In this article, we will present a modification attack and a dictionary attack to subvert the security of the Tseng scheme and the Ku-Wang scheme. As we know, no existing schemes of simple authenticated key agreement protocols can successfully withstand our modification attack.

This paper proposes a PCB plane model for generic circuit simulators (SPICE). The proposed model reflects two frequency-dependent losses, namely, skin and dielectric losses. Once power/ground plane pair is divided into arrays of unit-cells, each unit-cell is modeled using a transmission line and the loss model. The loss model is composed of a resistor for DC loss, series RL ladder circuit for skin loss and series RC ladder circuit for dielectric loss. To verify the validity of the proposed model, it is compared with SPICE ac analysis using frequency-dependent resistors. Also, we show that the estimation results using the proposed model have a good correlation with that of VNA measurement for the typical PCB stack-up structure of general desktop PCs. With the proposed model, not only ac analysis but also transient analysis can be easily done for circuits including various non-linear/linear devices since the model consists of passive elements only.

We newly propose a novel blind separation framework for Single-Input Multiple-Output (SIMO)-model-based acoustic signals using an extended ICA algorithm, SIMO-ICA. The SIMO-ICA consists of multiple ICAs and a fidelity controller, and each ICA runs in parallel under the fidelity control of the entire separation system. The SIMO-ICA can separate the mixed signals, not into monaural source signals but into SIMO-model-based signals from independent sources as they are at the microphones. Thus, the separated signals of SIMO-ICA can maintain the spatial qualities of each sound source. In order to evaluate its effectiveness, separation experiments are carried out under both nonreverberant and reverberant conditions. The experimental results reveal that the signal separation performance of the proposed SIMO-ICA is the same as that of the conventional ICA-based method, and that the spatial quality of the separated sound in SIMO-ICA is remarkably superior to that of the conventional method, particularly for the fidelity of the sound reproduction.

In order to produce precise enclosures from a multi-dimensional interval vector, we introduce a sharp interval sub-dividing condition for optimization algorithms. By utilizing interval inclusion properties, we also enhance the sampling of an upper bound for effective use in the interval quadratic method. This has resulted in an improvement in the algorithm for the unconstrained optimization problem by Hansen in 1992.

A new and fast fingerprint classification method based on direction patterns is presented in this paper. This method is developed to be applicable to today's embedded fingerprint authentication system, in which small area sensors are widely used. Direction patterns are well treated in the direction map at block level, where each block consists of 88 pixels. It is demonstrated that the search of directions pattern in specific area, generally called as pattern area, is able to classify fingerprints clearly and quickly. With our algorithm, the classification accuracy of 89% is achieved over 4000 images in the NIST-4 database, slightly lower than the conventional approaches. However, the classification speed is improved tremendously up to about 10 times as fast as conventional singular point approaches.

This paper provides a novel signal processing for detecting defects based on the spatio-temporal gradient analysis over the Lamb-wave field. The proposed processing classifies the wave field through the rank of the covariance matrix which is defined by the four-dimensional vector with following components: a vertical displacement, its vertical velocity, and a pair of out-of-plane shearing strains. The covariance matrix provides the information about defects. Its determinant, therefore, is proposed as the inhomogeneity-index of the object surface. In this study, the physical meanings of the proposed index are shown, the computational process in the Lamb-wave field near the defects is discussed and their behaviors are investigated through FDTD-simulations and acoustic experiments.

Classical designs of maximally flat finite impulse response digital filters need to perform inverse discrete Fourier transformation of the frequency responses, in order to calculate the impulse response coefficients. Several attempts have been made to simplify the designs by obtaining explicit formulas for the impulse response coefficients. Such formulas have been derived for digital differentiators having maximal linearity at zero frequency, using different techniques including interpolating polynomials and the Taylor series etc. We show that these formulas can be obtained directly by application of maximal linearity constraints on the frequency response. The design problem is formulated as a system of linear equations, which can be solved to achieve maximal linearity at an arbitrary frequency. Certain special characteristics of the determinant of the coefficients matrix of these equations are explored for designs centered at zero frequency, and are used in derivation of explicit formulas for the impulse response coefficients of digital differentiators of both odd and even lengths.

Novel contact materials such as extrinsic conducting polymers can improve contact performance and enable device miniaturisation. This paper presents an investigation of conducting polymer materials used to minimise the effect of fretting slip at the contact interface of connector terminals. Initial experimental studies are presented and a mechanical model is used to describe the polymer interface slip and stick characteristics. For fretting to be minimised the polymer must be sufficiently elastic, with a high coefficient of friction with the contacting terminal, and a sufficient normal force to ensure no slip. Experimental studies of a polymer-tin interface are conducted to characterise contact resistance performance under fretting conditions. A resistance model of the polymer interface is developed.

This paper describes a noise robustness technique that normalizes the cepstral amplitude range in order to remove the influence of additive noise. Additive noise causes speech feature mismatches between testing and training environments and it degrades recognition accuracy in noisy environments. We presume an approximate model that expresses the influence by changing the amplitude range and the DC component in the log-spectra. According to this model, we propose a cepstral amplitude range normalization (CARN) that normalizes the cepstral distance between maximum and minimum values. It can estimate noise robust features without prior knowledge or adaptation. We evaluated its performance in an isolated word recognition task by using the Noisex92 database. Compared with the combinations of conventional methods, the CARN could improve recognition accuracy under various SNR conditions.

Diagonal algebraic space time (DAST) block codes was proved to achieve the full transmit diversity over a quasi-static fading channel and to maintain 1 symbol/s/Hz. When the number of transmit antennas employed is larger than 2, DAST codes outperform the codes from orthogonal design with the equivalent spectral efficiency. However, due to the limitation on the signal constellation with complex integer points, no good 3bits/symbol DAST block code was given previously. In this paper, we propose a general form of 8-star-PSK constellations with integer points and present some theoretical results on the performance of the equivalent 8-star-PSK modulations. By using our proposed 8-star-PSKs, we present a searching algorithm to construct DAST codes with 3 bits per symbol under some criteria and investigate their performances over flat Rayleigh fading channels. It is shown that (5,2) 8-star-PSK scheme has a comparable performance to conventional 8PSK over additive white Gaussian noise (AWGN) channel and the corresponding DSAT codes constructed can achieve significant performance gain over flat Rayleigh fading channel.

The vacuum arc root properties (temperature, current density, radius of an arc root, fraction of current density carried by electrons, evaporation rate etc.) of graphite and copper are calculated with the cathode-fall voltage as parameter and the arc root properties of graphite are compared with that of copper. Especially, there is big difference between the evaporation rate of graphite and that of copper. This reason is thought that the thermal conductivity of graphite is low and its evaporating temperature is high.

To measure the arc motion in interruption process of low voltage molded case circuit breakers (MCCBs) more precisely, a set of novel 2-D optical fiber system is developed. To improve the spatial resolution of optical fibers, lens with inhomogeneous dielectric is fixed on the top of each fiber. Furthermore, the full hardware control logic facilitates the real-time, synchronous and high-speed processing and breaks through the restricted bus operation frequency range and data stream capacity of microprocessor. The Publisher-Subscribe behavioral design pattern is applied to the software and the loosely coupled relationship between glyph and experimental data is once established, the graphic configuration can be implemented for simulation analysis, and the flexibility and applicability of the whole system are obviously improved. It demonstrates that the system provides a better research technique especially for new generation MCCB with gas driven arc.

We propose a new algorithm for overdetermined blind source separation (BSS) based on multistage independent component analysis (MSICA). To improve the separation performance, we have proposed MSICA in which frequency-domain ICA and time-domain ICA are cascaded. In the original MSICA, the specific mixing model, where the number of microphones is equal to that of sources, was assumed. However, additional microphones are required to achieve an improved separation performance under reverberant environments. This leads to alternative problems, e.g., a complication of the permutation problem. In order to solve them, we propose a new extended MSICA using subarray processing, where the number of microphones and that of sources are set to be the same in every subarray. The experimental results obtained under the real environment reveal that the separation performance of the proposed MSICA is improved as the number of microphones is increased.

This paper discusses the orbital motion and elevation properties of a quasi-zenith satellite system using circular orbits. The satellites are deployed on inclined geosynchronous orbits with identical sub-satellite loci on earth. The satellites trace the locus at even intervals. This satellite system can provide mobile satellite communications and navigation services at very high elevations to middle-latitude regions. In general, the orbital parameters of the satellite system are determined by numerical simulation to maximize the minimum elevation angle in areas where satellite services are to be provided. However, an understanding of the properties of the orbit and consequent elevation properties are important for efficient constellation design. This paper formulates the orbital motion of inclined geosynchronous circular orbits, including the relative motion to the rotating earth. Although elliptical orbit constellations are also possible and can gain higher elevation, only circular orbits, which can be accurately formulated without using an analytically unsolvable Kepler's equation, are discussed in this paper. Elevation properties are evaluated using the geocentric angle between the sub-satellite point and an arbitrary point in the intended service area. This angle is a typical parameter that can be derived as a single-valued function of the elevation at a specific point. Optimum orbital parameters for an intended service area can be easily estimated without numerical simulation using the results of the evaluation described in this paper. These results can also be used to infer whether a circular-orbit constellation is applicable to an intended service area.

We have developed a code-division-multiplexing (CDM) transmission scheme for future cellular communication systems, which uses cyclic shifted-and-extended (CSE) codes generated from an M-sequence to enable seamless communication in highly mobile environments. Because the correlation characteristics of CSE codes are determined by the M-sequence, the cross-correlation values are accumulated as a result of combining transmitted signals with opposite polarities in parallel channels. The accumulated cross-correlation values significantly degrade transmission performance, especially with multi-level modulation schemes such as quadrature amplitude modulation (QAM). We thus propose a cancellation technique to eliminate the accumulated cross-correlation values. We have evaluated the transmission performance of the CDM transmission scheme with the proposed technique by computer simulation. The new scheme enables high-quality data transmission in fast-fading channels.

The AMR wideband speech codec was recently developed for high-quality wideband speech communications. Although it has an excellent performance due to expanded bandwidth of speech signal, it requires a huge amount of computation especially in codebook search. To solve this problem, this paper proposes an efficient codebook search method for AMR wideband codec. Starting from a poorly performing initial codevector, the proposed method enhances the performance of the codevector iteratively by exchanging the worst pulse in the codevector with a better one after evaluating the role of each pulse. Simulations show that the AMR wideband codec adopting the proposed codebook search method provides better performance with much less computational load than that using the standard method.

In this letter, adaptive multi-stage parallel interference cancellation (PIC) receiver is considered for multi-rate DS-CDMA system. In each stage of the adaptive multi-stage PIC receiver, multiple access interference (MAI) estimates are obtained by the sub-bit estimates from the previous stage and the adaptive weights for the sub-bit estimates. The adaptive weights are obtained by minimizing the mean squared error between the received signal and its estimate through normalized least mean square (LMS) algorithm. It is shown that the adaptive multi-stage PIC receiver achieves smaller BER than the matched filter receiver, multi-stage PIC receiver, and multi-stage partial PIC receiver for the multi-rate DS-CDMA system in a Rayleigh fading channel.

A method of scrambling MPEG video by exchanging the motion vector (MV) in the MPEG bitstream is proposed. It deals directly with the MPEG bitstream and exclusive MPEG encoders are unnecessary. The size of the scrambled bitstream does not increase and image quality is maintained after descrambling. Moreover, the structure of the MPEG bitstream is maintained and can be decoded with a standard MPEG video decoder. We demonstrate the effectiveness of this method through simulation results that reveal unchanged image quality and size of bitstreams.

The conventional algorithms in the echo canceling system have drawback when they are faced with double-talk condition in noisy environment. Since the double-talk and noise signal are exist, then the error signal is contaminated to estimate the gradient correctly. In this paper, we define a new class of adaptive algorithm for tap adaptations, based on the correlation function processing. The computer simulation results show that the Correlation LMS (CLMS) and the Extended CLMS (ECLMS) algorithms have better performance than conventional LMS algorithm. In order to implement the ECLMS algorithm, the Frequency domain Extended CLMS (FECLMS) algorithm is proposed to reduce the computational complexity. However the convergence speed is not sufficient. In order to improve the convergence speed, the Wavelet domain Extended CLMS (WECLMS) algorithm is proposed. The computer simulation results support the theoretical findings and verify the robustness of the proposed WECLMS algorithm in the double-talk situation.

A new algorithm, termed reduced-order Root-MUSIC, for high resolution direction finding is proposed. It requires finding all the roots of a polynomial with an order equaling twice the number of propagating signals. Some Monte Carlo simulations are used to test the effectiveness of the proposed algorithm.