Experiment is reported of enhanced backscattering of light in binary and ternary suspensions of rutile and/or alumina particles. With a conventional CCD camera system for observing the phenomena, the angular line shape and the enhancement factor were agreed with the theoretically predicted curve and value. Observation of the angular distribution scattered at the backscattered direction supported the hypothesis proposed by Pine et al. , in which the transport mean free path of the polydisperse mixture can be expressed in terms of summing its reciprocal values weighted over the particle sizes.

Feature interaction detection determines whether interactions occur or not between the new and existing telecommunication services. Most of conventional detection methods on state transition model utilize an exhaustive search. The exhaustive search is fundamentally very powerful in the sense that all interactions are exactly detected. However, it may suffer from the state explosion problem due to the exponential growth of the number of states in the model when the number of users and the number of features increase. In order to cope with this problem, we propose a new detection method using a state reduction technique. By means of a symmetric relation, called permutation symmetry, we succeed in reducing the size of the model while preserving the necessary information for the interaction detection. Experimental evaluation shows that, for practical interaction detection with three users, the proposed method achieves about 80% reduction in space and time, and is more scalable than the conventional ones especially for the increase of the number of users in the service.

Current computing systems are too slow for information processing because of the huge number of procedural steps required. A decrease in the number of calculation steps is essential for real-time information processing. We have developed two kinds of novel architectures for automatic elimination of redundant calculation steps. The first architecture employs the new digit-serial algorithm which eliminates redundant lower digit calculations according to the most-significant-digit-first (MSD-first) digit-serial calculation scheme. Basic components based on this architecture, which employ the redundant number system to limit carry propagation, have been developed. The MSD-first sequential vector quantization processor (VQP) is 3.7 times faster than ordinary digital systems as the result of eliminating redundant lower-bit calculation. The second architecture realizes a decrease in the number of complex calculation steps by excluding useless data before executing the complex calculations according to the characterized value of the data. About 90% of Manhattan-distance (MD) calculations in VQP are excluded by estimating the MD from the average distance.

A typical user is concerned only with the quality of service of a network on an end-to-end basis. Therefore, how end-to-end requirements are mapped into the local switching node requirements and maximum network utilization is a function of network internal design. In this paper, we address the problem of QOS allocation. We derived an optimal QOS allocation policy and decided the maximum utilization bound in a deterministic traffic model. We adopted the worst case delay bound as the end-to-end and local QOS requirement. With (σ, ρ) traffic model, we derived a formula for delay bound and the number of connections. We found that with the delay bound as the QOS metric, there is a significant difference in the performance of allocation policies. We also developed an evaluation strategy to analyze allocation policies. The numerical results for two simple network topologies: tandem network model and uneven traffic load model, compare the equal allocation policy with the optimal allocation policy and show the correctness and efficiency of QOS allocation policy.

The MPEG video coding is the most widely used video coding standard which usually generates variable bitrate (VBR) data streams. Although ATM can deliver VBR traffic, the burst traffic still has the possibility to be dropped due to network congestion. The cell loss can be minimized by using an enforced rate control method. However, the quality of the reproduced video may be sacrificed due to insufficient peak rate available. In this work, we propose an end-to-end quality adaptation mechanism for MPEG traffic over ATM. The adaptive quality control (AQC) scheme allocates a certain number of coding bits to each video frame based on the network condition and the type of next frame. More bits may be allocated if the network condition, represented by the connection-level, is good or the next frame is B-frame that usually consumes fewer bits. A high connection-level allows a relatively large number of tagged cells, which are non-guaranteed in delivery, for video frames with high peak rates. The connection-level adjustment unit at the encoder end adjusts the connection-level based on the message of the network condition from the quality monitoring unit at decoder. The simulation results show that the AQC system can effectively utilize the channel bandwidth as well as maintain satisfactory video quality in various network conditions.

Recently, many on-line control methods of partially observed discrete event systems(DES's) have been proposed. This paper proposes an algorithm for on-line control based on a supervisor under complete observation. It is shown that DES's controlled by the proposed on-line controller generate maximally controllable and observable sublanguages which include the supremal normal sublanguages. Moreover, computational complexity of the proposed algorithm is polynomial with respect to the numbers of the unobservable events and the state of the supervisor under complete observation.

As a means of CDMA network evolution toward future wireless services, a spectral overlay of narrowband CDMA (N-CDMA) and wideband CDMA (W-CDMA) systems is proposed in [8]. In order to justify the overlaying strategy, the reverse link capacity is examined in the same work. Although the capacity of conventional CDMA cellular systems is usually limited by the reverse link, the limit could occur at the forward link depending on the transmission technologies adopted by specific CDMA proposals. Especially, the number of users that can be simultaneously accommodated in the system would be limited by the forward link in future mobile service environments where unequal traffic is offered between two links. In this paper, we first examine the forward link capacity of the spectrally overlaid narrowband and wideband CDMA (N/W CDMA) system. And we compare it with the reverse link capacity to obtain the overall performance. The effects of various parameters on the capacity of N/W CDMA system are numerically evaluated for different mobile environments.

This paper presents a novel dead-beat synchronization scheme and applies it to communications in discrete-time chaotic systems. A well-known Henon system is considered as an illustrative example. In addition, a Henon-based image processing application effectively exploits the proposed scheme's effectiveness.

We proposed a new model for non-stationary time series analysis based on the IAR (inhomogeneous autoregressive) model, and a method for model parameter estimation when the set of basis is given. In this paper, we further propose a method for parameter estimation including that of basis set: we set a new condition that power of the input sequence is concentrated in low-frequency domain, and developed an iterative estimation method. We firstly select an initial set of basis, from which new sets are created in order to minimize the difference between the model and data. Among new sets of basis, we select a good one that gives minimum standard deviation of estimated frequencies.

A nonlinear time series predictor was proposed, in which a nonlinear sub-predictor (NSP) and a linear sub-predictor (LSP) are combined in a cascade form. This model is called "hybrid predictor" here. The nonlinearity analysis method of the input time series was also proposed to estimate the network size. We have considered the nonlinear prediction problem as a pattern mapping one. A multi-layer neural network, which consists of sigmoidal hidden neurons and a single linear output neuron, has been employed as a nonlinear sub-predictor. Since the NSP includes nonlinear functions, it can predict the nonlinearity of the input time series. However, the prediction is not complete in some cases. Therefore, the NSP prediction error is further compensated for by employing a linear sub-predictor after the NSP. In this paper, the prediction mechanism and a role of the NSP and the LSP are theoretically and experimentally analyzed. The role of the NSP is to predict the nonlinear and some part of the linear property of the time series. The LSP works to predict the NSP prediction error. Furthermore, predictability of the hybrid predictor for noisy time series is investigated. The sigmoidal functions used in the NSP can suppress the noise effects by using their saturation regions. Computer simulations, using several kinds of nonlinear time series and other conventional predictor models, are demonstrated. The theoretical analysis of the predictor mechanism is confirmed through these simulations. Furthermore, predictability is improved by slightly expanding or shifting the input potential of the hidden neurons toward the saturation regions in the learning process.

In many applications of digital video database systems such as digital library, video data is often compressed with MPEG video algorithms. It will be an important technique to insert the additional information data like indexes and contents effectively into video database which is compressed with MPEG, because we can always deal with the additional information with video data itself easily. We propose a method for inserting optional binary data such as index information of digital library into MPEG-1 and -2 bitstreams. The binary data inserted MPEG video bitstreams using our proposed scheme are also according to the specification of the MPEG video frame structure. The proposed method allows us to extract the inserted binary data perfectly though MPEG-1 and -2 video are lossy algorithms. And the quality of decoded images after extracting added information is almost the same as that of ordinary MPEG bitstreams. Furthermore, traditional standard MPEG-1 and -2 video decoder which can not extract inserted binary data can also decode images from the binary data inserted MPEG video bitstreams without obvious image degradation. There are some different points between the proposed insertion technique of the binary data and the watermarking technique. The technique of watermarking prepares to deal with alter watermarking by others. And the technique of watermarking is required for the identification of the signature and the perfect extraction of the inserted image signature is not required in the lossy MPEG video environment. On the other hand, we have to extract all of the inserted binary information data correctly with the insertion technique of the binary information. Simulations using MPEG video sequences with inserted binary data are presented to quantify some performance factors concerned. We have not heard about inserting data method which purpose is such as index and content information insertion.

The concept of controlled resource sharing and dynamic quality of service (QoS) on the next generation Internet has attracted much attention recently. It is suggested that, by imposing real-time revision of shared resource allocated to individual media streams or data flows according to user/application QoS demand and resource availability, more balanced and efficient multimedia services can be provided. In this paper, we present an Adaptive Control Framework (ACF), which is developed for controlled resource sharing and dynamic QoS in real-time multimedia service. We discuss main elements of ACF including 1) Control schemes applicable in the framework, and 2) Control mechanisms used in ACF. It is clearly shown in this paper that, with control schemes and mechanisms incorporated in ACF and supportive algorithms and protocols for ACF applications on the Internet, more flexible service and better overall performance in terms of packet loss, latency, signal-noise ratio and re-synchronization delay, can be offered.

In this paper, efficient data transmission methods in wire and wireless interworking environments are studied in the case of handover occurring. To provide efficient mobile data service based on TCP, the wire and wireless interworking module with analysis function of TCP frame header protects the duplicated transmission of TCP frames. Also, by using the forwarding mechanism of status information and buffered data in the interworking module, it is possible to recover frame loss occurring in handover quickly. According to simulation results, proposed method has better performance than conventional method in view of throughput and delay. The reason is due to the fact that duplicated transmission is prevented and data recover is performed very quickly because buffered data is forwarded without loss in the case of handover.

This paper proposes a new design method of nonlinear filtering and fixed-point smoothing algorithms in discrete-time stochastic systems. The observed value consists of nonlinearly modulated signal and additive white Gaussian observation noise. The filtering and fixed-point smoothing algorithms are designed based on the same idea as the extended Kalman filter derived based on the recursive least-squares Kalman filter in linear discrete-time stochastic systems. The proposed filter and fixed-point smoother necessitate the information of the autocovariance function of the signal, the variance of the observation noise, the nonlinear observation function and its differentiated one with respect to the signal. The estimation accuracy of the proposed extended filter is compared with the extended maximum a posteriori (MAP) filter theoretically. Also, the current estimators are compared in estimation accuracy with the extended MAP estimators, the extended Kalman estimators and the Kalman neuro computing method numerically.

Since ATM network is a connection-oriented network, the operation for connectionless service is required for data service in it. There are many ways to support connectionless service in ATM network. They are ATM LAN Emulation, Classical IP and ARP over ATM, Indirect approach, Direct approach, and IP switch. It is known that Direct approach is suited for public network. The connectionless server supports connectionless service in Direct approach. There have been presented two kinds of methods, that is, streaming forwarding method and reassembly forwarding method, to forward the frames in the connectionless server. Reassembly forwarding method can work well with AAL5 which has better efficient characteristics than AAL3/4 in terms of easy use and fewer overheads. This paper proposes an algorithm that can decrease the loss of frame by a proposed buffer management working with AAL5. This paper also investigates the structure of the proposed connectionless server and its performance with the one of the conventional connectionless server through simulations. The proposed connectionless server shows a less frame loss and transfer delay than that of the conventional connectionless server.

In this paper, we propose methods for testing the usability of graphical user interface (GUI) applications based on log files of user interactions. Log analysis by existing methods is not efficient because evaluators analyze a single log file or log files of the same user and then manually compare results. The methods proposed here solve this problem; the methods enable evaluators to analyze the log files of multiple users together by detecting interaction patterns that commonly appear in the log files. To achieve the methods, we first clarify usability attributes that can be evaluated by a log-based usability testing method and user interaction patterns that have to be detected for the evaluation. Based on an investigation on the information that can be obtained from the log files, we extract the attributes of clarity, safety, simplicity, and continuity. For the evaluations of clarity and safety, the interaction patterns that have to be detected include those from user errors. We then propose our methods for detecting interaction patterns from the log files of multiple users. Patterns that commonly appear in the log files are detected by utilizing a repeating pattern detection algorithm. By regarding an operation sequence recorded in a log file as a string and concatenating strings, common patterns are able to be detected as repeating patterns in the concatenated string. We next describe the implementation of the methods in a computer tool for log-based usability testing. The tool, GUITESTER, records user-application interactions into log files, generates usability analysis data from the log files by applying the proposed methods, and visualizes the generated usability analysis data. To show the effectiveness of GUITESTER in finding usability problems, we report an example of a usability test. In this test, evaluators could find 14 problems in a tested GUI application. We finally discuss the ability of the proposed methods in terms of its log analysis efficiency, by comparing the analysis/sequence time (AT/ST) ratio of GUITESTER with those of other methods and tools. The ratio of GUITESTER is found to be smaller. This indicates the methods make log analysis more efficient.

Future cellular systems are envisioned to support mixed traffic, and ultimately multimedia services. However, a mixture of voice and data requires novel service mechanisms that can guarantee quality of service. In order to transfer high-speed data, multislot channel allocation is seen as a favoured solution to the present systems with the least compromise to circuit- switched services. This paper evaluates the performance of narrowband voice calls and multislot data packet transmission in such integrated systems by using a matrix-analytic approach. This method achieves quadratic convergence compared to the conventional spectral methods. Mobility is also considered in a prioritized cellular environment where frequent handoff has the potential of degrading data performance. The voice call distribution, data packets throughput, delay and waiting time distribution are derived. Moreover, a new multiple priority-based distributed control algorithm and a voice rate control scheme are enforced to mitigate the queuing congestion of data packets. The numerical results derived from this study show that larger data packets incur longer latency and the use of these flexible schemes can improve the overall performance.

A new type of a linear decorrelating receiver, named Pseudo-Decorrelator, for asynchronous code division multiple access systems over a Rayleigh fading multipath channel is presented in this paper. Starting with the analyis of the multiple access components of the decision statistics, the outputs of a bank of matched filters, the (K 3K) cross-correlation matrix for each bit is obtained. The non-square cross-correlation matrix is then inverted using the concept of Penrose's generalized inverse of a matrix. In this receiver, the detection process can be started before the whole sequence is received at the receiver, and computing the inverse of a (KN KN) cross-correlation matrix, generally required for linear decorrelating receivers, can be avoided because it is enough to compute only the generalized inverse of a (K 3K) cross-correlation matrix for each data bit. Here, K is the number of users and N is the length of input data sequence. Simulation results are also presented for K-user systems over a Rayleigh fading multipath channel.

This paper proposes the UGLR parser as an extension of the GLR parser. A UGLR parser is powerful enough to parse deterministically any phrase structure language if it is in the class of recursive languages and can parse any context free language as fast as the conventional GLR parser. Natural language processing often requires a parser for languages belonging to classes larger than that of context free languages, and the proposed parser is useful for this purpose.

In universal personal telecommunication (UPT) environments, UPT networks retain information related to incall/outcall registration in UPT user service profiles in order to provide incoming UPT calls for UPT users in any location who have registered at a terminal. As UPT networks support incall registration, terminal users can be different from terminal owners, and several UPT users can register for incoming calls on a single terminal. Therefore, appropriate third-party protection procedures are needed to protect the rights of terminal owners. A terminal profile database can be used to store information regarding terminal states and incall UPT users registered on a terminal in order to enable third-party protection procedures. In order to manage information within both the terminal profile and the service profile, we propose a request-based scheme for incall registration/deregistration of UPT users and incall registration resets of terminal owners. We evaluate the performance of the scheme in terms of; 1) total cost and, 2) the number of terminal profile accesses per unit time for a terminal.