Machine learning and data mining algorithms are increasingly being used in the intrusion detection systems (IDS), but their performances are laggard to some extent especially applied in network based intrusion detection: the larger load of network traffic monitoring requires more efficient algorithm in practice. In this paper, we propose and design an anomaly intrusion detection (AID) system based on the vector quantization (VQ) which is widely used for data compression and high-dimension multimedia data index. The design procedure optimizes the performance of intrusion detection by jointly accounting for accurate usage profile modeling by the VQ codebook and fast similarity measures between feature vectors to reduce the computational cost. The former is just the key of getting high detection rate and the later is the footstone of guaranteeing efficiency and real-time style of intrusion detection. Experiment comparisons to other related researches show that the performance of intrusion detection is improved greatly.

An efficient and simple approach to consonant/vowel (C/V) segmentation by incorporating the SNR improvement of a speech enhancement system with the energy variation of two adjacent frames is proposed. Experimental results show that the proposed scheme performs well in segmenting C/V for a spontaneously spoken utterance.

An efficient adaptation technique of the delay is introduced for accomplishing more accurate adaptive linear equalization of nonminimum phase channels. It is focused that the filter structure and adaptation procedure of the adaptive Butler-Cantoni (ABC) equalizer is very suitable to deal with a variable delay for each iteration, compared with a classical adaptive linear transversal equalizer (LTE). We derive a cost function by comparing the system mismatch of an optimum equalizer coefficient vector with an equalizer coefficient vector with several delay settings. The cost function is square of difference of absolute values of the first element and the last element for the equalizer coefficient vector. The delay adaptation method based on the cost function is developed, which is involved with the ABC equalizer. The delay is adapted by checking the first and last elements of the equalizer coefficient vector and this results in an LTE providing a lower mean square error level than the other LTEs with the same order. We confirm the performance of the ABC equalizer with the delay adaptation method through computer simulations.

In this paper, we present the biometric authentication system based on the fusion of two user-friendly biometric modalities: Iris and Face. Using one biometric feature can lead to good results, but there is no reliable way to verify the classification. To achieve robust identification and verification we are combining two different biometric features. We specifically apply 2-D discrete wavelet transform to extract the feature sets of low dimensionality from the iris and face. And then to obtain Reduced Joint Feature Vector (RJFV) from these feature sets, Direct Linear Discriminant Analysis (DLDA) is used in our multimodal system. This system can operate in two modes: to identify a particular person or to verify a person's claimed identity. Our results for both cases show that the proposed method leads to a reliable person authentication system.

Extending the domain of the vector potential in the so-called Hallen's equation, four unknown constants are determined to satisfy the boundary conditions in the same way as the circuit theory, where the vector potential plays the leading role, from which the current density and the current itself are derived. Vanishing of the current density just outside the ends of the antenna is required. For a tube-shaped antenna with walls of infinitesimal thickness, further the current just inside the ends of the antenna should vanish, as a result, the current distribution becomes sinusoidal. Adoption of either the surface current distribution or axial current distribution incurs a crucial effect on the value of the currents calculated from the vector potential. The numerical results of the radiation impedance of a hslf-wave antenna show a tendency of consistency with that relatively newly obtained by employing the exact kernel. The problem on the nonsolvability of Hallen's equation is cleared up. Comments are given on the moment method in relation to the boundary value problems to recommend to add two more undecided constants to Hallen's equation.

An effective access control scheme in CDMA networks is proposed to provide multimedia services. The proposed scheme controls the access of multimedia traffic using the delay of data traffic and the transmission rate control of video traffic. Numerical results show that throughput and delay are improved by allowing delay of data traffic under low and medium traffic conditions, and by decreasing the transmission rate of video traffic under heavy traffic conditions.

Recently, Chien et al. proposed an efficient timestamp-based remote user authentication scheme using smart cards. The main merits include: (1) user-independent server, i.e., there is no password or verification table kept in the server; (2) users can freely choose their passwords; (3) mutual authentication is provided between the user and the server; and (4) lower communication and computation cost. In this paper, we show that Chien et al.'s scheme is insecure against forgery attack because one adversary can easily pretend to be a legal user, pass the server's verification and login to the remote system successfully. An improved scheme is proposed that can overcome the security risk while still preserving all the above advantages.

The system under study is a convolutionally coded and orthogonally modulated DS-CDMA system over time-varying frequency-selective Rayleigh fading channels in multiuser environments. Iterative soft demodulation and decoding using the Turbo principle can be applied to such a system to increase the system capacity and performance. To combat multiple access interference (MAI), we incorporate the interference cancellation (IC) and decision-directed channel estimation (CE) in the demodulator. However, both IC and CE are subject to performance degradation due to incorrect decisions. In order to prevent error propagation from the decision feedback, soft interference cancellation and channel estimation assisted demodulation is proposed in this paper. The performance of this strategy is evaluated numerically and proved to be superior to the hard decision-directed approach with a minor increase in complexity.

This paper investigates a modifying orthogonality factor for synchronous DS-CDMA uplink in dispersive Rician multipath fading channels, which reflects upon the effects of specular path power as well as decaying channel characteristics. Using this investigation, the orthogonal factors in indoor environments are evaluated and compared with the various parameters such as decaying factor, line-of-sight component, and the number of multipaths.

In this letter, we focus on rearranged pilot patterns for channel estimation in a mobile communication system using Orthogonal Frequency Division Multiplexing (OFDM). The conventional pilot patterns for channel estimation in OFDM systems do not have robust characteristics in time-varying channels. In order to overcome this weakness of the conventional pilot patterns, we propose the pilot patterns with robust mobility for OFDM systems, which can achieve a good error performance in time-varying multi-path fading channels. Simulation results show that the bit error rate (BER) performances of the proposed pilot patterns are better than those of the conventional pilot patterns in fast time-varying fading channels under the same pilot density and data rate.

In 2000, Xu and Tilborg proposed an ID-based conference key distribution scheme which builds on earlier work of Harn and Yang in the 2-party setting. Recently, Jung et al. have discovered security flaws in the Xu-Tilborg scheme and proposed an improvement of this scheme to fix the security flaws. However, Jung et al.'s improvement introduces another security weakness. We demonstrate this by showing that the improved scheme is vulnerable to a parallel session attack mounted by two colluding adversaries. Further, we recommend changes to the scheme that address this vulnerability.

Incremental modification and optimization in VLSI physical design is of fundamental importance. Based on the O-tree (ordered tree) representation which has more prominent advantages in comparison with other topological representations of non-slicing floorplans, in this paper, we present an incremental placement algorithm for BBL (Building Block Layout) design in VLSI physical design. The good performance of experimental results in dealing with some instances proves the effectiveness of our algorithm.

In this study, we propose a new acoustic model including the human ear canal and a thin tube earphone. The use of a tube earphone enables simultaneous listening of both virtual and real surrounding sound. First, we perform acoustic FDTD (finite difference time domain) simulations using an MRI head model with ear canals. The calculated external impedance viewed from the eardrum numerically shows that the influence of the inserted tube is small. A listening experiment with six subjects also confirms the effectiveness of a tube earphone. Second, we calculate HRTFs (head-related transfer functions) for eight directions in the horizontal plane to realize sound localization with a tube earphone. We also design inverse filters based on the propagation calculations including the characteristics of tube earphones. Finally we evaluate the localization system by another listening experiment with six subjects. The results reveal that the applicability of a system with tube earphones and inverse filters, particularly for the front directions.

Application-level multicast (ALM) is a feasible alternative to IP multicast. In ALM, multicast related features, such as group membership management, multicast routing and packet replication, are implemented at end-hosts instead of routers. A multicast distribution tree is constructed in the application layer, so all nodes in this tree are end-hosts. Packet transmission between end-hosts uses conventional IP unicast service. Therefore, all end-hosts can enjoy multicast communications without IP multicast service. However, ALM has a serious problem that the multicast distribution tree is intrinsically fragile and an end-host failure causes tree partitions. In this paper, to deal with this problem, we propose a new tree construction protocol which makes outdegrees of intermediate nodes be balanced. The degree-balanced distribution tree can reduce the average number of nodes decoupled by tree partitions. To investigate performance of our protocol, it is compared with an existing ALM protocol. Our simulation results show that our protocol outperforms the existing protocol from the viewpoints of robustness, loss probability and receiver-perceived delay.

This paper introduces new methods of robust speech recognition using discrete-mixture HMMs (DMHMMs). The aim of this work is to develop robust speech recognition for adverse conditions that contain both stationary and non-stationary noise. In particular, we focus on the issue of impulsive noise, which is a major problem in practical speech recognition system. In this paper, two strategies were utilized to solve the problem. In the first strategy, adverse conditions are represented by an acoustic model. In this case, a large amount of training data and accurate acoustic models are required to present a variety of acoustic environments. This strategy is suitable for recognition in stationary or slow-varying noise conditions. The second is based on the idea that the corrupted frames are treated to reduce the adverse effect by compensation method. Since impulsive noise has a wide variety of features and its modeling is difficult, the second strategy is employed. In order to achieve those strategies, we propose two methods. Those methods are based on DMHMM framework which is one type of discrete HMM (DHMM). First, an estimation method of DMHMM parameters based on MAP is proposed aiming to improve trainability. The second is a method of compensating the observation probabilities of DMHMMs by threshold to reduce adverse effect of outlier values. Observation probabilities of impulsive noise tend to be much smaller than those of normal speech. The motivation in this approach is that flooring the observation probability reduces the adverse effect caused by impulsive noise. Experimental evaluations on Japanese LVCSR for read newspaper speech showed that the proposed method achieved the average error rate reduction of 48.5% in impulsive noise conditions. Also the experimental results in adverse conditions that contain both stationary and impulsive noises showed that the proposed method achieved the average error rate reduction of 28.1%.

Automated reasoning of inductive theorems is considered important in program verification. To verify inductive theorems automatically, several implicit induction methods like the inductionless induction and the rewriting induction methods have been proposed. In studying inductive theorems on higher-order rewritings, we found that the class of the theorems shown by known implicit induction methods does not coincide with that of inductive theorems, and the gap between them is a barrier in developing mechanized methods for disproving inductive theorems. This paper fills this gap by introducing the notion of primitive inductive theorems, and clarifying the relation between inductive theorems and primitive inductive theorems. Based on this relation, we achieve mechanized methods for proving and disproving inductive theorems.

In recent years, considerable attention has been devoted to continuously running software systems whose performance characteristics are smoothly degrading in time. Software aging often affects the performance of a software system and eventually causes it to fail. A novel approach to handle transient software failures due to software aging is called software rejuvenation, which can be regarded as a preventive and proactive solution that is particularly useful for counteracting the aging phenomenon. In this paper, we focus on a high assurance software system with fault-tolerance and preventive rejuvenation, and analyze the stochastic behavior of such a highly critical software system. More precisely, we consider a fault-tolerant software system with two-version redundant structure and random rejuvenation schedule, and evaluate quantitatively some dependability measures like the steady-state system availability and MTTF based on the familiar Markovian analysis. In numerical examples, we examine the dependence of two fault tolerant techniques; design and environment diversity techniques, on the system dependability measures.

The Modified edge representation (MER) is the concept to be used in the line integral approximation for computing the surface radiation integrals of diffraction. The MER as applied to the physical optics (PO-MER), has remarkable accuracy in the surface-to-line integral reduction even for the curved surfaces and for sources very close to the scatterer. In the discussion of the mathematical foundation for this accuracy, the evaluation of the singularities in the integrand of the PO-MER line integration was left for further study.

A simple method for interleaving operation suitable for paralleled converter system is proposed. This method automatically detects the number of converters and adjusts phases between converter modules equally for any number of modules in the system. The method is realized by simple analog circuit which is easily implemented as conventional PWM controller IC. Principle of multiphase controlling circuit is introduced, and the influence of non-ideal circuit parameters on interleaving operation are discussed. A compensator for reducing phase error is also proposed to achieve wide-use application. Experimental and analytical results confirm the effectiveness of the proposed method.

Characteristics of a class of stepped-impedance resonators (SIRs) which is loaded with two dielectric rods, are investigated by a Finite-Difference Time-Domain (FDTD) method. Dielectric rods to be inserted between a strip conductor and the ground plane have higher relative permittivity than that of the substrate. When a tapped half-wavelength (λ/2) microstrip resonator is loaded with two dielectric rods, the electric length of a loaded λ/2 resonator becomes longer than λ/2, which makes its fundamental resonant frequency () to be generated on the region lower than that of an unloaded λ/2 resonator (fr) and its first spurious response (fsp1) is generated on the region higher than 2. Therefore, to shift back to fr, the resonator's length is to be reduced, and this, in turns, suppress the spurious responses. Then, the resonant characteristics of an SIR employing the proposed method has also been investigated, and it is found that this is capable of suppressing the spurious responses in wideband together with an attenuation pole in the stopband, and of further reducing the resonator's length. Therefore, wide exploitation of the presented method can be expected in the filter design based on the LTCC technique.