A (k,δ,ε)-locally decodable code C:Fqn FqN is an error-correcting code that encodes =(x1,x2,...,xn) ∈ Fqn to C() ∈ FqN and has the following property: For any ∈ FqN such that d(,C()) ≤ δ N and each 1 ≤ i ≤ n, the symbol xi of can be recovered with probability at least 1-ε by a randomized decoding algorithm looking at only k coordinates of . The efficiency of a (k,δ,ε)-locally decodable code C:Fqn FqN is measured by the code length N and the number k of queries. For a k-query locally decodable code C:Fqn FqN, the code length N was conjectured to be exponential of n, i.e., N=exp(nΩ(1)), however, this was disproved. Yekhanin [In Proc. of STOC, 2007] showed that there exists a 3-query locally decodable code C:F2n F2N such that N=exp(n1/log log n) assuming that infinitely many Mersenne primes exist. For a 3-query locally decodable code C:Fqn FqN, Efremenko [ECCC Report No.69, 2008] further reduced the code length to N=exp(nO((log log n/ log n)1/2)), and in general showed that for any integer r>1, there exists a 2r-query locally decodable code C:Fqn FqN such that N=exp(nO((log log n/ log n)1-1/r)). In this paper, we will present improved constructions for query-efficient locally decodable codes by introducing a technique of "composition of locally decodable codes," and show that for any integer r>5, there exists a 9 2r-4-query locally decodable code C:Fqn FqN such that N=exp(nO((log log n/ log n)1-1/r)).

Privacy preserving association rule mining algorithms have been designed for discovering the relations between variables in data while maintaining the data privacy. In this article we revise one of the recently introduced schemes for association rule mining using fake transactions (fs). In particular, our analysis shows that the fs scheme has exhaustive storage and high computation requirements for guaranteeing a reasonable level of privacy. We introduce a realistic definition of privacy that benefits from the average case privacy and motivates the study of a weakness in the structure of fs by fake transactions filtering. In order to overcome this problem, we improve the fs scheme by presenting a hybrid scheme that considers both privacy and resources as two concurrent guidelines. Analytical and empirical results show the efficiency and applicability of our proposed scheme.

In the present paper, we propose an evolutionary P2P networking technique that dynamically and adaptively optimizes several P2P network topologies, in which all of the nodes are included at the same time, in an evolutionary manner according to given evaluation criteria. In addition, through simulations, we examine whether the proposed evolutionary P2P networking technique can provide reliable search capability in dynamic P2P environments. In simulations, we assume dynamic P2P environments in which each node leaves and joins the network with its own probability and in which search objects vary with time. The simulation results show that topology reconstruction by the evolutionary P2P networking technique is better than random topology reconstruction when only a few types of search objects are present in the network at any moment and these search objects are not replicated. Moreover, for the scenario in which the evolutionary P2P networking technique is more effective, we show through simulations that when each node makes several links with other nodes in a single network topology, the evolutionary P2P networking technique improves the reliable search capability. Finally, the number of links that yields more reliable search capability appears to depend on how often nodes leave and join the network.

It is well known that the performance of turbo codes can be improved by optimizing the energy allocation on coded symbols. Based on this fact, we propose an optimized 16-quadrature amplitude modulation (QAM) constellation for binary turbo codes. In the proposed scheme, the energy allocated on turbo coded symbols is optimized by modifying the constellation of QAM. The proposed 16-QAM constellation combined with a binary turbo code offers better coding gain compared to the conventional combination of binary turbo code and QAM.

To improve speech recognition performance, acoustic feature transformation based on discriminant analysis has been widely used. For the same purpose, discriminative training of HMMs has also been used. In this letter we investigate the effectiveness of these two techniques and their combination. We also investigate the robustness of matched and mismatched noise conditions between training and evaluation environments.

This paper represents a feedback artificial immune system (FAIS). Inspired by the feedback mechanisms in the biological immune system, the proposed algorithm effectively manipulates the population size by increasing and decreasing B cells according to the diversity of the current population. Two kinds of assessments are used to evaluate the diversity aiming to capture the characteristics of the problem on hand. Furthermore, the processing of adding and declining the number of population is designed. The validity of the proposed algorithm is tested for several traveling salesman benchmark problems. Simulation results demonstrate the efficiency of the proposed algorithm when compared with the traditional genetic algorithm and an improved clonal selection algorithm.

This paper investigates the performance of ARQ-aided downlink Time Switched Transmit Diversity (TSTD) in the WCDMA Low Chip Rate (LCR)-Time Division Duplex (TDD) system, when antenna switching and power ramping are applied. With the help of the ARQ signal, where the receiver sends the acknowledgement (ACK or NACK) to the transmitter, the proposed TSTD scheme switches the transmit antenna and ramps up the transmission power for the retransmitted data, when the transmitter receives a NACK signal. Simulation results demonstrate, that when the mobile speed is 3 km/h and a frame error rate (FER) is set to 1%, the antenna switching scheme yields 2 dB to 3 dB performance gain in terms of average Eb/N0, and the power ramping gives 0.7 dB to 1.6 dB gain, compared with the conventional ARQ-aided TSTD. In addition, 6% of throughput gain is shown by amalgamating the antennas switching as well as the power ramping, when the average Eb/N0 is equal to 0 dB.

This paper describes a flash ADC using interpolation (IP) and cyclic background self-calibrating techniques. The proposed IP technique that is cascade of capacitor IP and gate IP with dynamic double-tail latched comparator reduces non-linearity, power consumption, and occupied area. The cyclic background self-calibrating technique periodically suppresses offset mismatch voltages caused by static fluctuation and dynamic fluctuation due to temperature and supply voltage changes. The ADC has been fabricated in 90-nm 1P10M CMOS technology. Experimental results show that the ADC achieves SNDR of 6.07 bits without calibration and 6.74 bits with calibration up to 500 MHz input signal at sampling rate of 600 MSps. It dissipates 98.5 mW on 1.2-V supply. FoM is 1.54 pJ/conv.

This paper presents a novel approach for robot localization using landmark maps. With recent progress in SLAM researches, it has become crucial for a robot to obtain and use large-size maps that are incrementally built by other mapper robots. Our localization approach successfully works with such incremental and large-size maps. In literature, RANSAC map-matching has been a promising approach for large-size maps. We extend the RANSAC map-matching so as to deal with incremental maps. We combine the incremental RANSAC with an incremental LSH database and develop a hybrid of the position-based and the appearance-based approaches. A series of experiments using radish dataset show promising results.

A technique for computing the quotient (⌊ ab/n ⌋) of Euclidean divisions from the difference of two remainders (ab (mod n) - ab (mod n+1)) was proposed by Fischer and Seifert. The technique allows a 2-bit modular multiplication to work on most -bit modular multipliers. However, the cost of the quotient computation rises sharply when computing modular multiplications larger than 2 bits with a recursive approach. This paper addresses the computation cost and improves on previous 2-bit modular multiplication algorithms to return not only the remainder but also the quotient, resulting in an higher performance in the recursive approach, which becomes twice faster in the quadrupling case and four times faster in the octupling case. In addition to Euclidean multiplication, this paper proposes a new 2-bit Montgomery multiplication algorithm to return both of the remainder and the quotient.

This paper presents a single-chip speech dialogue module and its evaluation on a personal robot. This module is implemented on an application processor that was developed primarily for mobile phones to provide a compact size, low power-consumption, and low cost. It performs speech recognition with preprocessing functions such as direction-of-arrival (DOA) estimation, noise cancellation, beamforming with an array of microphones, and echo cancellation. Text-to-speech (TTS) conversion is also equipped with. Evaluation results obtained on a new personal robot, PaPeRo-mini, which is a scale-down version of PaPeRo, demonstrate an 85% correct rate in DOA estimation, and as much as 54% and 30% higher speech recognition rates in noisy environments and during robot utterances, respectively. These results are shown to be comparable to those obtained by PaPeRo.

In this Letter, a residual acoustic echo suppression method is proposed to enhance the speech quality of hands-free communication in an automobile environment. The echo signal is normally a human voice with harmonic characteristics in a hands-free communication environment. The proposed algorithm estimates the residual echo signal by emphasizing its harmonic components. The estimated residual echo is used to obtain the signal-to-interference ratio (SIR) information at the acoustic echo canceller output. Then, the SIR based Wiener post-filter is constructed to reduce both the residual echo and noise. The experimental results confirm that the proposed algorithm is superior to the conventional residual echo suppression algorithm in terms of the echo return loss enhancement (ERLE) and the segmental signal-to-noise ratio (SEGSNR).

This letter proposes the Inference Algorithm through Effective Slot Allocation (ESA-IA). In ESA-IA, the tags which match the prefix of the reader's request-respond in the corresponding slot; the group of tags with an even number of 1's responds in slot 0, while the group with an odd number of 1's responds in slot 1. The proposed algorithm infers '00' and '11' if there are two collided bits in slot 0, while inferring '01' and '10' if there are two collided bits in slot 1. The ESA-IA decreases the time consumption for tag identification by reducing the overall number of queries.

A rigorous semi-analytical approach for the scalar field in a microstructured optical fiber, which is formed of layered cylindrical arrays of circular rods symmetrically distributed on each concentric cylindrical layer, is presented. The method uses the T-matrix of a circular rod in isolation and the generalized reflection and transmission matrices of cylindrical arrays. Numerical examples of the mode index for three-layered hexagonal structure of circular air holes are demonstrated and compared with those obtained by a variational method.

The NTRU cryptosystem is a public key system based on lattice problems. While its theoretical security has been well studied, little effort has been made to analyze its security against implementation attacks including power analysis attacks. In this paper, we show that a typical software implementation of NTRU is vulnerable to the simple power analysis and the correlation power analysis including a second-order power attack. We also present novel countermeasures to prevent these attacks, and perform experiments to estimate the performance overheads of our countermeasures. According to our experimental results, the overheads in required memory and execution time are only 8.17% and 9.56%, respectively, over a Tmote Sky equipped with an MSP430 processor.

Recently, proportionate adaptive algorithms have been proposed to speed up convergence in the identification of sparse impulse response. Although they can improve convergence for sparse impulse responses, the steady-state misalignment is limited by the constant step-size parameter. In this article, based on the principle of least perturbation, we first present a derivation of normalized version of proportionate algorithms. Then by taking the disturbance signal into account, we propose a variable step-size proportionate NLMS algorithm to combine the benefits of both variable step-size algorithms and proportionate algorithms. The proposed approach can achieve fast convergence with a large step size when the identification error is large, and then considerably decrease the steady-state misalignment with a small step size after the adaptive filter reaches a certain degree of convergence. Simulation results verify the effectiveness of the proposed approach.

For achieving low cross-polarization component in addition to circular-coverage pattern in compact structure, this paper proposes a novel multimode horn with arbitrary coaxial-corrugation configuration which plays two roles of mode converters and chokes. The proposed horn can be designed by iteration of non-linear optimization procedure based on generalized scattering matrices pre-calculated by the mode-matching technique. We show a compact horn with four coaxial corrugations for shaping circular-coverage beam over frequency range of bandwidth 20%. The effectiveness of the designed horn is discussed by evaluating VSWR and radiation characteristics in X-band numerically and experimentally.

Many existing software reliability models (SRMs) are based on the assumption that fault correction activities take a negligible amount of time and resources, which is often invalid in real-life situations. Consequently, the estimated and predicted software reliability tends to be over-optimistic, which could in turn mislead management in related decision-makings. In this paper, we first make an in-depth analysis of real-life software testing process; then a Markovian SRM considering fault correction process is proposed. Parameter estimation method and software reliability prediction method are established. A numerical example is given which shows that by using the proposed model and methods, the results obtained tend to be more appropriate and realistic.

In this paper, we propose a rapid model adaptation technique for emotional speech recognition which enables us to extract paralinguistic information as well as linguistic information contained in speech signals. This technique is based on style estimation and style adaptation using a multiple-regression HMM (MRHMM). In the MRHMM, the mean parameters of the output probability density function are controlled by a low-dimensional parameter vector, called a style vector, which corresponds to a set of the explanatory variables of the multiple regression. The recognition process consists of two stages. In the first stage, the style vector that represents the emotional expression category and the intensity of its expressiveness for the input speech is estimated on a sentence-by-sentence basis. Next, the acoustic models are adapted using the estimated style vector, and then standard HMM-based speech recognition is performed in the second stage. We assess the performance of the proposed technique in the recognition of simulated emotional speech uttered by both professional narrators and non-professional speakers.

Recently, a new multi-carrier CDMA (MC-CDMA) system with cyclic-shift orthogonal keying (CSOK) has been proposed and shown to be more spectral and power efficient than conventional MC-CDMA systems. In this paper, a novel extension called the multiplexed CSOK (MCSOK) MC-CDMA system is proposed to further increase the data rate while maintaining a low peak-to-average power ratio (PAPR). First, the data stream is divided into multiple parallel substreams that are mapped into QPSK-CSOK symbols in terms of cyclic shifted Chu sequences. Second, these sequences are repeated, modulated, summed, and placed on IFFT subcarriers, resulting in a constant-modulus multiplexed signal that preserves the desired orthogonality among substreams. The receiver performs frequency-domain equalization and uses efficient demultiplexing, despreading, and demapping schemes to detect the modulation symbols. Furthermore, an alternate MCSOK system configuration with high link quality is also presented. Simulations show that the proposed MCSOK system attains lower PAPR and BER, as compared to conventional MC-CDMA system using Walsh codes. Under a rich multipath environment, the high link quality configuration exhibits excellent performance with both diversity gain and MCSOK modulation gain.