frequency domain, and 3-D filter design is also studied.

We have developed an advanced tool for dimensioning circuit-switched networks, called CNEP (Circuit-Switched Network Evaluation Program) , for effective design of digital networks. CNEP features a high-reliability network structure (node dispersion, double homing, etc) , both-way circuit operation, and circuit modularity (or big module size), all of which are critical for digital networks. CNEP also solves other dimensioning problems such as the cost difference between existing and newly installed circuits, and handles multi-hour traffic conditions, dynamic routing, and multiple-switching-unit nodes. Operations Research techniques are applied to produce exact and heuristic algorithms for these problems. Algorithms with good time-performance trade-off characteristics are chosen for CNEP.

Syntax-directed editors have several advantages in editing programs because programming is guided by the syntax and free from syntax errors. Nevertheless, they are less popular than text editiors. One of the reason is that they force a priori specified editing structures on the user and do not allow him to use his own structure. ACE (Algorithmically Customizable syntax-directed Editor) provides a solution for this problem by using a technique of machine learning; ACE has a special function of customizing the grammar algorithmically and interactively based on the learning method for grammars from examples and queries. The grammar used in the editor is customized through interaction with the user so that the user can edit his program in a more familiar structure. The customizing function has been implemented based on the methods for learning of context-free grammars from structural examples, for which the correctness and the efficiency are proved formally. This guarantees the soundness and the efficiency of customization. Furthermore, ACE can be used as an algorithmic and interactive tool to design grammars, which is required for several purposes such as compiler design and pretty-printer design.

This paper proposes an efficient design method of three-dimensional (3-D) recursive digital filters for video signal processing via decomposition of magnitude specifications. A given magnitude specification of a 3-D digital filter is decomposed into specifications of 1-D digital filters with three different (horizontal, vertical, and temporal) directions. This decomposition can reduce design problems of 3-D digital filters to design problems of 1-D digital filters, which can be designed with ease by conventional methods. Consequently, design of 3-D digital filters can be efficiently performed without complicated tests for stability and large amount of computations. In order to process video signal in real time, the 1-D digital filters with temporal direction must be causal, which is not the case in horizontal and vertical directions. Since the proposed method can approximate negative magnitude specifications obtained by the decomposition with causal 1-D R filters, the 1-D digital filters with temporal direction can be causal. Therefore the 3-D digital filters designed by the proposed method is suitable for real time video signal processing. The designed 3-D digital filters have a parallel separable structure having high parallelism, regularity and modularity, and thus is suitable for high-speed VLSI implementation.

Based on the Fornasini-Marchesini second model, an efficient algorithm is developed to derive the characteristic polynomial and the inverse of the system matrix from the state-space parameters. As a result, the external description of the Fornasini-Marchesini second model is clarified. A technique for designing 2-D recursive digital filters in the frequency domain is then presented by using the Fornasini-Marchesini second model. The resulting filter approximates both magnitude and group delay specifications and its stability is always guaranteed. Finally, three design examples are given to illustrate the utility of the proposed technique.

There has been much interest in building corporate private communication networks. This aim requires a method that optimizes the economical aspect of network configurations. The authors propose a network design method supported by knowledge-base. This method is used to design a cost-effective corporate communication network, using leased lines. The knowledge-base description depends on the production rule. The network configuration is improved by introducing expert knowledge, after designing the network by a conventional algorithm. Design results show the advantages of the proposed method.

Extended form of interpolatory approximation is presented for tne n-dimensional (n-D) signals whose generalized spectrums have weighted norms smaller than a given positive number. The presented approximation has the minimum measure of approximation error among all the linear and the nonlinear approximations using the same generalized sample values.

General estimation technique using covariance information is proposed for white Gaussian and white Gaussian plus coloured observation noises in linear stationary stochastic systems. Namely, autocovariance data of signal and coloured noise appear in a semi-degenerate kernel, which represents functional expression of the autocovariance data, in the current technique. Then the signal is estimated by directly using autocovariance data of signal and coloured noise. On the other hand, in the previous technique, the covariance information is expressed in the form of a semi-degenerate kernel, but its elements do not include any autocovariance data.

In this paper, voltage-input current-output Membership Function Circuit (MFC) and Normalization Locked Loop (NLL) are proposed. They are useful building blocks for the current-mode analog fuzzy hardware. The voltage-input current-output MFC consists of one source coupled type Operational Transconductance Amplifier (OTA). The MFC is used in the input parts of the analog fuzzy hardware system. The fuzzy hardware system can execute the singleton fuzzy control algorithm. In the algorithm, the weighted average operation is processed. When the weighted average operation is directly realized by analog circuits, a divider must be implemented. Here, the NLL circuit, which can process the weighted average operation without the divider, is implemented using one source coupled type OTA. The proposed circuits were designed by using 2 µm CMOS design rules and its operations were confirmed using SPICE simulations.

This paper discusses and evaluates, from the viewpoints of definition, analysis, and performance, frequency assignment schemes that enable the efficient assignment of multiple-bandwidth carriers on the transponder in SCPC/FDMA systems with demand assignment operation. The system considered handles carriers of two different bandwidths, and assigns only consecutive slots on the transponder band to broadband carriers. Three types of frequency assignment schemes are proposed, each of which incorporates one or both of two assignment concepts: (1) pre-establishment of assignment priorities on the transponder band, and (2) establishment of broadband slots to guide broadband carrier assignment. Following a definition of the schemes, equations are derived to theoretically analyze performance factors such as call loss for the narrowband and broadband carriers, and system utilization efficiency. Finally, theoretical performance calculated for various traffic and system conditions are presented and evaluated, for the purpose of comparison between the three schemes. Computer simulation results are also presented, to demonstrate the accuracy of the derived equations and to supply data for models too large for theoretical computation. Main results obtained are as follows. (1) Regardless of traffic or system conditions, the assignment scheme incorporating both assignment priorities and broadband slots shows the best performance in terms of broadband call loss and system utilization efficiency. (2) The establishment of broadband slots improves performance when the ratio of broadband traffic to the total traffic volume is high, but worsens performance when the narrowband traffic ratio is higher. (3) All aspects of performance improve with the increase of the total number of assignable slots on the transponder band.

This paper describes the design of a 38-GHz high power MMIC amplifier using an improved load-pull technique. We improved the load-pull technique accuracy by using MMIC transtormers to match the input and output impedances of a GaAs MESFET to about 50 ohms. We used this technique to measure the large-signal load impedance of a FET with a 600-µm-wide gate. Using the data obtained, we developed an MMIC amplifier composed of two of these FET cells. At 38 GHz, the amplifier has an output power of 23.5 dBm for a 1 dB gain compression level.

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.

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.

An investigation into the spatial properties of the motion compensation prediction error signal has been carried out to provide a better understanding of it and to model the spatial power spectrum of the error signal. To construct a theoretical model, the motion compensation prediction process is analyzed, including the interpolation process used for motion compensation with decimal place precision, in the horizontal and vertical directions separately, thereby deriving its statistical power gain function. Properties of the input processing system are also examined. Based on these analyses, this paper proposes a theoretical model of the error signal, clarifies its spatial properties that are distinctive of the interlace scanned picture signal, and collates the obtained data with the real picture, thereby verifying the validity of the model. This model is especially useful for the evaluation, selection and detailed designing of the coding techniques of the error signal.

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.

This paper deals with the problem of assigning tasks to the processors of a multiprocessor system such that the sum of execution and communication costs is minimized. If the number of processors is two, this problem can be solved efficiently using the network flow approach pioneered by Stone. This problem is, however, known to be NP-complete in the general case, and thus intractable for systems with a large number of processors. In this paper, we propose a network flow approach for the task assignment problem in homogeneous hypercube networks, i.e., hypercube networks with functionally identical processors. The task assignment problem for an n-dimensional homogeneous hypercube network of N (=2n) processors and M tasks is first transformed into n two-terminal network flow problems, and then solved in time no worse than O(M3 log N) by applying the Goldberg-Tarjan's maximum flow algorithm on each two-terminal network flow problem.

We propose a new distributed signal (analog or digital) transmission system which has the immunity against the noisy channel. An information signal in transmitter is distributed by distributor and the distributed signal is transmitted. Received signal is reconstructed by the inverse distributor in receiver. In this system, an impulsive interference noise which disturbs the transmission signal in the channel passes decoder only, and this interference noise is distributed by the inverse distributor while the transmitted signal is reconstructed. Some appended signals make it possible to estimate the noise components which inversely distributed with the Fourier transformation as the distributor. Basing upon this principle, the transmission system will have an ability to suppress the impulsive interference, and the channel will have high noise immunity. The construction of receiver which can eliminate the impulsive noise is derived.

In general, dynamic channel assignment has a better performance than fixed channel assignment in a cellular mobile communication system. However, it is complex to control the system and a lot of equipments are required in each cell when dynamic channel assignment is applied to a large service area. Therefore, it is effective to limit the size of the service area in order to correct the defects of dynamic channel assignment. So, we propose an application of dynamic channel assignment to a part of a service area when fixed channel assignment is applied to the remaining part of the area. In the system, the efficiency of channel usage in some cells sometimes becomes terribly low. The system has such a problem to be improved. We show that the rearrangement of the channel allocation is effective on the problem.