This paper presents an enhanced side-channel attack using a phase-based waveform matching technique. Conventionally, side-channel attacks such as Simple Power Analysis (SPA) and Differential Power Analysis (DPA) capture signal waveforms (e.g., power traces) with a trigger signal or a system clock, and use a statistical analysis of the waveforms to reduce noise and to retrieve secret information. However, the waveform data often includes displacement errors, and this degrades the accuracy of the statistical analysis. The use of a Phase-Only Correlation (POC) technique makes it possible to estimate the displacements between the signal waveforms with higher resolution than the sampling resolution. The accuracy of side-channel attacks can be enhanced using the POC-based matching method. Also, a popular DPA countermeasure of creating distorted waveforms with random delays can be defeated by our method. In this paper, we demonstrate the advantages of the proposed method in comparison with conventional approaches of experimental DPA and Differential ElectroMagnetic Analysis (DEMA) against DES software and hardware implementations.

Human's ability to perceive elevation of a sound and distinguish whether a sound is coming from the front or rear strongly depends on the monaural spectral features of the pinnae. In order to realize an effective virtual auditory display by HRTF (head-related transfer function) customization, the pinna responses were isolated from the median HRIRs (head-related impulse responses) of 45 individual HRIRs in the CIPIC HRTF database and modeled as linear combinations of 4 or 5 basic temporal shapes (basis functions) per each elevation on the median plane by PCA (principal components analysis) in the time domain. By tuning the weight of each basis function computed for a specific height to replace the pinna response in the KEMAR HRIR at the same height with the resulting customized pinna response and listening to the filtered stimuli over headphones, 4 individuals with normal hearing sensitivity were able to create a set of HRIRs that outperformed the KEMAR HRIRs in producing vertical effects with reduced front/back ambiguity in the median plane. Since the monaural spectral features of the pinnae are almost independent of azimuthal variation of the source direction, similar vertical effects could also be generated at different azimuthal directions simply by varying the ITD (interaural time difference) according to the direction as well as the size of each individual's own head.

Various wireless systems are being developed to meet users' needs, and the rapid increase in frequency demand that accompanies the increasing popularity of wireless services means that more effective use of frequency resources is urgently needed. However, existing base stations are making no effort to use frequency resources effectively, and cooperation among wireless system base stations is needed to use frequency resources more effectively. Base stations can cooperate more efficiently if they are able to use multiple channels of many wireless systems simultaneously. We propose an autonomous adaptive base station (AABS) that can switch among various wireless systems the way software defined radio (SDR) base stations do. AABS can autonomously select and use the most suitable wireless system on the basis of user traffic and its hardware resources. Moreover, frequency resources are used effectively because AABS prevents unnecessary radio wave transmission when the number of users in the wireless systems decreases. AABS is also suitable for "multi-link communication" because it can use multiple channels of multiple wireless systems simultaneously. We developed AABS prototype and evaluated its performance. Our experimental and computer simulation results show the performance of AABS and its efficiency.

In this paper we propose an algorithm of factoring any integer N which has k different prime factors with the same bit-length, when about ()log2 N high-order bits of each prime factor are given. For a fixed ε, the running time of our algorithm is heuristic polynomial in (log2 N). Our factoring algorithm is based on a lattice-based algorithm of solving any k-variate polynomial equation over Z, which might be an independent interest.

We focus on the relationship between the linearization method and linear complexity and show that the linearization method is another effective technique for calculating linear complexity. We analyze its effectiveness by comparing with the logic circuit method. We compare the relevant conditions and necessary computational cost with those of the Berlekamp-Massey algorithm and the Games-Chan algorithm. The significant property of a linearization method is that it needs no output sequence from a pseudo-random number generator (PRNG) because it calculates linear complexity using the algebraic expression of its algorithm. When a PRNG has n [bit] stages (registers or internal states), the necessary computational cost is smaller than O(2n). On the other hand, the Berlekamp-Massey algorithm needs O(N2) where N ( 2n) denotes period. Since existing methods calculate using the output sequence, an initial value of PRNG influences a resultant value of linear complexity. Therefore, a linear complexity is generally given as an estimate value. On the other hand, a linearization method calculates from an algorithm of PRNG, it can determine the lower bound of linear complexity.

For the purpose of reducing the quantization noise and power consumption of UWB-OFDM transceiver, a new time domain-based interpolator and decimator structure is proposed to realize five-bit D/A and A/D converters in the five-bit 128-tone sigma-delta modulation (SDM) UWB-OFDM transceiver. The five-bit 128-tone SDM UWB-OFDM transceiver using time domain-based interpolator and decimator in place of time spreader and de-spreader can obtain time-domain spread spectrum processing gain and reduce quantization noise simultaneously. The structure of the five-bit 128-tone SDM A/D converter, which employs 32 parallel analog SDM circuits without up-sampling, is designed. Simulation results demonstrate that BER of the proposed five-bit 128-tone SDM D/A and A/D converters based on time domain-based interpolator and decimator scheme can satisfy the performance requirements of the five-bit 128-tone SDM UWB-OFDM transceiver for the QPSK, 16-QAM and 64-QAM modulations.

In this paper, we discuss the collision search for hash functions, mainly in terms of their advanced message modification. The advanced message modification is a collision search tool based on Wang et al.'s attacks. Two advanced message modifications have previously been proposed: cancel modification for MD4 and MD5, and propagation modification for SHA-0. In this paper, we propose a new concept of advanced message modification, submarine modification. As a concrete example combining the ideas underlying these modifications, we apply submarine modification to the collision search for SHA-0. As a result, we show that this can reduce the collision search attack complexity from 239 to 236 SHA-0 compression operations.

In this letter, we consider the uplink packet scheduling for non-real-time data users in a DS-CDMA system. As an effort to jointly optimize throughput and fairness, we formulate a time-span minimization problem incorporating the time-multiplexing of different simultaneous transmission schemes. Based on simple rules, we propose efficient scheduling algorithms and compare them with the optimal solution obtained by linear programming.

We propose a simple asynchronous UWB (Ultra Wide Band) position location algorithm with low complexity, power consumption and processing delay. In the proposed algorithm, only a single RTTX (Round-Trip Transmission) of UWB pulses is utilized based on the ToA (Time of Arrival) principle. Hence, the proposed algorithm decreases power consumption and processing delay as compared to the basic ToA based on triple RTTXs. Moreover, unlike the TDoA (Time Difference of Arrival) algorithm, the proposed algorithm can perform the position location with low complexity since it does not require strict synchronization between multiple beacons. Simulation results using IEEE 802.15.4a UWB channel models reveal that the proposed algorithm achieves closely comparable position location performance of the basic ToA and TDoA algorithms.

The concept of placing enormous Solar Power Satellite (SPS) systems in space represents one of a handful of new technological options that might provide large scale, environmentally clean base load power to terrestrial markets. Recent advances in space exploration have shown a great need for antennas with high resolution, high gain and low side lobe level (SLL). The last characteristic is of paramount importance especially for the Microwave Power Transmission (MPT) in order to achieve higher transmitting efficiency (TE) and higher beam collection efficiency (BCE). In order to achieve low side lobe levels, statistical methods play an important role. Various interesting properties of a large antenna arrays with randomly, uniformly and combined spacing of elements have been studied, especially the relationship between the required number of elements and their appropriate spacing from one viewpoint and the desired SLL, the aperture dimension, the beamwidth and TE from the other. We propose a new unified approach in searching for reducing SLL by exploiting the interaction of deterministic and stochastic workspaces of proposed algorithms. Our models indicate the side lobe levels in a large area around the main beam and strongly reduce SLL in the entire visible range. A new concept of designing a large antenna array system is proposed. Our theoretic study and simulation results clarify how to deal with the problems of side lobes in designing a large antenna array, which seems to be an important step toward the realization of future SPS/MPT systems.

Image Registration can be seen as an optimization problem to find a cost function and then use an optimization method to get its minimum. Normalized mutual information is a widely-used robust method to design a cost function in medical image registration. Its calculation is based on the joint histogram of the fixed and transformed moving images. Usually, only a discrete joint histogram is considered in the calculation of normalized mutual information. The discrete joint histogram does not allow the cost function to be explicitly differentiated, so it can only use non-gradient based optimization methods, such as Powell's method, to seek the minimum. In this paper, a parzen-window based method is proposed to estimate the continuous joint histogram in order to make it possible to derive the close form solution for the derivative of the cost function. With this help, we successfully apply the gradient-based optimization method in registration. We also design a new kernel for the parzen-window based method. Our designed kernel is a second order polynomial kernel with the width of two. Because of good theoretical characteristics, this kernel works better than other kernels, such as a cubic B-spline kernel and a first order B-spline kernel, which are widely used in the parzen-window based estimation. Both rigid and non-rigid registration experiments are done to show improved behavior of our designed kernel. Additionally, the proposed method is successfully applied to a clinical CT-MR non-rigid registration which is able to assist a magnetic resonance (MR) guided microwave thermocoagulation of liver tumors.

We experimentally demonstrated polarization-mode dispersion (PMD) compensation by distributing polarizers with only 1 degree of freedom (DOF) along the transmission line. The average power penalty was measured to be 0.4 dB by inserting four compensators, where average differential group delay was 47% of bit period.

We previously proposed a method of sound source and microphone localization. The method estimates the locations of sound sources and microphones from only time differences of arrival between signals picked up by microphones even if all their locations are unknown. However, there is a problem that some estimation results converge to local minimum solutions because this method estimates locations iteratively and the error function has multiple minima. In this paper, we present a new iterative method to solve the local minimum problem. This method achieves accurate estimation by selecting effective initial locations from many random initial locations. The computer simulation and experimental results demonstrate that the presented method eliminates most local minimum solutions. Furthermore, the computational complexity of the presented method is similar to that of the previous method.

For formal verification of large-scale digital circuits, a method using satisfiability checking of logic with equality and uninterpreted functions has been proposed. This logic, however, does not consider specific properties of functions or predicates at all, e.g. associative property of addition. In order to ease this problem, we introduce "equivalence constraint" that is a set of formulas representing the properties of functions and predicates, and check the satisfiability of formulas under the constraint. In this report, we show an algorithm for checking satisfiability with equivalence constraint and also experimental results.

Recently, many techniques have been proposed to improve speaker identification in noise environments. Among these techniques, we consider the feature recombination technique for the multi-band approach in noise robust speaker identification. The conventional feature recombination technique is very effective in the band-limited noise condition, but in broad-band noise condition, the conventional feature recombination technique does not provide notable performance improvement compared with the full-band system. Even though the speech is corrupted by the broad-band noise, the degree of the noise corruption on each sub-band is different from each other. In the conventional feature recombination for speaker identification, all sub-band features are used to compute multi-band likelihood score, but this likelihood computation does not use a merit of multi-band approach effectively, even though the sub-band features are extracted independently. Here we propose a new technique of sub-band likelihood computation with sub-band weighting in the feature recombination method. The signal to noise ratio (SNR) is used to compute the sub-band weights. The proposed sub-band-weighted likelihood computation makes a speaker identification system more robust to noise. Experimental results show that the average error reduction rate (ERR) in various noise environments is more than 24% compared with the conventional feature recombination-based speaker identification system.

A compact OTA suitable for low-voltage active-RC and MOSFET-C filters is presented. The input stage of the OTA utilises the NMOS pseudo-differential amplifier with PMOS active load. The output stage relies upon the dual-mode feed-forward class-AB technique (based on an inverter-type transconductor) with common-mode rejection capability that incurs no penalty on transconductance/bias-current efficiency. Simulation results of a 0.5-V 100-kHz 5th-order Chebyshev filter based on the proposed OTA in a 0.18 µm CMOS process indicate SNR and SFDR of 68 dB and 63 dB (at 50 kHz+55 kHz) respectively. The filter consumes total power consumption of 60 µW.

In this paper, we investigate a detection ordering scheme of OSIC (Ordered Successive Interference Cancellation) systems suitable for power controlled MIMO transmission. Most studies about power controlled systems have mainly focused on strategies for transmitter, while the ordering scheme optimized at open-loop system has not been modified. In a conventional ordering scheme, the ordering process is done according to the largeness and smallness relation of each sub-stream's SNR. Unlike the conventional scheme, we derive an optimized detection ordering scheme that uses proximity to the optimal SNR. Because of error propagation, our proximity based algorithm is not valid for open-loop MIMO system in many cases. An optimization problem analysis and simulation results show that the system using the proposed ordering scheme outperforms the system using the conventional ordering scheme. Furthermore, due to the nature of QR decomposition, the proposed scheme shows not only lower implementation complexity but also better BER performance compared with the conventional scheme based on pseudo-inverse.

This paper proposes a method for reinforcing noun countability prediction, which plays a crucial role in demarcating correct determiners in machine translation and error detection. The proposed method reinforces countability prediction by introducing a novel heuristics called one countability per discourse. It claims that when a noun appears more than once in a discourse, all instances will share identical countability. The basic idea of the proposed method is that mispredictions can be corrected by efficiently using one countability per discourse heuristics. Experiments show that the proposed method successfully reinforces countability prediction and outperforms other methods used for comparison. In addition to its performance, it has two advantages over earlier methods: (i) it is applicable to any countability prediction method, and (ii) it requires no human intervention to reinforce countability prediction.

The Ubiquitous sensor network (USN) node is required to operate for several months with limited system resources such as memory and power. The typical USN node is in the active state for less than 1% of its several month lifetime and waits in the inactive state for the remaining 99% of its lifetime. This paper suggests a power adjustment dual priority scheduler (PA-DPS) that offers low power consumption while meeting the USN requirements by estimating power consumption in the USN node. PA-DPS has been designed based on the event-driven approach and the dual-priority scheduling structure, which has been conventionally suggested in the real-time system field. From experimental results, PA-DPS reduced the inactive mode current up to 40% under the 1% duty cycle.

A simplified equalization method based on the band structure of the frequency domain channel matrix is proposed for the single carrier systems employing cyclic prefix (SC-CP) over time-varying wireless channels. Using both theoretical analysis and computer simulation, it is shown that the complexity of this method is proportional to the number of symbols in one SC-CP block and is less than that of traditional block equalization methods. We also show that they have similar performance.