This paper analyzes a generalized secret-key authentication system from a viewpoint of the information-spectrum methods. In the generalized secret-key authentication system, for each n 1 a legitimate sender transmits a cryptogram Wn to a legitimate receiver sharing a key En in the presence of an opponent who tries to cheat the legitimate receiver. A generalized version of the Simmons' bounds on the success probabilities of the impersonation attack and a certain kind of substitution attack are obtained.

One of the basic GPS features is its inherent anti-jamming (AJ) characteristics based on great processing gain as a result of using direct sequence spread spectrum (DS-SS) technique. For calculated jamming threshold for GPS receivers operating with L1 C/A, L1 P(Y) and L2 P(Y) signals, jamming range has been analyzed with respect to the characteristic jamming signals and under various effective radiated power of jamming signals. The tests of C/A GPS receiver jamming resistance on different jamming strategies, during the tracking and acquisition phase have been performed. The influence of jamming strategy on jamming efficiency has been proved. At the same time AJ characteristics of GPS receiver have been checked.

In this paper, the distributions of two spectral intensities along the axis of an arc column of the breaking arc are measured by using a combination of a CCD color camera and an additional filter, and arc temperature and metal-vapor quantity are calculated, when copper electrodes interrupt circuits of dc 50 V/3.3 A and 5.0 A. As results; The spectral intensities of excited copper atoms are the strongest near the cathode and become weaker with distance from the cathode in the small number of breaking arcs. The spectral intensities become strong near the anode in the large number of breaking arcs. The average arc temperature in the cross-section of an arc column is high near both the cathode and the anode, and the temperature distribution in the cross-section of the arc column is high at the axis of the arc column, and the arc temperature along the axis of the arc column is high near both the cathode and the anode. The metal-vapor quantity is low near both the cathode and the anode, and it is much at the center of the arc column.

The interconnect analysis of on- and off-chips is very important in the design of high-speed signal processing, digital communication, and microwave electronic systems. When the interconnects are characterized by sampled data via electromagnetic analysis, the circuit-level simulation of the network requires rational approximation of the sampled data. Since the frequency band of the sampled data is more than 10 GHz, the rational function must fit into it at many frequency points. The rational function is approximated using the orthogonal least-squares method. With an increase in the number of the fitting data, the least-squares method suffers from a singularity problem. To avoid this, the sampled data are hierarchically approximated in this paper. Moreover, to reduce the computational cost of the circuit-level simulation, the parameter matrix of the interconnects is approximated by a rational matrix with one common denominator polynomial, and the selective orthogonalization procedure is presented.

Flow control algorithm in high speed networks is a resource-sharing policy implemented in a distributed manner. This paper introduces a novel concept of backlog balancing and demonstrates its application to network flow control and congestion control by presenting a rate-based flow control algorithm for ATM networks. The aim of flow control is to maximize the network utilization for achieving high throughput with tolerable delay for each virtual circuit (VC). In a resource-sharing environment, this objective may also cause network congestion when a cluster of aggressive VC's are contending for the same resource at a particular node. The basic idea of our algorithm is to adjust the service rate of each node along a VC according to backlog discrepancies between neighboring nodes (i.e., to reduce the backlog discrepancy). The handshaking procedure between any two consecutive nodes is carried out by a link-by-link binary feedback protocol. Each node will update its service rate periodically based on a linear projection model of the flow dynamics. The updated service rate per VC at a node indicates its explicit demand of bandwidth, so a service policy implementing dynamic bandwidth allocation is introduced to enforce such demands. Simulation study has validated the concept and its significance in achieving the goal of flow control and yet preventing network congestion at the same time.

Speckle statistically brings series connections of dark pixels, which can be observed as dark line features in synthetic aperture radar (SAR) images. The dark lines have no physical meaning. In this paper, line features of that kind in high-resolution SAR images whose intensity obeys a K-distribution are studied. It is stochastically explained that the dark line features in 1-look K-distributed images can be observed more distinctly than those in exponential distributed images. It is further revealed that such line features are detectable enough, even if the K-distributed images are multilooked. The experiments on simulated images as well as on actual SAR images confirm the explanation.

The use of intelligent agents is on the rise, fueled by the unprecedented growth in the Internet and web based applications. Consequently, agent-oriented software is becoming large and complex. To support a systematic development of such software, an agent-oriented software development methodology is necessary. This paper focuses on the modeling phase of agent-oriented software life cycle and, presents an approach for agent modeling consisting of Agent Elicitation, Intra, and Inter Agent modeling methods. Agent Elicitation deals with identifying and extracting agents from "classes" in the real world. Intra Agent Modeling involves expressing agent characteristics - Goal, Belief, Plan and Capability - whereas, Inter Agent modeling incorporates agent mobility and communication in a multi-agent system.

Developments in new frequency bands for wireless communications make a broadband channel for new services possible. Great effort has been made researching and developing broadband wireless communication in the 60-GHz millimeter-wave band since the early 1990s. In this paper, we design an ATM (asynchronous transfer mode)-based indoor millimeter-wave wireless local area network (WLAN) that supports multimedia transmissions and focus on the wireless access topic for implementation of wireless ATM. We propose an integrated multimedia transmission protocol, based on the MAC (medium access control) protocol, called RS-ISMA (reservation-based slotted idle signal multiple access). It supports CBR (constant bit rate), VBR (variable bit rate), ABR (available bit rate) and UBR (unspecified bit rate) transmissions and provides QoS (quality of service)-dependent adaptive retransmissions. An RS-ISMA-based prototype full-duplex indoor high-speed WLAN in the 60-GHz band was developed.

This paper presents an efficient public-key protocol for mutual authentication and key exchange designed for mobile information services in UMTS. Certain modifications to the ASPeCT registration protocol is proposed to introduce Temporary Mobile User Certificate in UMTS environments. Proposed Temporary Mobile User Certificate, digitally-signed by the private signing key of the TTP located in the user's visiting domain, will be issued to the user during the registration protocol. ASPeCT AIP protocol can be modified to reduce communication overhead and speed up verification process using Temporary Mobile User Certificate between the user and the Value-Added Service Provider (VASP) located in the user's visiting domain. Furthermore, Temporary Mobile User Certificate can provide anonymity for a mobile user during the ASPeCT AIP protocol.

This paper proposes a new digital beamforming adaptive array antenna (DBFAAA) that is effective in severe multipath environments in which timing and carrier synchronization circuits cannot function ideally resulting in the DBFAAA losing control. The proposed DBFAAA has two stages. In the first, the DBFAAA captures the desired signal and establishes synchronization. In the second, the DBFAAA optimizes the beam pattern of the signal. The proposed configuration employs an eigenvector beam of the maximum eigenvalue in the first stage beam-forming. In addition, a fractionally-spaced-tapped-delay-line (FS-TDL) with real tap weights, which is placed after the beam-former, is applied to achieve timing synchronization. The behavior of the proposed DBFAAA for asynchronous sampling data is investigated and the results indicate that the proposed configuration enables asynchronous sampling at the A/D converter. A prototype of the proposed DBFAAA achieving 38-Mbps real-time data communication is introduced and the transmission performance is shown.

This paper presents a new technique for the synthesis of orthogonal-base-set sequences suitable for applications requiring sets of uncorrelated pseudo-white-noise sources. The synthesized sequences (vectors) are orthogonal to each other, and each sequence also has a flat power spectrum and low peak factor. In order to construct the orthogonal-base-set sequences, the new application of ta-sequence (trigonometric function aliasing sequence) introduced in this paper uses Latin-squares and Walsh-Hadamard sequences. The ta-sequence itself is a very new concept, and the method presented here provides the means for generating various orthogonal-base-set sequences at sizes required for such applications as system measurement (needing uncorrelated test signals), pseudo noise synthesis for spread spectrum communication, and audio signal processing (needing synthesis of stereo or multichannel signals from mono sources).

We examine the detection of coded hybrid direct-sequence frequency-hopped spread-spectrum signals in the presence of narrowband interference. Since blind interference suppression requires a reliable estimate of the data, while at the same time data decoding requires interference suppression, we advocate an iterative ("turbo") detection scheme whereby information is exchanged between the interference suppressor and the soft-input soft-output decoder. Several suppression schemes are examined and compared. Simulation results show that this new scheme is robust, i.e., exhibits good performance under a modicum of assumptions on the interference structure. Turbo codes and convolutional codes are compared, showing that the former perform better.

To evaluate or compare the convergence speed of adaptive digital filters (ADF) with least mean squared (LMS) algorithm, the condition numbers of correlation matrices of tap-input vectors are often used. In this paper, however, the comparison of the conventional fullband ADF and the subband ADF based on their condition numbers is shown to be invalid. In some cases, the over-sampled subband ADF converges faster than the fullband ADF, although the former has larger condition numbers. To explain the above phenomenon, an expression for the convergence behavior of the subband ADF and simulation results are provided.

A Schottky contact model was implemented as a boundary condition for Monte Carlo device simulations. Unlike the ideal ohmic contact, the thermal equilibrium is unnecessary around the Schottky contact. Therefore, the wide region with high impurity concentration around the contact is not required to maintain the thermal equilibrium, which means that it is possible to avoid assigning a lot of particles to the low-field region. The validity of the present boundary condition for contacts was verified by simulating a rectifying characteristic of a Schottky barrier diode. As an application example using the present contact model, we simulated transport in n+nn+ structures with sub-0.1 µm channel lengths. We observed direction dependence of the electron velocity dispersion, which indicates that the direction dependence of the diffusion constant or the carrier temperature should be taken into account in the hydrodynamic simulation for sub-0.1 µm devices.

This paper proposes higher-order spread spectrum direct optical switching CDMA system and an aliasing canceler to remove the aliasing distortion caused by higher-order bandpass sampling. Theoretical analysis of the signal quality shows that the 3rd order bandpass sampling scheme can improve the carrier-to-interference-power ratio compared with the conventional 1st order bandpass sampling scheme, by 5 dB.

A generalized algorithm for designing an optimum VQ source codec in systems with channel coding is presented. Based on an AWGN channel model, the algorithm derives the distribution of the channel decoder soft-output and substitutes it in the expression for the system end-to-end distortion. The VQ encoder/decoder pair is then optimized by minimizing this end-to-end distortion. For a Gauss-Markov source, the proposed algorithm outperforms the conventional SOVQ source coding scheme by 5.0 dB in the decoded source SNR. Application of this algorithm for designing optimum low-bit-rate speech codec is given. A 4.0 kbps VQ based CELP codec is designed for performance evaluations, where all the CELP parameter encoder/decoder pairs are optimized by minimizing their end-to-end distortions, respectively. As a result, the speech distortion over the noisy channel is minimized. Subjective tests show that the proposed algorithm improves the decoded speech quality by 2.5 MOS relative to a regular SOVQ CELP speech coding system. The performances of the algorithm under channel mismatch conditions are also shown and discussed.

This paper presents a pseudo glottal excitation model for the type of linear prediction vocoders with speech being coded at 1.6 kbps. While unvoiced speech and silence intervals are processed with a stochastic codebook of 512 entries, a glottal codebook with 32 entries for voiced excitation is used to describe the glottal phase characteristics. Steps of formulating the pseudo glottal excitation for one pitch period consist of 1) applying a polynomial model to simulate the low-frequency constituent of the residual, 2) inserting a magnitude-adjustable pulse sequence to characterize the main excitation, and 3) introducing turbulent noise in series with the resulting excitation. Procedures are described for codebook construction in addition to analysis and synthesis of the pseudo glottal excitation. Results in a mean opinion score (MOS) test show that the quality produced by the proposed coder is almost as good as that by 4.8 kbps CELP coder for male utterances, but the quality for female utterances is yet somewhat inferior.

In this paper, we propose a peak-weighted cepstral lifter (PWL) for enhancing the spectral peaks of an all-pole model spectrum in the cepstral domain. The design parameter of the PWL is the degree of pole enhancement or pole shifting toward the unit circle. The optimal pole shifting factor is chosen by considering the sensitivity to spectral resonance peaks, the variability of cepstral variances, and the recognition accuracy. Next, we generalize the PWL so that the optimal shifting factor is adaptively determined in frame-by-frame basis. Compared with other cepstral lifters, a speech recognizer employing the frame-adaptive PWL provides better recognition performance.

In long-haul wavelength-division-multiplexed (WDM) transmission systems, signals with shorter and longer wavelengths have self-phase modulation group-velocity-dispersion (SPM-GVD) penalty caused by to the dispersion slope even after the dispersion-compensation at the receiver has been optimized. As a countermeasure, we have already proposed both pre-compensation and post-compensation of chromatic dispersion at the transmitter and receiver for each channel. This method can decrease the channel variation of path-averaged chromatic dispersion along the transmission line, and it can improve the eye opening of the waveform after transmission. We investigated the optimized parameter of chromatic dispersion and chirping at the transmitter. The optimized pre-dispersion compensation parameter R was about 50%. The optimized chirping parameter α was about 3 when the signal wavelength was less than the mean zero-dispersion wavelength. In a single-channel, 5.3-Gbit/s NRZ signal transmission experiment over a 4,760-km straight line, this method decreased SPM-GVD penalty. In a 32-channel, 5.3-Gbit/s WDM transmission experiment over 9,879 km using a circulating loop, this method improved Q-factors for the 1st and 32nd channels by more than 1.5 dB.

This paper presents a new method for statistical modelling of prosody control in speech synthesis. The proposed method, which is referred to as Constrained Tree Regression (CTR), can make suitable representation of complex effects of control factors for prosody with a moderate amount of learning data. It is based on recursive splits of predictor variable spaces and partial imposition of constraints of linear independence among predictor variables. It incorporates both linear and tree regressions with categorical predictor variables, which have been conventionally used for prosody control, and extends them to more general models. In addition, a hierarchical error function is presented to consider hierarchical structure in prosody control. This new method is applied to modelling of speech segmental duration. Experimental results show that better duration models are obtained by using the proposed regression method compared with linear and tree regressions using the same number of free parameters. It is also shown that the hierarchical structure of phoneme and syllable durations can be represented efficiently using the hierarchical error function.