An estimation method of source location of undesired electromagnetic wave from electronic devices by using the MUSIC algorithm is proposed. The MUSIC algorithm can estimate the direction of arrival accurately, however, the estimation error is large in the case of short range multiple coherent sources. In order to overcome this problem, a method to improve the estimation accuracy is presented. Experimental results show that the proposed method can reduce the maximum estimation error from 7 cm of the conventional method to 2 cm.

This paper considers the problem of finding two-dimensional (2-D) direction of arrivals (DOAs) for coherent cyclostationary signals using a 2-D array with random position errors. To alleviate the performance degradation due to the coherence between the signals of interest (SOIs) and the random perturbation in 2-D array positions, a matrix reconstruction scheme in conjunction with an iterative algorithm is presented to reconstruct the correlation matrices related to the received array data so that the resulting correlation matrices possess the eigenstructures required for finding 2-D DOAs. Then, using the reconstructed matrices, we create a subspace orthogonal to the subspace spanned by the direction vectors of the SOIs. Therefore, the 2-D DOAs of the SOIs can be estimated based on a subspace-fitting concept and the created subspace. Finally, several simulation examples are presented for illustration and comparison.

When a dependency parser analyzes long sentences with fewer subjects than predicates, it is difficult for it to recognize which predicate governs which subject. To handle such syntactic ambiguity between subjects and predicates, we define an "a subject clause (s-clause)" as a group of words containing several predicates and their common subject. This paper proposes a two-phase method for S-clause segmentation. The first phase reduces the number of candidates of S-clause boundaries, and the second performs S-clause segmentation using decision trees. In experimental evaluation, the S-clause information turned out to be effective for determining the governor of a subject and that of a predicate in dependency parsing. Further syntactic analysis using S-clauses achieved an improvement in precision of 5 percent.

We propose a new call admission control (CAC) scheme for voice calls in cellular mobile communication networks. It is assumed that the rejection of a hand-off call is less desirable than that of a new call, for a hand-off call loss would cause a severe mental pain to a user. We consider the pains of rejecting new and hand-off calls as different costs. The key idea of our CAC is to restrict the admission of new calls in order to minimize the total expected costs per unit time over the long term. An optimal policy is derived from a semi-Markov decision process in which the intervals between successive decision epochs are exponentially distributed. Based on this optimal policy, we calculate the steady state probability for the number of established voice connections in a cell. We then evaluate the probability of blocking new calls and the probability of forced termination of hand-off calls. In the numerical experiments, it is found that the forced termination probability of hand-off calls is reduced significantly by our CAC scheme at the slight expense of the blocking probability of new calls and the channel utilization. Comparison with the static guard channel scheme is made.

In this paper, we propose a new error feedback (EF) structure for 2-D separable-denominator digital filters described by a rational transfer function. In implementing two-dimensional separable-denominator digital filters, the minimum delay elements structures are common. In the proposed structure, the filter feedback-loop corresponding to denominator polynomial is placed at a different location compared to the commonly used structures. The proposed structure can minimize the roundoff noise more than the previous structure though the number of multipliers is less than that of previous one. Finally, we present a numerical example by designing the EF on the proposed structure and demonstrate the effectiveness of the proposed method.

It is well known that the trellises of lattices can be employed to decode efficiently. It was proved in [1] and [2] that if a lattice L has a finite trellis under the coordinate system , then there must exist a basis (b1,b2,,bn) of L such that Wi=span() for 1in. In this letter, we prove this important result in a completely different method, and give an efficient method to compute all bases of this type.

The reduction of the signaling load associated with IP mobility management is one of the significant challenges to IP mobility support protocols. Hierarchical Mobile IPv6 (HMIPv6) aims to reduce the number of the signaling messages in the backbone networks, and improve handoff performance by reducing handoff latency. However, this does not imply any change to the periodic binding update (BU) to the home agent (HA) and the correspondent node (CN), and now a mobile node (MN) additionally should send it to the mobility anchor point (MAP). Moreover, the MAP should tunnel the received packets to be routed to the MN. These facts mean that the reduction of the BU messages in the backbone networks can be achieved at the expense of the increase in the signaling bandwidth consumption within a MAP domain. On the other hand, it is observed that an MN may habitually stay for a relatively long time or spend on using much Internet in a specific cell (hereafter, home cell) covering its home, office or laboratory, etc. Thus, considering the preceding facts and observation, HMIPv6 may not be favorable especially during a home cell residence time in terms of signaling bandwidth consumption. To overcome these drawbacks of HMIPv6, we propose a history-based auxiliary mobility management strategy (H-HMIPv6) to enable an MN to selectively switch its mobility management protocols according to whether it is currently in its home cell or not in HMIPv6 networks. The operation of H-HMIPv6 is almost the same as that of HMIPv6 except either when an MN enters/leaves its home cell or while it stays in its home cell. Once an MN knows using its history that it enters its home cell, it behaves as if it operates in Mobile IPv6 (MIPv6), not in HMIPv6, until it leaves its home cell; No periodic BU messages to the MAP and no packet tunneling occur during the MN's home cell residence time. The numerical results indicate that compared with HMIPv6, H-HMIPv6 has apparent potential to reduce the signaling bandwidth consumption and the MAP blocking probability.

Recently, automatic speech recognition in a car has practical uses for applications like car-navigation and hands-free telephone dialers. For noise robustness, the current successes are based on the assumption that there is only a stationary cruising noise. Therefore, the recognition rate is greatly reduced when there is music or news coming from a radio or a CD player in the car. Since reference signals are available from such in-vehicle units, there is great hope that echo cancellers can eliminate the echo component in the observed noisy signals. However, previous research reported that the performance of an echo canceller is degraded in very noisy conditions. This implies it is desirable to combine the processes of echo cancellation and noise reduction. In this paper, we propose a system that uses echo cancellation and spectral subtraction simultaneously. A stationary noise component for spectral subtraction is estimated through the adaptation of an echo canceller. In our experiments, this system significantly reduced the errors in automatic speech recognition compared with the conventional combination of echo cancellation and spectral subtraction.

When video data are transmitted via the network, the quality of video data must be carefully chosen to be best under the condition that the transmission is not influenced by other internet services. They often use the simulcast type, which uses independent streams that are stored and transmitted for the quality, considering implementation, when they select the video quality. On the other hand, we had already proposed the scalable structure, which consists of base and enhancement data, but when they require the high quality video, these data are combined using the transcoding methods. In this paper, we propose the video contents delivery methods with scalable transcoding, in which users can update the quality of video data even after the transmission by base data and differential data. In order to reduce the total time of not only users' access time, but also watching time, we compare simulcast method with proposed methods in the total content utilization time using a video contents access model, and evaluate required transcoding time to reduce the waiting time of users.

The authors realize a 50% length reduction of short-slot couplers in a post-wall dielectric substrate by two techniques. One is to introduce hollow rectangular holes near the side walls of the coupled region. The difference of phase constant between the TE10 and TE20 propagating modes increases and the required length to realize a desired dividing ratio is reduced. Another is to remove two reflection-suppressing posts in the coupled region. The length of the coupled region is determined to cancel the reflections at both ends of the coupled region. The total length of a 4-way Butler matrix can be reduced to 48% in comparison with the conventional one and the couplers still maintain good dividing characteristics; the dividing ratio of the hybrid is less than 0.1 dB and the isolations of the couplers are more than 20 dB.

With reference to video motion estimation in the framework of the new H.264/AVC video coding standard, this paper presents algorithmic and architectural solutions for the implementation of context-aware coprocessors in real-time, low-power embedded systems. A low-complexity context-aware controller is added to a conventional Full Search (FS) motion estimation engine. While the FS coprocessor is working, the context-aware controller extracts from the intermediate processing results information related to the input signal statistics in order to automatically configure the coprocessor itself in terms of search area size and number of reference frames; thus unnecessary computations and memory accesses can be avoided. The achieved complexity saving factor ranges from 2.2 to 25 depending on the input signal while keeping unaltered performance in terms of motion estimation accuracy. The increased efficiency is exploited both for (i) processing time reduction in case of software implementation on a programmable platform; (ii) power consumption reduction in case of dedicated hardware implementation in CMOS technology.

Based on the Si CMOS process, a low operating voltage and low power light emitting device is presented. It has a power transfer efficiency of 1 to 2 orders higher than previous reports and can be used as a high efficiency photodiode. Configurations using the same structure as both the light emitter and the optical receiver, and employing a simple modulation instrument is then proposed for applications in the chip-to-chip optical alignment and the signal transmission. Only single power supply is required in the emitter-receiver circuits and is compatible with other integrated circuits made by the CMOS process.

An approach based on signal subspace analysis is proposed to blind estimation of the PN (Pseudo Noise) sequence from lower SNR (Signal to Noise Ratios) DS/SS (Direct Sequence Spread Spectrum) signals. The received signal is divided into vectors according to a temporal window, from which an autocorrelation matrix is computed and accumulated. The PN sequence can be reconstructed from principal eigenvectors of the matrix.

In this paper, two techniques are proposed for accelerating and stabilizing the Levenberg-Marquardt (LM) method where its conventional stabilizer matrix (identity matrix) is superseded by (1) a diagonal matrix whose elements are column norms of Jacobian matrix J, or (2) a non-diagonal square root matrix of J TJ. Geometrically, these techniques make constraint conditions of the LM method fitted better to relevant cost function than conventional one. Results of numerical simulations show that proposed techniques are effective when both column norm ratio of J and mutual interactions between arguments of the cost function are large. Especially, the technique (2) introduces a new LM method of damped Gauss-Newton (GN) type which satisfies both properties of global convergence and quadratic convergence by controlling Marquardt factor and can stabilize convergence numerically. Performance of the LMM techniques are compared also with a damped GN method with line search procedure.

This paper presents the design of a novel method for improvement of the operation of distance relays during capacitive voltage transformer transients using artificial neural network. The proposed module uses voltage and current signals to learn the hidden relationship existing in the input patterns. Simulation studies are preformed and the influence of changing system parameters, such as fault resistance and source impedance is studied. Details of the design procedure and the results of performance studies with the proposed relay are given in the paper. Performance studies results show that the proposed algorithm decreases the effects of CVT transients and is fast and accurate.

This paper describes a method for separating a target sound from other noise arriving in a single direction when the target cannot, therefore, be separated by directivity control. Microphones are arranged in a line toward the sources to form null sensitivity points at given distances from the microphones. The null points exclude non-target sound sources on the basis of weighting coefficients for microphone outputs determined by blind source separation. The separation problem is thereby simplified to instantaneous separation by adjustment of the time-delays for microphone outputs. The system uses a direct (i.e. non-iterative) algorithm for blind separation based on second-order statistics, assuming that all sources are non-stationary signals. Simulations show that the 2-microphone system can separate a target sound with separability of more than 40 dB for the 2-source problem, and 25 dB for the 3-source problem when the other sources are adjacent.

In this letter, we consider a problem of global exponential stabilization of a class of approximately feedback linearized systems. With a newly proposed LMI-condition, we propose a controller design method which is shown to be improved over the existing methods in several aspects.

Most branch predictors use the PC information of the branch instruction and its dynamic Global Branch History (GBH). In this letter, we suggest a Branch Direction History (BDH) as the third component of the branch prediction and analyze its impact upon the prediction accuracy. Additionally, we propose a new branch predictor, direction-gshare predictor, which utilizes the BDH combined with the GBH. At first, we model a neural network with (PC, GBH, and BDH) and analyze their actual impact upon the branch prediction accuracy, and then we simulate our new predictor, the direction-gshare predictor. The simulation results show that the aliasing in Pattern History Table (PHT) is significantly reduced by the additional use of BDH information. The direction-gshare predictor outperforms bimodal predictor, two-level adaptive predictor and gshare predictor up to 15.32%, 5.41% and 5.74% respectively, without additional hardware costs.

This paper proposes a novel boundary scan test scheme for intellectual property (IP) core identification via watermarking. The core concept is embedding a watermark identification circuit (WIC) and a test circuit into the IP core at the behavior design level. The procedure depends on current IP-based design flow. This scheme can detect the identification of the IP provider without the need to examine the microphotograph after the chip has been manufactured and packaged. This scheme can successfully survive synthesis, placement, and routing and identify the IP core at various design levels. Experimental results have demonstrated that the proposed approach has the potential to solve the IP identification problem.

Future broadband mobile communication systems are necessary to achieve the bit rates of 100 Mbit/s. Orthogonal Frequency Division Multiplexing (OFDM) transmission is an attractive technology because it can remove the influence of frequency selective fading in broadband transmission by adding a suitable guard interval to each OFDM symbol. However, peak-to-average power ratio (PAPR) is very large in OFDM transmission. In this paper, we propose a new PAPR reduction method which can be applied even when unusable bands are inside the system band. In the proposed method, peak reduction signals are generated by iterative signal processing only in the usable frequency band, and filtering to remove out-of-band components of the peak reduction signals is incorporated into the iterative signal processing. The results of computer simulation show that the proposed method can effectively reduce peak power without expanding the spectrum both outside the system band and into unusable bands inside the system band. By using the proposed method, the broadband mobile communication system with low peak power and high flexibility of frequency band use can be realized.