Land mobile communications in Japan have shown remarkable progress in recent years. The total number of all types of radio stations has exceeded 750 million as of March, 1992 and more than 80% of them are used for land mobile communications. The more radio telecommunications becomes popular, the more demand for communicating at any time, at any place and with anyone, intensifies. Various new land mobile systems such as digital cellular telephones have been developed and to be introduced. These new systems are designed to promote effective frequency use in order to meet the exploding demand for it. The digitalization of land mobile communication systems will be the key technology which enable to bring the new possibility in the land mobile communications.

This paper describes voice communication connection controls in digital public land mobile networks (D-PLMNs). Voice communications in the D-PLMNs are carried at about 10 kbit/s over narrow-band TDMA channels with highly efficient cellular voice encoding schemes. Extensive research is being carried on half-rate voice encoding schemes that will effectively double radio resources. We first outline the configuration of voice communication connection between a cellular phone in the D-PLMN and a telephone in a fixed network, and we describe the optimum position for the CODECs that transform cellular voice codes to the conventional voice codes used in the fixed network, and vice versa. Then we propose a CODEC-bypassed communication control scheme that improves the quality of voice communication between cellular phones. And we propose a cellular voice code negotiation scheme in the D-PLMN which supports different cellular voice encoding schemes. We also propose an efficient channel reassignment scheme for effectively assigning TDMA channels to voice calls with two different bitrates (full-rate and half-rate), and we analyze this scheme's traffic capability. Finally, we describe a dual-tone multiple-frequency (DTMF) signal transmission scheme and estimate the number of DTMF signal senders required in the D-PLMN.

Stability property of an optically injection-locked semiconductor laser taking account of gain saturation is discussed. Numerical analysis shows that stable locking region is broadened due to gain saturation. This is because of rapid damping of relaxation oscillation due to gain saturation. It is also found that stable locking region is also broadened with increasing injection current since damping of relaxation oscillation becomes strong with increasing injection current. Numerical calculations of lasing spectrum show that the magnitude of sidepeaks appeared at harmonics of relaxation oscillation frequency under unstable locking condition are suppressed due to gain saturation.

This article discusses a synthesis procedure of a discrete-time asynchronous neural network whose information is a limit cycle. The synthesis procedure uses a novel connection matrix and can be reduced into a linear epuation. If all elements of desired limit cycles are independent at each transition step, the equation can be solved and all desired limit cycles can be stored. In some experiments, our procedure exhibits much better storing performance than previous ones.

This paper presents a new architecture of a copy network which employs the principle of recursive generation of copy cells. The proposed architecture achieves high utilization of the links and buffers of the copy network, and preserves the cell sequence. The architecture lends itself modularity so that large multicast ATM switches can be fabricated by employing the proposed copy network. Two different modular structures - one for reduced latency of the unicast cell and the master cell from which copies are made, and the other for reduced hardware overhead - for realizing large multicast ATM switches are configured. The hardware of functional elements of the copy network are the same as those of the functional elements of a modular point-to-point switch proposed earlier, thereby resulting in the modularity of functional elements as well. Simulation studies show that the proposed copy network achieves high throughput and low cell loss probability, and the required buffer sizes are small. The delay of cells is found to be very small for traffic loads up to 90%.

A multi-valued addressing scheme is proposed for a high speed, high packing density memory system. This scheme is a level-multiplex addressing scheme instead of standard time-multiplex addressing scheme, and provides all address signals to the DRAM at the same time without increasing the address pin counts. This scheme makes memory matrix strechable and achieves the low power dissipation using the enhanced partial array activation. The 16 Mb stretchable memory matrix DRAM (16MbSTDRAM) is examined using this addressing design. A power dissipation of 121.5 mW, access time of 30 ns, and 20 pin have been estimated for 3.3 v 16MbSTDRAM with X/Y=15/9 adress configuration. The low power battery-drive memory system for such as the note-book or the handheld-type personal computers can be realized by the STDRAMs with the multi-valued addressing scheme.

Fahlman and Lebiere's (FL) learning algorithm begins with a two-layer network and in course of training, can construct various network architectures. We applied FL algorithm to the same three-layer network architecture as a back propagation (BP) network and compared their generalization properties. Simulation results show that FL algorithm yields excellent saturation of hidden units which can not be achieved by BP algorithm and furthermore, has more desirable generalization ability than that of BP algorithm.

A new design method is proposed for realizing a hypercube network (HC) structured multicomputer system on a wafer using wafer-scale integration (WSI). The probability that an HC can be constructed on a wafer is higher in this method than in the conventional method; this probavility is called a construction probability. We adopt the FUSS method for the processor (PE) address allocation in our desing because it has a high success probability in the allocation. Even if the design renders the address allocation success probalility hegher, it is of no use if it makes either the maximum wiring length between PEs or the array size (wiring area) larger. A new wiring channel structure capable of connecting PEs on a wafer is proposed in this paper, where a channel, called a basic channel, is used. A one-dimensional-array sub-HC row network (RN) or column networks (CN) can be constructed using the basic channel. The sub-HC construction method, which embeds wirings into the basic channel, is also proposed. It requires almost the same wiring width as conventional method. However, it has an advantage in that maximum wiring length between PEs can be about half that of the conventional method. If PEs must be shifted in the case of PE defects, they can be shifted and connected to the basic channel using other PE shifting channels, and an RN or CN can be constructed. The maximum wiring length between PEs, array size, and construction probability will also be derived, and it will be shown that the proposed design is superior to the conventional one.

A newly developed planar inductor and its application to dc-dc converters are described. The planar inductor consists of a planar spiral coil and soft magnetic sheets, it has a small size (11110.8mm), 33µH inductance and a maximum quality factor of 14. The step down chopper dc-dc converter has been developed by using planar inductor, which has small size (20154mm), 5V-2W typical output and output power/volume ratio of 1.7W/cc. The switching converter can be miniaturized by using the planar inductor.

There have been few studies on formal approaches to the specification and realization of asynchronous sequential circuits. For synchronous sequential circuits, an algebraic method is proposed as one of such approaches, but it cannot be applied to asynchronous ones directly. This paper describes an algebraic method of specifying the abstract behavior of asynchronous sequential circuits. We select an daisy chain arbiter as an example of them. In the arbiter, state transitions are caused by input changes, and all the modules do not always make state transitions simultaneously. These are main obstacles to specify it in the same way as sychronous sequential circuits. In order to remove them, we modify the meaning of input in specifications and introduce pseudo state transitions so that we can regard all the modules as if they make state transitions simultaneously. This method can be applied to most of the other asynchronous sequential circuits.

The process of room acoustic planning & design can be aided by Binaural Technology. To this end, a three-stage modelling process is proposed that consists of a "sound"-specification phase, a design phase and a work-plan phase. Binaural recording, reproduction and room simulation techniques are used throughout the three phases allowing for subjective/objective specification and surveillance of the design goals. The binaural room simulation techniques involved include physical scale models and computer models of different complexity. Some basics of binaural computer modelling of room acoustics are described and an implementation example is given. Further the general structure of a software system that tries to model important features of the psychophysics of binaural interaction is reported. The modules of the model are: outer-ear simulation, middle-ear simulation, inner-ear simulation, binaural processors, and the final evaluation stage. Using this model various phenomena of sound localization and spatial hearing, such as lateralization, multiple-image phenomena, summing localization, the precedence effect, and auditory spaciousness, can be simulated. Finally, an interesting application of Binaural Technology is presented, namely, a so called Cocktail-Party-Processor. This processor uses the predescribed binaural model to estimate signal parameters of a desired signal which may be distored by any type of interfering signals. In using this strategy, the system is able to even separate the signals of competitive speakers.

The pressure waveforms indicated on a catheter manometer system are subject to serious distortion due to the resonance of the catheter itself, or the compliance of a particular transducer. Although several methods have been proposed for improving those characteristics, they ahave never been put into practice. We have focused on the transfer function of the catheter manometer, and made a pilot system, using the natural observation method. This method has been suggested as a means of studying the structure of the instantaneous waveform. In this manner, we were able to increace the bandwidth in the ferquency domain and reduce the ringing in the time domain. Correction was performed automatically, using a step wave. Reproduction of the waveform with a flushing device, was a task of equal simplicity, that allowed us to estimate the system parameters so that the response waveform became step-like. In the experiment, our system provided distortion-free left-ventricular pressure waveform measurements and exact evaluation of the cardiac pumping system. The values obtained came much closer to the original figures arrived at by the catheter-tip manometer system.

This paper describes a new trial of dynamical parameter estimation for the actual room acoustic system, in a practical case when the input excitation is polluted by a background noise in contrast with the usual case when the output observation is polluted. The room acoustic system is first formulated as a discrete time model, by taking into consideration the original standpoint defining the system parameter and the existence of the background noise polluting the input excitation. Then, the recurrence estimation algorithm on a reverberation time of room is dynamically derived from Bayesian viewpoint (based on the statistical information of background noise and instantaneously observed data), which is applicable to the actual situation with the non-Gaussian type sound fluctuation, the non-linear observation, and the input background noise. Finally, the theoretical result is experimentally confirmed by applying it to the actual estimation problem of a reverberation time.

In this paper we present and analyze multi-level quorum schemes for maintaining the consistency of replicated data in the presence of concurrency and failures in a large distributed environment. The multi-level quorum method operates on a logical hierarchy of the nodes in the network and applies well known flat voting algorithms for replicated data concurrency control in a layered fashion. We show how the number of hierarchy levels, the number of logical entities per level and the voting algorithms used on each level affect the costs and the degree of availability associated with a wide range of multi-level quorum schemes. The results of the analysis are used to provide guidelines for designing the most suitable multi-level quorum strategy for a given application scenario. Comparative performance measurements in a simulated network are presented to illustrate the properties of multi-level approaches when some of the assumptions of the analytical investigation do not hold.

This paper proposes a new adaptive algorithm of the FIR type digital filter for an acoustic echo canceller and similar application fields. Unlike an echo canceller for line, an acoustic echo canceller requires a large number of taps, and it must work appropriately while it is driven by colored input signal. By controlling the filter tap length and updating filter coefficients multiple times during a single sampling interval, the proposed algorithm improves the convergence characteristics of adaptation even if colored input signal is introduced. This algorithm is maned VT-LMS after variable tap length LMS. The results of simulation show the effectiveness of the proposed algorithm not only for white noise but also for colored input signal such as speech. The VT-LMS algorithm has better convergence characteristice with very little extra computational load compared to the conventional algorithm.

This paper relates to a novel algorithm for fast estimation of the coefficients of the adaptive FIR filter. The novel algorithm is derived from a first order IIR filter experssion clarifying the estimation process of the NLMS (normalized least mean square) algorithm. The expression shows that the estimation process is equivalent to a procedure extracting the cross-correlation coefficient between the input and the output of an unknown system to be estimated. The interpretation allows to move a subtraction of the echo replica beyond the IIR filter, and the movement gives a construction with the IIR filter coefficient of unity which forms the arithmetic mean. The construction in comparison with the conventional NLMS algorithm, improves the covergence rate extreamly. Moreover, when we use the construction with a simple technique which limits the term of calculating the correlation coefficient in the beginning of a convergence process, the convergence delay becomes negligible. This is a very desirable performance for acoustic echo canceller. In this paper, double-talk and echo path fluctuation are also studied as the first stage for application to acoustic echo canceller. The two subjects can be resolved by introducing two switches and delays into the evaluation process of the correlation coefficient.

This paper describes the derivation of a parallel program from a nondeterministic sequential program using a bottom-up parser as an example. The derivation procedure consists of two stages: exploitation of AND-parallelism and exploitation of OR-parallelism. An interpreter of the sequential parser BUP is first transformed so that processes for the nodes in a parsing tree can run in parallel. Then, the resultant program is transformed so that a nondeterministic search of a parsing tree can be done in parallel. The former stage is performed by hand-simulation, and the latter is accomplished by the compiler of ANDOR-, which is an AND/OR parallel logic programming language. The program finally derived, written in KL1 (Kernel Language of the FGCS Project), achieves an all-solution search without side effects. The program generated corresponds to an interpreter of PAX, a revised parallel version of BUP. This correspondence shows that the derivation method proposed in this paper is effective for creating efficient parallel programs.

MISRs are widely used as signature circuits for VLSI built-in self tests. To improve the aliasing probability of MISRs, multiple MISRs and M-stage MISRs with m inputs are available, where M is grater than m. The aliasing probability as a function of the test length is analyzed for the compaction circuits for a binary symmetric channel. It is observed that the peak aliasing probability of the double MISRs is less than that of M-stage MISRs with m inputs. It is also shown that the final aliasing probability for a multiple MISR with d MISRs is 2dm and that for an M-stage MISR with m imputs is 2M if it is characterized by a primitive polynomial.

Strongly Fault-Secure (SFS) circuits are known to achieve the TSC goal of producing a non-codeword as the first erroneous output due to a fault. Strongly Code-Disjoint (SCD) circuits always map non-codeword inputs to non-codeword outputs even in the presence of faults so long as the faults are undetectable. This paper presents a new generalized design method for the SFS and SCD realization of combinational circuits. The proposed design is simple, and always gives an SFS and SCD combinational circuit which implements any given logic function. The resulting SFS/SCD circuits can be connected in cascade with each other to construct a larger SFS/SCD circuit if each interface is fully exercised.

This paper presents a new method for estimating both the parameters of a nonminimum phase system and its unknown input signal. An approximate inverse system method is used to estimate the unknown input signal, and then, by using a Kalman filter, approximately consistent parameter estimates of the nonminimum phase system can be obtained effectively. This method can be used to estimate the parameters of a nonminimum phase system and a minimum phase one in the case when the input signal is a white noise or an impulse sequence.