Theoretical prediction of propagation is required for the future urban mobile communications, in order to make possible precise and universal prediction for arbitrary conditions. The necessity and the fundamental concept of theoretical prediction are introduced, and the theoretical prediction of mean field strength in urban areas is reviewed and discussed. Theoretical method is important particularly in prediction of multipath delay characteristics, in relation to the prediction of error rates in digital mobile radio communications.

An optimal algorithm for decomposing a special type of the Hasse diagram into a minimum set of disjoint paths is described. It is useful for testing the consistency of functional dependencies.

This paper discusses three approaches for combining an efficient LR parser and phoneme-context-dependent HMMs and compares them through continuous speech recognition experiments. In continuous speech recognition, phoneme-context-dependent allophonic models are considered very helpful for enhancing the recognition accuracy. They precisely represent allophonic variations caused by the difference in phoneme-contexts. With grammatical constraints based on a context free grammar (CFG), a generalized LR parser is one of the most efficient parsing algorithms for speech recognition. Therefore, the combination of allophonic models and a generalized LR parser is a powerful scheme enabling accurate and efficient speech recognition. In this paper, three phoneme-context-dependent LR parsing algorithms are proposed, which make it possible to drive allophonic HMMs. The algorithms are outlined as follows: (1) Algorithm for predicting the phonemic context dynamically in the LR parser using a phoneme-context-independent LR table. (2) Algorithm for converting an LR table into a phoneme-context-dependent LR table. (3) Algorithm for converting a CFG into a phoneme-context-dependent CFG. This paper also includes discussion of the results of recognition experiments, and a comparison of performance and efficiency of these three algorithms.

A unification-based Japanese parser has been implemented for an experimental Japanese-to-English spoken language translation system (SL-TRANS). The parser consists of a unification-based spoken-style Japanese grammar and an active chart parser. The grammar handles the syntactic, semantic, and pragmatic constraints in an integrated fashion using HPSG-based framework in order to cope with speech recognition errors. The parser takes multiple sentential candidates from the HMM-LR speech recognizer, and produces a semantic representation associated with the best scoring parse based on acoustic and linguistic plausibility. The unification-based parser has been tested using 12 dialogues in the conference registration domain, which include 261 sentences uttered by one male speaker. The sentence recognition accuracy of the underlying speech recognizer is 73.6% for the top candidate, and 83.5% for the top three candidates, where the test-set perplexity of the CFG grammar is 65. By ruling out erroneous speech recognition results using various linguistic constraints, the parser improves the sentence recognition accuracy up to 81.6% for the top candidate, and 85.8% for the top three candidates. From the experiment result, we found that the combination of syntactic restriction, selectional restriction and coordinate structure restriction can provide a sufficient restriction to rule out the recognition errors between case-marking particles with the same vowel, which are the type of errors most likely to occur. However, we also found that it is necessary to use pragmatic information, such as topic, presupposition, and discourse structure, to rule out the recognition errors involved with topicalizing particles and sentence final particles.

In this paper, we propose an extended LR parsing algorithm, called LR parsing with a category reachability test (the LR-CRT algorithm). The LR-CRT algorithm enables a parser to efficiently recognize those sentences that belong to a specified grammatical category. The key point of the algorithm is to use an augmented LR parsing table in which each action entry contains a set of reachable categories. When executing a shift or reduce action, the parser checks whether the action can reach a given category using the augmented table. We apply the LR-CRT algorithm to improve a speech recognition system based on two-level LR parsing. This system uses two kinds of grammars, inter- and intra-phrase grammars, to recognize Japanese sentential speech. Two-level LR parsing guides the search of speech recognition through two-level symbol prediction, phrase category prediction and phone prediction, based on these grammars. The LR-CRT algorithm makes possible the efficient phone prediction based on the phrase category prediction. The system was evaluated using sentential speech data uttered phrase by phrase, and attained a word accuracy of 97.5% and a sentence accuracy of 91.2%

A precise method for determining AlGaAs/GaAs HBT large-signal circuit parameters is presented. In this method, the parameters are extracted from noise parameters and small-signal S-parameters measured under various bias conditions. The measured noise parameters are fitted to the calculated noise parameters derived from an approximation of Hawkins' equations applied to the macroscopic equivalent circuit. The small-signal S-parameters help to determine the large-signal circuit parameters. The derived large-signal parameters were used to design an HBT oscillator. The simulated results using these parameters were in good agreement with the fabricated device performance.

In the paper entitled "The effect of varying routing probability in two parallel queues with dynamic routing under a threshold-type scheduling", Kojima et al. derive an expression in the form of a product of powers for the state probabilities of a threshold-type shortest queue problem. In this note it is demonstrated that this expression is essentially more complicated and has the form of an infinite sum of products of powers. In fact, Kojima et al. find the first term in this infinite sum only.

This paper discusses the problems facing spoken dialogue processing and the prospects for future improvements. Research on elemental topics like speech recognition, speech synthesis and language understanding has led to improvements in the accuracy and sophistication of each area of study. First, in order to handle a spoken dialogue, we show the necessity for information exchanges between each area of processing as seen through the analysis of spoken dialogue characteristics. Second, we discuss how to integrate those processes and show that the memory-basad approach to spontaneous speech interpretation offers a solution to the problem of process integration. The key to this is setting up a mental state affected by both speech and linguistic information. Finally, we discuss how those mental states are structured and a method for constructing them.

This paper describes the higher-order moment analysis of superposed Markov jumping processes. A superposed Markov jumping process is defined as a linear superposition of a finite number of piecewise constant real valued stochastic process whose value changes are associated with state transitions in an underlying descrete state continuous time Markov process. Some phenomena are modeled well by the process such as membrane current fluctuations observed at bio-membranes or load fluctuations in electrical power systems. Theoretical formula of the moment function of any order k is derived and the parameter estimation problem utilizing higher-order moment functions is discussed. A new method of estimating the kinetic parameters of membrane current fluctuations is proposed as a possible application.

With the use of a two-dimensional model, single laser beam trapping of particles is analyzed as the electromagnetic boundary-value problem. From the numerical results, it is found that the trapping mechanism for this system depends upon the surface field distribution of the object.

Separability is a valuable property of nonlinear mappings. By exploiting this property, computational complexity of many numerical algorithms can be substantially reduced. In this letter, a new algorithm is presented that detects the separability of nonlinear mappings using the concept of "computational graph". A hybrid algorithm using both the top-down search and the bottom-up search is proposed. It is shown that this hybrid algorithm is advantageous in detecting the separability of nonlinear simultaneous functions.

The embedded Markov processes associated with Markovian queueing systems are closely related, and their relationships are important for establishing an analytical basis for performance evaluation techniques. As a first step, we analyze the embedded processes associated with a general M/M/1 queueing system. Linear transformations between the infinitesimal generators and the transition probability matrices of embedded processes at arrival and departure times are explicitly derived. Based upon these linear transformations, the equilibrium distributions of the system states at arrival and departure times are obtained and expressed in terms of the equilibrium distribution at arbitrary times. The approach presented here uncovers an underlying algebraic structure of M/M/1 queueing systems, and establishes an algebraic methodology for analyzing the equilibrium probabilities of the system states at arrival and departure times for more general Markovian queueing systems.

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.

Unidirectional/Asymmetric error control codes have extensively been studied, not only from theoretical interest but from application to computer systems or communication systems. Recently, attention has been focused on detecting only d, not all, unidirectional errors, that is, d bits unidirectional error ditecting (d-UED) codes. Borden proposed an optimal nonsystematic d-UED code. This paper shows a new design method for cost-effective self-testing checker for the optimal d-UED code. The checking policy is to check whether condition of the Borden code satisfies or not. The proposed checker includes the parallel weight counter, the comparator and th e modulo adder in which new residue operation is defined and hence this makes the circuit self-testing. These circuits are designed to have all possible input patterns in order to satisfy self-testing property. Finally, the proposed checker has greatly reduced hardware amount compared to the existing one.

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.

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.

This paper presents an efficient heuristic algorithm for min-cut bisection of weighted hypergraphs. The proposed algorithm is based on a heuristic algorithm proposed by Kahng, which was devised for non-weighted hypergraph bisection, adopting a non-weighted graph called intersection graph to represent a given hypergraph. In the proposed algorithm, instead of an intersection graph, a bipartite graph called netgraph is newly introduced to explicitly represent the weights of nodes of a hypergraph. Using the netgraph, it is easy to partition a weighted hypergraph into two hypergraphs with same size. Computation time of the proposed method is O(m2), where m is the number of nodes of a given hypergraph. Experimental results with real circuit data show that the proposed method produces better solutions in shorter computation time compared with existing methods.

This paper surveys modeling techniques for telephone call control based on a Finite State Machine (FSM) concept, and studies model simplification techniques. First, the basic concept and fundamental issues of call control modeling are described. Then, based on the analysis of layered call control configuration, it is clarified that the call control machine decomposition within the two-party service control layer has the effect of reducing the apparent size of each mate's machine. Using this effect, guidelines for call control modeling are derived, by which multiple services can be modeled independently. Finally implementation techniques and a few examples of application will be presented.

The paper describes a parallel algorithm for the manipulation of binary decision diagrams on a shared memory multi-processor system. Binary decision diagrams are very efficient representations of logic functions, and are widely used in computer aided design of logic circuits. Logic operations on logic functions such as AND and OR are reduced to operations on binary decision diagrams representing these functions. Operations on binary decision diagrams are time-consuming in some cases, and a fast manipulation method is needed. As with the manipulation, we focus on the construction of a binary decision diagram from a logic formula, and devised a parallel algorithm for the construction. In the construction, there are many logic operations to be processed, and some of them can be processed in parallel. At first, we introduce an extraction method and a parallel-execution method for such parallelizable operations. This is the parallel execution method for an operation sequence (or a set of operations). To extract more parallelism, we introduce a dynamic expansion method of a logic operation. The dynamic expansion is a method to obtain sub-operations from a logic operation using the modified Shannon's expansion. These sub-operations are executed in parallel and the results of these sub-operations are merged to obtain the result of the original operation. Our parallel algorithm, which is based on the construction of shared binary decision diagrams with the negative edge and the operation cache, is implemented in C on a shared memory multi-processor system Sequent S-81 (CPU 80386 (16 MHz)28, 86.75MB), and applied to multiplier examples and ISCAS benchmarks. The speed-up ratio becomes 14 for multipliers, and becomes 11 for c1908 in ISCAS benchmarks.

GaAs MESFETs with a p-buffer layer (or a buried p-layer) are important devices for high-speed GaAs ICs. To study what conditions are required as a good substrate for ICs, we have investigated, by two-dimensional simulation, small-signal parameters and drain-current transients of GaAs MESFETs with a p-buffer layer on the semi-insulating substrate. It is shown that the introduction of a p-buffer layer is effective to improve the transconductance and the cuttoff frequeycy. These parameters are not degrade even if the p-layer doping is increased and a neurtral p-region exists. It is also shown that drain-current drifts and hysteresis in I-V curves can occur in a case with a p-buffer layer, too. It is concluded that the introduction of a relatively highly-doped p-layer on a substrate with low acceptor and electron trap (EL2) densities is effective to realize the stable and high performance of GaAs MESFETs.