Automatic repeat request (ARQ) for wireless ATM (WATM) operating at 20 Mbit/s or higher is required to achieve high throughput performance as well as high transmission quality, i.e., low CLR (cell loss ratio). Selective Repeat (SR) and Go-Back-N (GBN) are typical ARQ schemes. Though SR-ARQ is superior to GBN-ARQ in throughput performance, the implementation complexity of SR-ARQ's control procedures is disadvantageous to its application to high-speed wireless systems. In addition, when PDU (protocol data unit) length on wireless link is short, the capacity for ARQ control messages can be significantly large. GBN-ARQ, on the other hand, cannot avoid serious throughput degradation due to fairly high BER caused by multipath fading and shadowing, though its implementation is simple. To solve the above-mentioned problems, this paper proposes a novel ARQ scheme named PRIME-ARQ (Partial selective Repeat superIMposEd on gbn ARQ). PRIME-ARQ achieves high throughput performance, almost equal to selective repeat ARQ, with a simple algorithm resulting in reduced implementation complexity for high speed operation. This paper describes the design, implementation, and performance of the proposed PRIME-ARQ. In addition, it shows the experimental results using an experimental PRIME-ARQ hardware processor and proto-type AWA equipment.

It is well known that based on the structure of a transversal filter, the RLS equaliser provides the fastest convergence in stationary environments. This paper addresses an adaptive transversal equaliser which has the potential to provide more faster convergence than the RLS equaliser. A comparison is made with respect to computational complexity required for each update of equaliser coefficients, and computer simulations are demonstrated to show the superiority of the proposed equaliser.

This paper proposes a method of supporting behavioral decision-making with a rule-based system to be used for the design stage of service specifications or service scenarios based on a state transition model. We consider telecommunication services as service scenarios of state transitions. A state transition can be described as a transition rule that is represented as the notation of an IF-THEN rule : "if < condition > then < action >. " Thus, behaviors or service scenarios are assumed to be treated as a set of IF-THEN rules in this paper. In general, an "if < condition > then < action >" rule is called a production rule, and production rules are often used to represent knowledge in expert systems. Thus, rules treated in this paper are a kind of production rule. It is still difficult to decide if behaviors are concurrent, cooperative, or exclusive as a whole system when many service scenarios are combined, even if each rule is clearly described and easy to understand. When several service scenarios are combined, it is necessary for the sub-elements of each state transition to cooperate. The method proposed in this paper consists of the following techniques: (1) decompositions of a state transition to sub-elements, (2) rule descriptions for the decomposed sub-elements and the application of rules, (3) use of conflict and cooperation strategies, and (4) support for behavioral decision-making by showing recommendations for cooperative, exclusive, or concurrent behavior. We provide examples of telecommunication services to show the effectiveness of the proposed method in an easy-to-understand manner. We also show cooperative resolution in the interactions among several services.

It is always difficult to monitor stringent cell loss ratio (CLR), e.g. 10-9, in asynchronous transfer mode (ATM) networks, because its measurement period is too long for real-time measurement. In this paper, we propose new performance monitoring mechanisms for stringent CLR. We consider virtual ATM switches whose resources are much smaller than the real system and thus much higher CLRs will be obtained within a relatively short measurement period. By applying some regression methods in the CLRs obtained from the virtual system, we can estimate the actual CLR of the real system quite accurately and our performance monitoring mechanisms will be operated based on the estimation. Through the numerical examples, our mechanisms are not only more accurate than the traditional methods, but also have shorter measurement periods compared with direct measurement.

In this paper, we investigate the performance of Fast Reservation Protocols (FRP) for burst-level bandwidth allocation in ATM networks. FRP schemes can be classified into delayed transmission (DT) and immediate transmission (IT) methods according to reservation procedure. Moreover, according to the responsibility for negative acknowledgment (NAK) cell generations when burst blocking occurs, FRP schemes can be further classified into blocking node NAK (BNAK) and destination node NAK (DNAK) schemes. We analize the FRP schemes with different reservation and NAK methods for single node and multihop network models, respectively. We then discuss the dependence of performance for each FRP scheme on propagation delay, peak rate, and the number of hops.

This paper proposes three MAC protocols over APON to provide residential and small business customers with multimedia services. The proposed protocols support the frame structure of ITU-T recommendation G.983.1 and also provide diverse ATM service classes such as CBR, rtVBR, nrtVBR, ABR, and UBR traffics. Each service is allocated on the basis of priority. Especially, for allocating CBR and rtVBR services, each protocol uses different cell arrival timing information which is achieved with specific coding and ranging procedure. Focusing the difference of cell arrival timing information, we will investigate the performance of proposed protocols. For the proposed MAC protocols, we present grant field format, minislot format, and bandwidth allocation algorithm. Computer simulation shows the performance of the proposed protocols in terms of CDV and delay, comparing with the normal FIFO protocol.

A speaker adaptation technique that maximizes the observation probability of an input speech is proposed. It is applied to semi-continuous hidden Markov model (SCHMM) speech recognizers. The proposed algorithm adapts the mean µ and the covariance Σ iteratively by the gradient search technique so that the features of the adaptation speech data could achieve maximum observation probabilities. The mixture coefficients and the state transition probabilities are adapted by the model interpolation scheme. The main advantage of this scheme is that the means and the variances, which are common to all states in SCHMM, are adapted independently from the other parameters of SCHMM. It allows fast and precise adaptation especially when there is a large acoustic mismatch between the reference model and a new speaker. Also, it is possible that this scheme could be adopted to other areas which use codebook. The proposed adaptation algorithm was evaluated by a male speaker-dependent, a female speaker-dependent, and a speaker-independent recognizers. The experimental results on the isolated word recognition showed that the proposed adaptation algorithm achieved 46.03% average enhancement in the male speaker-dependent recognizer, 52.18% in the female speaker-dependent recognizer, and 9.84% in the speaker-independent recognizer.

Terahertz radiation imaging has been employed to diagnose the supercurent and vortex distribution in high-Tc superconductive thin film strips. We observe them in the YBCO films patterned into the strip with and without ordered arrays of small antidots. Comparison with the theoretically expected distribution reveals that the present technique can provide a powerful tool for the noncontact, nondestructive, and free-space evaluation of the supercurrent and the vortex distribution with good quantitative agreement. The effect of the antidot formation in the strips is explained by the decrease of the effective critical current. The remanent state after removal of the relatively large magnetic field cannot be explained by the conventional model for the superconductive thin films, and the discrepancy is more notable in the antidot-formed area.

In recent years, more emphasis is placed on the performance of massive databases. It is often required not only that database systems provide high throughputs with rapid response times, but also that they are fully available 24-hours-per-day and 7-days-per-week. Requirements for throughput and response time can be satisfied by upgrading the hardware. As a result, databases in the old hardware environment have to be moved to the new one. Moving a database, however, generally requires taking the database off line for a long time, which is unacceptable for numerous applications. In this paper, a very practical and important subject is addressed: how to upgrade the hardware on line, i.e., how to move a database from an old hardware environment to a new one concurrently with users' reading and writing of the database. A technique for this purpose is proposed. We have implemented a prototype based on this technique. Our experiments with the prototype shown that compared with conventional off-line approach, the proposed technique could give a performance improvement by more than 85% in the query-bound environment and 40% in the update-bound environment.

In this paper we propose a subsequence matching algorithm that supports moving average transform of arbitrary order in time-series databases. Moving average transform reduces the effect of noise and has been used in many areas such as econometrics since it is useful in finding the overall trends. The proposed algorithm extends the existing subsequence matching algorithm proposed by Faloutsos et al. (SUB94 in short). If we applied the algorithm without any extension, we would have to generate an index for each moving average order and would have serious storage and CPU time overhead. In this paper we tackle the problem using the notion of index interpolation. Index interpolation is defined as a searching method that uses one or more indexes generated for a few selected cases and performs searching for all the cases satisfying some criteria. The proposed algorithm, which is based on index interpolation, can use only one index for a pre-selected moving average order k and performs subsequence matching for arbitrary order m ( k). We prove that the proposed algorithm causes no false dismissal. The proposed algorithm can also use more than one index to improve search performance. The algorithm works better with smaller selectivities. For selectivities less than 10-2, the degradation of search performance compared with the fully-indexed case--which is equivalent to SUB94--is no more than 33.0% when one index is used, and 17.2% when two indexes are used. Since the queries with smaller selectivities are much more frequent in general database applications, the proposed algorithm is suitable for practical situations.

This paper is, in half part, written in review nature, and presents recent theoretical results on linear-filtering and -prediction problems of nonstationary Gaussian processes. First, the basic concepts, signal and noise, are mathematically characterized, and information sources are defined by linear stochastic differential equations. Then, it is shown that the solution to a conventional problem of filtering or prediction of a nonstationary time series is, in principle, reducible to a problem, of which solution is given by Kalman-Bucy's theory, if one can solve a problem of finding the canonical representation of a Gaussian process such that it has the same covariance functions as those of the time series under consideration. However, the problem mentioned above is left open. Further, the problem of time-frequency analysis is discussed, and physical realizability of the evolutionary, i.e., the online, spectral analyzer is shown. Methods for dealing with differential operators are presented and their basic properties are clarified. Finally, some of related open problems are proposed.

This paper proposes a group key distribution scheme with a user exclusion. The user exclusion is how to distribute an encryption key over a broadcast channel shared by n users so that all but d excluded users can get the group key. In the broadcast channel such as Pay-TV, Internet multicast and a mobile telecommunication for a group, a manager should exclude a dishonest user or an unauthorized terminal as soon as possible to protect the secrecy of the group communication. However, it takes a long time for the user exclusion on a large group, if the distributor distributes the group key to each user except the excluded one. We propose a scheme in which the amount of transmission and the key storage of each user do not depend on the number of users of the group. Moreover, our scheme does not require a fixed and privileged distributor.

This paper deals with deadlock and fairness issues that may arise when network users request resources for guaranteed service with the resource reservation protocol (RSVP). A deadlock occurs when a request can only be satisfied if the resources reserved for another request are released, but the reserved resources are never released. The fairness issue occurs when some reservation requests may be satisfied but only after a very long wait. Our approach to these issues is based on our belief that a network should provide stable throughput and fairness whatever the behavior of the user. Our methods are unique in two respects. First, during the session setup phase, a node directly connected to the requesting users terminates the users' behavior and makes reservations fairly and efficiently in place of the users. Second, our three admission control methods allocate resources for each reservation request by considering not only the current residual bandwidth but also the properties of the requesting session; e.g., its weight (the number of resources it requires) or its age (how long it has been waiting for session setup). Our methods do not maximize the throughput since they always keep a certain amount of resources unreserved for fairness. From simulation results, however, they do provide quite fair behavior, and their throughput is stable regardless of the network size and the session holding time.

This letter deals with multiuser detection under imprecise knowledge of the received signature code of desired user. In order to improve the revision capability for spread-signature mismatch, an enhanced subspace-based (SSB) technique is developed. By incorporating the orthogonal property of spreading codes into the SSB algorithm, the proposed technique not only can correct the spread-signature mismatch, but also can decorrelate the spreading codes in asynchronous DS-CDMA system. Significant performance improvement of the enhanced SSB technique is demonstrated by simulation results.

Nonlinear characteristics of a DBR (Distributed Bragg Reflector) Cherenkov laser are investigated with the aid of particle simulation, allowing for the nonlinear properties of the electron beam. Numerical results show that the EM power extracted from the cavity is considerably suppressed by the nonlinear effect of the electron beam. Additionally, the extracted EM power is found to be critically dependent on the reflection coefficient of the DBR at the output end. Thus the DBRs at both ends of the cavity should be carefully designed in order to extract the EM power from the cavity efficiently.

This paper presents a class of binary block codes capable of correcting single synchronization errors and single reversal errors with fewer check bits than the existing codes by 3 bits. This also shows a decoding circuit and analyzes its complexity.

Debye's asymptotic series is frequently used for calculation of cylindrical functions. However, it seems that until now this series has not been used in all-purpose programs for numerical calculation of the cylindrical functions. The authors attempt to develop these all-purpose programs. We present some improvements for the numerical calculation. As the results, Debye's series can be used for the all-purpose programs, and it is found out that the series gives sufficient accuracy if some conditions are satisfied.

When a single-mode LD is subjected to distant reflection, relative intensity noise and the width of the optical spectrum are drastically increased. This phenomenon is known as 'coherence collapse. ' This letter demonstrates that penalty-free operation is possible at 2.5 Gbit/s even when a DFB-LD is in a state of coherence collapse. In addition, an LD in a state of coherence collapse is applied to a situation where signal light suffers from interferometric crosstalk. The results show that the LD reduces the influence of interferometric noise because of its wide spectral width.

Combinational logic circuits are usually implemented as multi-level networks of logic nodes. Multi-level logic simplification using the don't cares on each node is widely used. Large don't cares give good simplification results, but suffer from huge memory area and computation time. Extraction of useful don't cares and reduction of the size of the don't cares are important problems on the simplification using don't cares. In the paper, we propose a new robust heuristic method for the selection of don't cares. We consider an adaptive subnetwork for each simplified node in the network and introduce a stepwise enhancement method of the subnetwork considering the memory area and the network structure. The don't cares extracted from the adaptive subnetworks are called the local don't cares. We have implemented our method for satisfiability don't cares and observability don't cares. We have applied the method on MCNC89 benchmarks, and compared the experimental results with those of the SIS system. The results demonstrate the superiority of our method on the quality of the results and on the size of applicable circuits.

A new timing estimation algorithm for asynchronous DS/CDMA multiuser communication system is proposed in this paper. The algorithm is based on the Minimum Variance Distortionless Response (MVDR) beamforming technique that minimizes the beamformer's output power with the constraint that only the signal with exact timing is distortionlessly passed. Exploiting the characteristics that the MVDR beamformer's output power is severely degraded according to erroneous timing estimation, we develop an efficient algorithm to estimate each user's timing by scanning the beamformer's output power variation. Compared to the maximum a posteriori (MAP) or maximum likelihood (ML) based multiuser timing estimator, the complexity is extensively reduced by separating the multi-dimensional optimization problem into several one-dimensional optimization problems. Furthermore, the algorithm is computationally feasible than the subspace-based timing estimator since no eigendecomposition (EVD) is required. Moreover, the proposed algorithm is near-far resistant since the MVDR beamformer is inherently energy independent to the interferers.