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This paper describes the second part of our study devoted to automated anonymization of personal data. The overview and analysis of research prospects is supplemented with a practical result. An anonymization model is proposed, which reduces anonymization of personal data to manipulation of samples of random elements of different types. The key idea of our approach to anonymization of personal data with preservation of their usefulness is the use of the synthesis method, i.e., the complete replacement of all non-anonymized data with synthetic values. In the proposed model, a set of element types is selected, for which corresponding synthesys templates are proposed. The set of templates constitutes a synthesis-based anonymization algorithm. Technically, each template is based on a well-known statistical tool: frequency estimates of probabilities, Parzen–Rosenblatt kernel density estimates, statistical means, and covariances. The proposed approach is illustrated by a simple example from civil aviation. [ABSTRACT FROM AUTHOR]

The paper describes a new approach for non-destructive testing of the structural and geometric parameters of dielectric objects using capacitance techniques. The novelty of this approach lies in the design of a capacitance sensor comprising an array of electrodes with changeable potential distribution on them during a measurement process. This makes solvable the problem of measuring independently three input parameters. To demonstrate the capabilities of the developed measurement algorithms, the case studies based on computer simulation have been carried out. [ABSTRACT FROM AUTHOR]

In this paper, based on the gravity principle of classical physics, we propose a tunable gravity-based model, which considers tag usage pattern to weigh both the mass and distance of network nodes. We then apply this model in solving the problems of information filtering and network evolving. Experimental results on two real-world data sets, Del.icio.us and MovieLens, show that it can not only enhance the algorithmic performance, but can also better characterize the properties of real networks. This work may shed some light on the in-depth understanding of the effect of gravity model. [ABSTRACT FROM AUTHOR]

The accuracy of most of the existing semisupervised clustering algorithms based on small size of labeled dataset is low when dealing with multidensity and imbalanced datasets, and labeling data is quite expensive and time consuming in many real-world applications. This paper focuses on active data selection and semisupervised clustering algorithm in multidensity and imbalanced datasets and proposes an active semisupervised clustering algorithm. The proposed algorithm uses an active mechanism for data selection to minimize the amount of labeled data, and it utilizes multithreshold to expand labeled datasets on multidensity and imbalanced datasets. Three standard datasets and one synthetic dataset are used to demonstrate the proposed algorithm, and the experimental results show that the proposed semisupervised clustering algorithm has a higher accuracy and a more stable performance in comparison to other clustering and semisupervised clustering algorithms, especially when the datasets are multidensity and imbalanced. [ABSTRACT FROM AUTHOR]

Abstract: Integrated environmental modelling is gaining momentum for addressing grand scientific challenges such as monitoring the environment for change detection and forecasting environmental conditions along with the consequences for society. Such challenges can only be addressed by a multi-disciplinary approach, in which socio-economic, geospatial, and environmental information becomes inter-connected. However, existing solutions cannot be seamlessly integrated and current interaction paradigms prevent mainstream usage of the existing technology. In particular, it is still difficult to access and join harmonized data and processing algorithms that are provided by different environmental information infrastructures. In this paper we take a novel approach for integrated environmental modelling based on the notion of inter-linked resources on the Web. We present design practices for creating resource-oriented interfaces, driven by an interaction protocol built on the combination of valid linkages to enhance resource integration, accompanied by associated recommendations for implementation. The suggested resource-oriented approach provides a solution to the problems identified above, but still requires intense prototyping and experimentation. We discuss the central open issues and present a roadmap for future research. [Copyright &y& Elsevier]

This paper proposes a simple, fast, and accurate method for calibrating the free-space antenna factors of broadband electromagnetic compatibility (EMC) antennas. This method adopts a fixed-height calibration configuration and a multiple-signal-classification-based (MUSIC) data processing algorithm. Because this configuration does not require height scans, it significantly shortens measurement time and removes height-dependent calibration errors. In addition, the MUSIC-based data processing algorithm can remove unexpected reflections from the ground plane or any other reflecting objects, and transfer calibration into the free-space transmission model, from which the free-space antenna factors are obtained. The algorithm of this method can also automatically compensate for the phase center shift, which makes measurement easier and more convenient. To verify this method, numerical simulations for half-wave dipoles and experimental measurements for log-periodic dipole array (LPDA) and horn antennas are conducted. The simulated results for the half-wave dipoles indicate that the error from the theoretical \AFFS can be within ±0.5 dB. In the case of the experimental measurements, the calibrated results are compared with other published standard methods: the mean differences can be as low as 0.25 dB for the LPDA, and as low as 0.36 dB for the horn antennas. The good agreement between results confirms that the proposed method can yield accurate estimations of free-space antenna factors for broadband EMC antennas. This method also provides a new alternative for obtaining fast and accurate antenna calibration. [ABSTRACT FROM PUBLISHER]

This paper presents a modified decorrelated delay lock loop (MD-DLL) for synchronous direct-sequence code-division multiple-access (DS-CDMA) systems under multiuser environments. The structure of the MD-DLL is similar to that of a traditional delay lock loop (T-DLL), aside from the fact that the despreading signals in T-DLL are replaced by a linear combination of all the users' code sequences with various weightings and timing offsets. The basic principles of MD-DLL are given as follows: 1) to use the decorrelating technique of a multiuser detector to remove the multiple-access interference at the on-time code position instant and 2) to choose a proper timing offset for the code sequences to reduce the weighting imbalance between the two arms of the tracking loop and suppress noise enhancement. The proposed MD-DLL scheme solves the tracking bias problems in a decorrelated delay lock loop under a low-signal-to-noise-ratio (SNR) condition. Computer simulations based on Gold code sequences are used to verify the findings. It is shown that MD-DLL achieves a better and stable S-curve with negligible tracking bias, better rms tracking error and mean-time-to-lose-lock performance under high-SNR conditions, and stronger near-far resistance than T-DLL. [ABSTRACT FROM AUTHOR]

As a loudspeaker's physical size decreases, its ability to reproduce low frequencies is hindered. In this paper we propose a method of creating virtual bass using an efficient frequency tracking algorithm. The proposed algorithm has low computational complexity but excellent performance. The results of computer simulations and subjective tests are presented to validate the algorithm. [ABSTRACT FROM AUTHOR]

In this paper, a signal-filtering method based on empirical mode decomposition is proposed. The filtering method is a fully data-driven approach. A noisy signal is adaptively decomposed into intrinsic oscillatory components called intrinsic mode functions (IMFs) by means of an algorithm referred to as a sifting process. The basic principle of the method is to make use of partial reconstructions of the signal, with the relevant IMFs corresponding to the most important structures of the signal (low-frequency components). A criterion is proposed to determine the IMF, after which, the energy distribution of the important structures of the signal overcomes that of the noise and that of the high-frequency components of the signal. The method is illustrated on simulated and real data, and the results are compared to well-known filtering methods. The study is limited to signals that were corrupted by additive white Gaussian noise and is conducted on the basis of extended numerical experiments. [ABSTRACT FROM AUTHOR]

This paper investigates quantization methods for feeding back the channel information through a low-rate feedback channel in the context of multiple-input single-output (MISO) systems. We propose a new quantizer design criterion for capacity maximization and develop the corresponding iterative vector quantization (VQ) design algorithm. The criterion is based on maximizing the mean-squared weighted inner product (MSwIP) between the optimum and the quantized beamforming vector. The performance of systems with quantized beamforming is analyzed for the independent fading case. This requires finding the density of the squared inner product between the optimum and the quantized beamforming vector, which is obtained by considering a simple approximation of the quantization cell. The approximate density function is used to lower-bound the capacity loss due to quantization, the outage probability, and the bit error probability. The resulting expressions provide insight into the dependence of the performance of transmit beamforming MISO systems on the number of transmit antennas and feedback rate. Computer simulations support the analytical results and indicate that the lower bounds are quite tight. [ABSTRACT FROM AUTHOR]

In this paper, a new numerical algorithm for medium-voltage overhead lines, autoreclosure, is described. The subfunction of the autoreclosure scheme that would inhibit the first shot after detecting a solid fault (as compared with an arc fault) is evaluated and presented. It is based on one terminal data processing and it is derived in the time domain. In the algorithm the fault nature (arcing or arcless fault) is estimated using linear least error squares estimation technique. The arc, occurring on the fault point during arcing faults on overhead lines, is included in the problem consideration. In addition, by introducing the prefault load current in the existing model, better algorithm performance and a more reliable adaptive algorithm for autoreclosure are achieved. The algorithm is derived for the case of three-phase symmetrical fault. The results of the algorithm testing through computer simulation are presented. Particularly the algorithm sensitivity to arc elongation effects, supplying network parameters, and processing of the signals in the presence of harmonics are tested and analyzed. [ABSTRACT FROM AUTHOR]

Path selection under multiple additive QoS constraints in high-speed networks is an NP-complete problem. Most of the algorithms proposed for QoS routing suffer from either excessive computational complexities or low performance. In this paper, we propose two practical heuristic algorithms that are based on iterative shortest-path computations with a dynamic cost function. The cost function comprises a combination of several QoS constraints and a single optimization metric. A simulation program evaluates the performance of the algorithms. The simulation results indicate that our algorithms give very high performance and fast running time. [ABSTRACT FROM AUTHOR]

An important problem of the adaptive array antenna proposed previously in regard to its installation is the increased equipment size. This is because the adaptive array antenna requires as many RF circuits as the number of antenna elements. In this paper, in the case where there is an input in one system, we study an adaptive array antenna with sequential processing (AAA-SP) in which the MMSE (minimum mean square error) criterion is used to control array weights by means of sequential processing of sampled signal data. Since there is only one input system in this case, it is found that the SINR is degraded due to reduction of correlation among elements if the conventional MMSE criterion is used as the adaptive control algorithm. Hence, in order to improve the deterioration of the SINR, a tapped-delay-line adaptive array antenna with sequential processing (TDL-AAA-SP) is proposed in which oversampling and a TDL-AAA are combined. It is theoretically proven that the TDL-AAA-SP improves the SINR by recovering from reduction of correlation by means of correlation on the time axis. A computer simulation demonstrates that the proposed scheme is effective. © 2003 Wiley Periodicals, Inc. Electron Comm Jpn Pt 1, 87(2): 59–67, 2004; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecja.10142 [ABSTRACT FROM AUTHOR]

In this paper, we present a novel postprocessing technique based on the theory of the projection onto convex sets (POCS) in order to reduce the blocking artifacts in digital high definition television (HDTV) images. By detecting and eliminating the undesired high-frequency components, mainly caused by blocking artifacts, we propose a new smoothness constraint set (SCS) and its projection operator in the DCT domain. In addition, we propose an improved quantization constraint set (QCS) using the correlation of DCT coefficients between adjacent blocks. In the proposed technique, the range of the QCS is efficiently reduced as close to the original DCT coefficient as possible to yield better performance of the projection onto the QCS. Computer simulation results indicate that the proposed schemes perform better than conventional algorithms. Furthermore, we introduce a fast implementation method of the proposed algorithm. The conventional POCS-based postprocessing techniques require the forward/inverse discrete cosine transform (DCT/IDCT) operations with a heavy computational burden. To reduce the computational complexity we introduce a fast implementation method of the proposed algorithm that does not perform DCT/IDCT operations. Estimates of computation savings vary between 41% and 64% depending on the task. [ABSTRACT FROM AUTHOR]

In this paper, a new improved recursive Newton-type algorithm suitable for various measurement applications in electric power systems is presented. It is used for the power system frequency and spectra estimation. The recursive algorithm form is improved with a strategy of sequential tuning of the forgetting factor. By this, the algorithm convergence and accuracy are significantly improved. To show the main algorithm features, the results of computer simulation, laboratory testing (load rejection and unsuccessful synchronization tests) and field data processing are given. [ABSTRACT FROM AUTHOR]

Soft decision decoding is a decoding method which can cut the decoding error probability as compared with hard decision decoding by utilizing the channel measurement information effectively. In this paper, a new soft decision decoding method by which decoding operations can be efficiently carried out by utilizing the hierarchy of linear block codes is proposed. In addition, an improved algorithm which can reduce significantly the maximum computational complexity is also proposed. The proposed decoding algorithm utilizing the hierarchical structure of codes is a decoding method by which decoding is carried out sequentially from codes with high coding rates or codes of higher hierarchy in the inclusion order, and soft decision decoding can be efficiently carried out by utilizing a number of classes of codes having inclusion relationships in a manner which is superior to Chase algorithm 2 in the decoding error probability and the computational complexity for decoding. Computer simulation results confirm that the proposed algorithm realizes decoding error probability characteristics close to those of the maximum likelihood decoding method and that decoding can be carried out efficiently at a lower computational complexity than Chase algorithm 2 when the SNR is large. In addition, the improved algorithm which can reduce significantly the maximum computational complexity can realize decoding error probability characteristics close to those of the maximum likelihood decoding method on practical communication channels with relatively large SNR. © 1999 Scripta Technica, Electron Comm Jpn Pt 3, 83(3): 108–114, 2000 [ABSTRACT FROM AUTHOR]

Recently, differentially encoded 16 amplitude/phase shift keying (16 DAPSK) has been investigated intensively for mobile radio applications. Since the bit error rate (BER) performance of differential detection (DD) is inferior to that of coherent detection (CD), multisymbol DD of 16 DAPSK has been proposed. However, as the observation interval of multisymbol DD is increased to improve the BER performance, computational complexity increases. In this paper, the differential detection scheme of 16 DAPSK based on the maximum likelihood sequence estimation (MLSE) using Viterbi algorithm is proposed and the BER performance of the proposed scheme is investigated by computer simulation. It is shown that the BER performance of Viterbi-decoding DD outperforms that of three-symbol DD without increasing the computational complexity. [ABSTRACT FROM AUTHOR]

The real-time processing of the recursive least-square algorithm is difficult to execute by the conventional Neumann-type processor because of its large computational complexity. On the other hand, it is known that the recursive least-square algorithm based on the UD decomposition, which is equivalent to the recursive least-square algorithm, can be realized by the systolic array proposed by Rung. The systolic array to execute this algorithm is difficult to realize since it requires a tremendous number of elements and corrections. In the construction of the dedicated hardware, in general, the word length of the processor affects the processing speed and the hardware area. From such a viewpoint, it is important in the construction of the systolic array to execute the recursive least-square algorithm based on the UD decomposition that the word length of the processor should be minimized. To evaluate the word length, an error analysis is required for the finite word length operation of the recursive least-square algorithm based on the UD decomposition. This paper presents the finite word length floating-point error analysis for the recursive least-square algorithm, based on the UD decomposition, and evaluates the operation error. Then the convergence of the algorithm and the number of updates of the algorithm are evaluated analytically. Finally, by a computer simulation, the validity of the theoretical analysis for the convergence and the number of updates is verified. [ABSTRACT FROM AUTHOR]