Face perception and recognition have attracted more attention recently in multidisciplinary fields such as engineering, psychology, neuroscience, etc. with the advances in physical/physiological measurement and data analysis technologies. In this paper, our main interest is building computational models of human face recognition based on psychological experiments. We specially focus on modeling human face recognition characteristics of average face in the dimension of distinctiveness. Psychological experiments were carried out to measure distinctiveness of face images and their results are explained by computer analysis results of the images. Two psychological experiments, 1) Classical experiment of distinctiveness rating and, 2) Novel experiment of recognition of an average face were performed. In the later experiment, we examined on how the average face of two face images was recognized by a human in a similarity test respect to the original images which were utilized for the calculation of the average face. To explain results of the psychological experiments, eigenface spaces were constructed based on Principal Component Analysis (PCA). Significant correlation was found between human and PCA based computer recognition results. Emulation of human recognition of faces is one of the expected applications of this research.

This paper presents a new technique to smooth speech feature vectors for text-independent speaker verification using an adaptive band-pass IIR filer. The filter is designed by considering the probability density of modulation-frequency components of an M-dimensional feature vector. Each dimension of the feature vector is processed and filtered separately. Initial filter parameters, low-cut-off and high-cut-off frequencies, are first determined by the global mean of the probability densities computed from all feature vectors of a given speech utterance. Then, the cut-off frequencies are adapted over time, i.e. every frame vector, in both low-frequency and high-frequency bands based also on the global mean and the standard deviation of feature vectors. The filtered feature vectors are used in a SVM-GMM Supervector speaker verification system. The NIST Speaker Recognition Evaluation 2006 (SRE06) core-test is used in evaluation. Experimental results show that the proposed technique clearly outperforms a baseline system using a conventional RelAtive SpecTrA (RASTA) filter.

Flat Panel Displays (FPDs) are manufactured using many pieces of different processing equipment arranged sequentially in a line. Although the constant inter-arrival time (i.e., the tact time) of glass substrates in the line should be kept as short as possible, the collision probability between glass substrates increases as tact time decreases. Since the glass substrate is expensive and fragile, collisions should be avoided. In this paper, we derive a closed form formula of the approximate collision probability for a model, in which the processing time on each piece of equipment is assumed to follow Erlang distribution. We also compare some numerical results of the closed form and computer simulation results of the collision probability.

In this paper, through extrinsic information transfer (EXIT) band chart analysis, an adaptive iterative decoding approach (AIDA) is proposed to reduce the iterative decoding complexity and delay for finite-length differentially encoded Low-density parity-check (DE-LDPC) coded systems with multiple-symbol differential detection (MSDD). The proposed AIDA can adaptively adjust the observation window size (OWS) of the MSDD soft-input soft-output demodulator (SISOD) and the outer iteration number of the iterative decoder (consisting of the MSDD SISOD and the LDPC decoder) instead of setting fixed values for the two parameters of the considered systems. The performance of AIDA depends on its stopping criterion (SC) which is used to terminate the iterative decoding before reaching the maximum outer iteration number. Many SCs have been proposed; however, these approaches focus on turbo coded systems, and it has been proven that they do not well suit for LDPC coded systems. To solve this problem, a new SC called differential mutual information (DMI) criterion, which can track the convergence status of the iterative decoding, is proposed; it is based on tracking the difference of the output mutual information of the LDPC decoder between two consecutive outer iterations of the considered systems. AIDA using the DMI criterion can adaptively adjust the out iteration number and OWS according to the convergence situation of the iterative decoding. Simulation results show that compared with using the existing SCs, AIDA using the DMI criterion can further reduce the decoding complexity and delay, and its performance is not affected by a change in the LDPC code and transmission channel parameters.

BLOCKSUM, also known as KEISANBLOCK in Japanese, is a Latin square filling type puzzle, such as Sudoku. In this paper, we prove that the decision problem whether a given instance of BLOCKSUM has a solution or not is NP-complete.

We evaluate the rich-club property of the Internet topology at the autonomous system (AS) level by comparing the Internet AS graphs of traceroute and BGP, and the synthetic graphs of PFP model. The results indicate that, for rich-club coefficient, PFP model can exactly match traceroute AS graphs in the early years around 2002, but it has significantly deviated from the grown AS graphs since about 2010.

This paper proposes an alternate time-switched transmission technique for single carrier modulation system with frequency domain equalization. Additional maximal ratio combining diversity gain is obtained by adding a transmit antenna and a switch. Alternating transmit symbols result in zeros which make maximal ratio receive combining possible in the receiver. Simulation results show that it has better performance than the traditional algorithm at the expense of one additional antenna.

With the fast development of mobile communication technologies, mobile multimedia services like mobile Video on Demand (VOD) are becoming prevalent. However, VOD streaming requires dedicated bandwidth to satisfy Quality of Service (QoS), and the limited wireless bandwidth will become insufficient to support the increasing number of mobile VOD users. In order to solve the problem, this paper proposes a Call Admission Control (CAC) approach which can accept new users even when the system bandwidth is insufficient. Our approach also guarantees continuous playback for subscribers by taking into account the service end time and the delay bound of the users. Simulation results demonstrate that the proposed scheme can increase the number of concurrent users and reduce the connection blocking probability significantly without playback interruption.

In our previous work [2], we proposed a new concept of utility functions for rate control in communication networks. Unlike conventional utility-based rate control in which the utility function of each user is defined as a function of its transmitting data rate, in [2], we defined the utility function of each user as a function of not only its transmitting data rate but also it receiving data rate. The former is called a session-level utility function and the latter is called a user-level utility function. The user-level utility function reflects the satisfaction with the service of a user with two-way communication, which consists of transmitting and receiving sessions, better than the session-level utility function, since user's satisfaction depends on not only the satisfaction with its transmitting session but also that for its receiving session. In [2], an algorithm that required each user to know the exact utility function of its correspondent was developed. However, in some cases, this information might not be available due to some reasons such as security and privacy issues, and in such cases, the algorithm developed in [2] cannot be used. Hence, in this paper, we develop a new distributed algorithm that does not require each user to know the utility function of its correspondent. Numerical results show that our new algorithm, which does not require the utility information of the correspondent, converges to the same solution to that with the algorithm that requires the utility information of the correspondent.

We show an improved throughput scaling law for an ultra-wide band (UWB) ad hoc network by using a modified hierarchical cooperation (HC) strategy; the n wireless nodes are assumed to be randomly sited. In a dense network of unit area, our result indicates that the derived throughput scaling depends on the path-loss exponent α for certain operating regimes due to the power-limited characteristics. It also turns out that the use of HC is helpful in improving the throughput scaling of our UWB network in some conditions. More specifically, assuming that the bandwidth scales faster than nα+1(log n)α/2, it is shown that the HC protocol outperforms nearest multi-hop routing for 2 < α < 3 while using nearest multi-hop routing leads to higher throughput for α ≥ 3.

We humans are easily able to instantaneously detect the regions in a visual scene that are most likely to contain something of interest. Exploiting this pre-selection mechanism called visual attention for image and video processing systems would make them more sophisticated and therefore more useful. This paper briefly describes various computational models of human visual attention and their development, as well as related psychophysical findings. In particular, our objective is to carefully distinguish several types of studies related to human visual attention and saliency as a measure of attentiveness, and to provide a taxonomy from several viewpoints such as the main objective, the use of additional cues and mathematical principles. This survey finally discusses possible future directions for research into human visual attention and saliency computation.

It has been reported that the temporal change of current during the deposition shows a plateau and a break, similar to those found in a photocurrent profile taken by the time-of-flight technique for the investigation of photocarrier dynamics in condensed matters, enabling the estimation of electrophoretic mobility of colloidal particles in the suspension. The estimation of the electrophoretic mobility from transient current during the deposition by the simple drift model is based on the assumption that a constant electric field is uniformly applied between the positive and negative electrodes. Therefore, it is important to check if this assumption is satisfied. It is also important to measure the temporal evolution of film thickness, because this may give information about uniformity of colloidal size in the suspension. This study addresses these topics and validity of the assumption is confirmed.

This paper is focused on discussing a low-voltage system for lightning, and in particular the testing equipment of surge arresters. Only by demonstrating the performance and applicability of arresters can we seek the most feasible and economic low-voltage solutions. After performing repeated experiments with the same testing samples, using different testing equipment, we compare the different test results in order to select the most suitable and applicable testing equipment. In addition, the basis of a surge current parameter design theory is confirmed and verified through the test results using a simple and compact Impulse Current Generator to test a wide range of samples. By performing the actual analyzes and experiments, we can understand deeply how R, L, and C affect surge current, current wave, and current wave time. The ideal testing equipment standards have been set as follows: (1) Test Voltage up to 20 kV; (2) Expand current range from 1.5 kA to 46.5 kA, with resolution 1.5 kA; and (3) Simple operational procedures.

Process variation causes significant fluctuations in the timing performance of analog circuits, which causes a fraction of circuits to fail specifications. By testing the delay-performance, we can recognize the failed circuits during production testing. In this paper, we have proposed a low overhead and process tolerant delay evaluation circuit for built-in self test (BIST) function for analog differential circuits. This circuit contains a delay generation cell, an input differential signal generation cell, a delay matching cell, a sample-hold circuit, and a comparator. This circuit was implemented with 0.18 µm CMOS process. Simulation results over process variation, devices mismatch and layout parasitics, but without silicon measurement, show that the accuracy in delay detection is within 5 ps. A case study was done over a feed-forward equalizer (FFE). A typical use of this circuit is testing the delay of various FIR (Finite Impulse Response) filters.

In this letter, unlike the previous work in [2], the optimal power allocation in a non-orthogonal, amplify-and-forward (AF) relay-assisted transmission is investigated in the uplink. Here, the inter-user-interference among the signals from MTs and relays exists due to non-zero interference suppression factor (ISF), i.e., finite spreading factor. In this letter, we show that the optimal solution to achieve a 'max-min fairness' among mobile terminals can be alternatively obtained by solving its inverse problem. The impact of various ISFs as well as the Jain's fairness is investigated in comparison with the equal power allocation.

To support mobility, multihoming, routing scalability, and security, there are a lot of proposals based on ID/Locator split approach not only for the current Internet but also for the future Internet. However, none of them meet the requirements for practical operation such as (1) support heterogeneous network layer protocols, (2) scalability of ID/Locator mapping system, (3) independence of mapping information management, and (4) avoidance of locator leakage beyond the administrative boundary. This paper proposes a network layer protocol called Z Network Protocol (ZNP) for the future Internet based on the clean slate approach. ZNP supports heterogeneity of network layer protocols by “Internetworking with a Common ID Space”. Its mapping systems meet the requirements (1)–(4) described above. For manipulating the mapping systems, Z Control Message Protocol (ZCMP) is designed. For resolving the link layer (L2) address from the ZNP Locator, Z Neighbor Discovery Protocol (ZNDP) is designed. We implement ZNP and ZNDP in the Linux kernel, ZCMP in the user space and measure the times needed for transmission, reception, forwarding, and locator conversion. The results show the practicability of ZNP as a network layer protocol for the future Internet.

Recently, the number of users downloading video content on the Internet has dramatically increased, and it is highly anticipated that downloading huge size, rich content such as movie files will become a popular use of the Internet in the near future. The transmission bandwidth consumed by delivering rich content is enormous, so it is urgent for ISPs to design an efficient delivery system that minimizes the amount of network resources consumed. To deliver web content efficiently, a content delivery network (CDN) is often used. CDN providers collocate a huge number of servers within multiple ISPs without being informed of detailed network information, i.e., network topologies, from ISPs. Minimizing the amount of network resources consumed is difficult because a CDN provider selects a server for each request based on only rough estimates of response time. Therefore, an ordinary CDN is not suited for delivering rich content. P2P-based delivery systems are becoming popular as scalable delivery systems. However, by using a P2P-based system, we still cannot obtain the ideal delivery pattern that is optimal for ISPs because the server locations depend on users behaving selfishly. To provide rich content to users economically and efficiently, an ISP itself should optimally provide servers with huge storage capacities at a limited number of locations within its network. In this paper, we investigate the content deployment method, the content delivery process, and the server allocation method that are desirable for this ISP-operated CDN. Moreover, we evaluate the effectiveness of the ISP-operated CDN using the actual network topologies of commercial ISPs.

This paper presents a joint linear processing scheme for two-hop and half-duplex distributed amplify-and-forward (AF) relaying networks with one source, one destination and multiple relays, each having multiple antennas. By using the minimum mean-square error (MMSE) criterion and the Wiener filter principle, the joint relay and destination design with perfect channel state information (CSI) is first formulated as an optimization problem with respect to the relay precoding matrix under the constraint of a total relay transmit power. The constrained optimization with an objective to design the relay block-diagonal matrix is then simplified to an equivalent problem with scalar optimization variables. Next, it is revealed that the scalar-version optimization is convex when the total relay power or the second-hop SNR (signal to noise ratio) is above a certain threshold. The underlying optimization problem, which is non-convex in general, is solved by complementary geometric programming (CGP). The proposed joint relay and destination design with perfect CSI is also extended for practical systems where only the channel mean and covariance matrix are available, leading to a robust processing scheme. Finally, Monte Carlo simulations are undertaken to demonstrate the superior MSE (mean-square error) and SER (symbol error rate) performances of the proposed scheme over the existing relaying method in the case of relatively large second-hop SNR.

For opportunistic spectrum access (OSA), spectrum management is a key function to effectively utilize white space without causing harmful interference to incumbent receivers. Geo-location database approaches using radio propagation estimation have been regarded as practical spectrum management methods. However, propagation models inevitably fail to accurately estimate the path loss in actual radio environments, resulting in estimation error of carrier to interference ratio (CIR) of the incumbent receivers. This could prevent white space from being efficiently utilized, because the allowable transmit power of the opportunistic system has to be limited to keep the CIR at the required level. To improve the accuracy of CIR estimation, we propose the new concept of Interference Monitoring which works in combination with spectrum management. In this method, a monitoring node located near the incumbent receivers actually measures both the interference signals and the incumbent signals. Using the measurement results, the CIR estimates are corrected based on the minimum mean square error (MMSE) criterion. The proposed Interference Monitoring can be extended to establish cooperation among multiple monitoring nodes and thus spatial diversity. Analytical evaluations assuming a simple cellular system model show that Interference Monitoring can more accurately estimate CIR, and thus it can significantly increase the allowable transmit power. For an urban macro cell, Interference Monitoring with a single node achieved about a 6.5 dB increase in the transmit power; Cooperative Interference Monitoring with 4 nodes achieved about a 13.5 dB increase. Thus, Interference Monitoring-based spectrum management can maximize opportunities for white space utilization without imposing additional interference to the incumbent system.