We present a new multi-path routing methodology, MLB-routing, that is based on the multinomial logit model, which is well known in the random utility field. The key concept of the study is to set multiple paths from the origin to the destination, and distribute packets in accordance with multinomial logit type probability. Since MLB-routing is pure multi-path routing, it reduces the convergence on some links and increases bandwidth utilization in the network. Unlike existing multi-path routing schemes, which pre-set alternate paths, the proposed method can dynamically distribute packets to every possible path and thus is more efficient. Furthermore, it should be mentioned that this methodology can be implemented as either a link-state protocol or a distance-vector protocol. Therefore, it well supports the existing Internet. Simulations show that this methodology raises network utilization and significantly reduces end-to-end delay and jitter.

Antipodal Fermi antenna (APFA) that uses an antipodal feeding section is proposed and its fundamental characteristics are presented. It is shown that the cross polarization level is decreased by 5–10 dB by the presence of the corrugation. It is also found that high gain, low VSWR and low side lobes and low back lobes are obtained. The mechanism of operation principles is discussed by using FDTD analysis. It is found that the corrugation transforms the current of parallel line mode to the current of traveling wave radiation mode and the effective aperture is enlarged which yields high gain characteristics.

This paper proposes a built-in self-test (BIST) scheme for noise-tolerant testing of a digital-to-analogue converter (DAC). The proposed BIST calculates the differences in output voltages between a DAC and test modules. These differences are used as the inputs of an integrator that determines integral nonlinearity (INL). The proposed method has an advantage of random noise cancelation and achieves a higher test accuracy than do the conventional BIST methods. The simulation results show high standard noise-immunity and fault coverage for the proposed method.

UWB (Ultra Wide-Band) pulse radar is promising for surveillance systems because it has an outstanding high range-resolution. To realize an accurate UWB radar imaging system, we propose a new approach that employs multipath echoes from a target in an indoor environment. Using multipath echoes, the proposed system can accurately estimate images, even for targets in a shadow region where the targets are out of sight of the antenna. We apply a simple interferometry technique using the multiple mirror image antennas generated by multipath propagation. We find that this simple method also produces many undesired false image points. To tackle this issue, we also propose an effective false image reduction algorithm to obtain a clear image. Numerical simulations verify that most of the false image points are removed and the target shape is accurately estimated.

The aim of this study is to realize a simplified gait analysis system using wearable sensors. In this paper, a joint angle measurement method using Kalman filter to correct gyroscope signals from accelerometer signals was examined in measurement of hip, knee and ankle joint angles with a wireless wearable sensor system, in which the sensors were attached on the body without exact positioning. The lower limb joint angles of three healthy subjects were measured during gait with the developed sensor system and a 3D motion measurement system in order to evaluate the measurement accuracy. Then, 10 m walking measurement was performed under different walking speeds with a healthy subject in order to find the usefulness of the system as a simplified gait analysis system. The joint angles were measured with reasonable accuracy, and the system showed joint angle changes that were similar to those shown in a previous report as walking speed changed. It would be necessary to examine the influence of sensor attachment position and method for more stable measurement, and also to study other parameters for gait evaluation.

We study a joint optimal design of the receive equalizer and relaying architecture in terms of the weighted sum of mean square errors for the multi-user multiple-input multiple-output (MIMO) relaying system. We show that the optimal relaying architecture depends on the weight factors and that conventional diagonal relaying is a special case of the proposed relaying architecture. Interestingly, for unequal weights, it turns out that the optimal relaying is different from the existing diagonal relaying schemes. By simulation, we show that the proposed joint design improves the weighted sum of mean square error (MSE) over the existing MIMO relaying schemes.

Recently, Yu, Gong and Tang found new constructions of binary sequences of period 4N with optimal autocorrelation magnitude by different interleaved structure of sequences and sequences which had special correlation property, respectively. In this paper, we derive more results on binary sequences of period 4N which also have optimal autocorrelation.

In this article, we first review key derivation functions specified in NIST SP 800-108 and one proposed by Krawczyk. Then, we propose parallelizable key derivation functions obtained by modifying or using the existing schemes. We also define two measures of efficiency of key derivation functions, and evaluate their performance in terms of the two measures.

We propose a system that, using video information, segments the mouth region from a face image and then detects the protrusion of the tongue from inside the oral cavity. Initially, under the assumption that the mouth is closed, we detect both mouth corners. We use a set of specifically oriented Gabor filters for enhancing horizontal features corresponding to the shadow existing between the upper and lower lips. After applying the Hough line detector, the extremes of the line that was found are regarded as the mouth corners. Detection rate for mouth corner localization is 85.33%. These points are then input to a mouth appearance model which fits a mouth contour to the image. By segmenting its bounding box we obtain a mouth template. Next, considering the symmetric nature of the mouth, we divide the template into right and left halves. Thus, our system makes use of three templates. We track the mouth in the following frames using normalized correlation for mouth template matching. Changes happening in the mouth region are directly described by the correlation value, i.e., the appearance of the tongue in the surface of the mouth will cause a decrease in the correlation coefficient through time. These coefficients are used for detecting the tongue protrusion. The right and left tongue protrusion positions will be detected by analyzing similarity changes between the right and left half-mouth templates and the currently tracked ones. Detection rates under the default parameters of our system are 90.20% for the tongue protrusion regardless of the position, and 84.78% for the right and left tongue protrusion positions. Our results demonstrate the feasibility of real-time tongue protrusion detection in vision-based systems and motivates further investigating the usage of this new modality in human-computer communication.

Numerous noise suppression methods for speech signals have been developed up to now. In this paper, a new method to suppress noise in speech signals is proposed, which requires a single microphone only and doesn't need any priori-information on both noise spectrum and pitch. It works in the presence of noise with high amplitude and unknown direction of arrival. More specifically, an adaptive noise suppression algorithm applicable to real-life speech recognition is proposed without assuming the Gaussian white noise, which performs effectively even though the noise statistics and the fluctuation form of speech signal are unknown. The effectiveness of the proposed method is confirmed by applying it to real speech signals contaminated by noises.

The applicability in harsh optical environments, such as dark smog, or strong backlight of ultra-wide band (UWB) pulse radar has a definite advantage over optical ranging techniques. We have already proposed the extended Synthetic Aperture Radar (SAR) algorithm employing double scattered waves, which aimed at enhancing the reconstructible region of the target boundary including shadow region. However, it still suffers from the shadow area for the target that has a sharp inclination or deep concave boundary, because it assumes a mono-static model, whose real aperture size is, in general, small. To resolve this issue, this study proposes an extension algorithm of the double scattered SAR based on a multi-static configuration. While this extension is quite simple, the effectiveness of the proposed method is nontrivial with regard to the expansion of the imaging range. The results from numerical simulations verify that our method significantly enhances the visible range of the target surfaces without a priori knowledge of the target shapes or any preliminary observation of its surroundings.

In this paper, we present an approach of detecting speech presence for which the decision rule is based on a combination of multiple features using a sigmoid function. A minimum classification error (MCE) training is used to update the weights adjustment for the combination. The features, consisting of three parameters: the ratio of ZCR, the spectral energy, and spectral entropy, are combined linearly with weights derived from the sub-band domain. First, the Bark-scale wavelet decomposition (BSWD) is used to split the input speech into 24 critical sub-bands. Next, the feature parameters are derived from the selected frequency sub-band to form robust voice feature parameters. In order to discard the seriously corrupted frequency sub-band, a strategy of adaptive frequency sub-band extraction (AFSE) dependant on the sub-band SNR is then applied to only the frequency sub-band used. Finally, these three feature parameters, which only consider the useful sub-band, are combined through a sigmoid type function incorporating optimal weights based on MSE training to detect either a speech present frame or a speech absent frame. Experimental results show that the performance of the proposed algorithm is superior to the standard methods such as G.729B and AMR2.

In this paper, we study the problem of efficient processing of conjunctive queries in Peer-to-Peer systems based on Distributed Hash Tables (P2P DHT, for short). The basic idea of our approach is to cache the search result for the queries submitted in the past, and to use them to improve the performance of succeeding query processing. More concretely, we propose to adopt Bloom filters as a concrete implementation of such a result cache rather than a list of items used in many conventional schemes. By taking such an approach, the cache size for each conjunctive query becomes as small as the size of each file index. The performance of the proposed scheme is evaluated by simulation. The result of simulation indicates that the proposed scheme is particularly effective when the size of available memory in each peer is bounded by a small value, and when the number of peers is 100, it reduces the amount of data transmissions of previous schemes by 75%.

The past decade has seen a surge of research activities in the fields of mobile computing and wireless communication. In particular, recent technological advances have made portable devices, such as PDA, laptops, and wireless modems to be very compact and affordable. To effectively operate portable devices, energy efficiency and Quality of Service (QoS) provisioning are two primary concerns. Dynamic Voltage Scaling (DVS) is a common method for energy conservation for portable devices. However, due to the amount of data that needs to be dynamically handled in varying time periods, it is difficult to apply conventional DVS techniques to QoS sensitive multimedia applications. In this paper, a new adaptive DVS algorithm is proposed for QoS assurance and energy efficiency. Based on the repeated learning model, the proposed algorithm dynamically schedules multimedia service requests to strike the appropriate performance balance between contradictory requirements. Experimental results clearly indicate the performance of the proposed algorithm over that of existing schemes.

This paper presents a new low-dropout (LDO) regulator with low-quiescent, high-drive and fast-transient performance. This is based on a new composite power transistor composed of a shunt feedback class-AB embedded gain stage and the application of dynamic-biasing schemes to both the error amplifier as well as the composite power transistor. The proposed LDO regulator has been simulated and validated using BSIM3 models and GLOBALFOUNDRIES 0.18-µm CMOS process. The simulation results have shown that the LDO regulator consumes 4.7 µA quiescent current at no load, regulating the output at 1 V from a minimum 1.2 V supply. It is able to deliver up to 450 mA load current with a dropout of 200 mV. It can be stabilized using a 4.7 µF output capacitor with a 0.1 Ω ESR resistor. The maximum transient output voltage is 64.6 mV on the basis of a load step change of 450 mA/10 ns under typical condition. The full load transient response is less than 350 ns.

This paper presents a technique to enhance in-band mismatch noise reduction of multi-stage second order Dynamic Element Matching (DEM) in multi-level ΔΣ Digital-to-Analog Converters (DACs). The presented technique changes an operational behavior of multi-stage DEM to reduce mismatch noise at in-band frequency. This change improves mismatch noise shaping performance for small amplitude input signals. Simulation result using 2-stage second order DEM and a third order 17-level ΔΣ modulator with 0.5% analog element mismatch shows 3.4 dB dynamic range improvement.

This paper proposes a resource allocation method based on TCP (Transmission Control Protocol) throughput for base station diversity systems. A goal of this study is to achieve high throughput wireless Internet access by utilizing multiple wireless links effectively. The conventional work showed that base station diversity techniques can improve TCP performance. However, the improvement depends on the wireless environment of the wireless terminal. The proposed resource allocation method allocates wireless links to a wireless terminal based on its estimated TCP throughput and current traffic of each base station. Our method can take account of some network protocols such as TCP and UDP (User Datagram Protocol) by measuring the current traffic of each base station. In addition, wireless links are preferentially assigned to the wireless terminal that has the largest performance improvement per wireless link. Therefore, the proposal provides better overall system performance than the previous technique.

In this paper a third-order inter-modulation cancellation technique using Pre-Post-Distortion is proposed to design a wideband high linear low-power LNA in deep submicron. The IM3 cancellation is achieved by post-distorting signal inversely after it is pre- distorted in the input trans-conductance stage during amplification process. The operating frequency range of the LNA is 800 MHz–5 GHz. The proposed technique increases input-referred third-order intercept point (IIP3) and input 1 dB Compression point (P-1 dB) to 12–25 dBm and -1.18 dBm, respectively. Post layout simulation results show a noise figure (NF) of 4.1–4.5 dB, gain of 13.7–13.9 dB and S11 lower than -13 dB while consumes 8 mA from 1.2 V supply. The LNA is designed in a 65 nm standard CMOS technology. The layout schematic shows that the LNA occupies 0.150.11 mm2 of silicon area.

A low-complexity Reed-Solomon (RS) decoder design based on the modified Euclidean (ME) algorithm proposed by Truong is presented in this paper. Low complexity is achieved by reformulating Truong's ME algorithm using the proposed polynomial manipulation scheme so that a more compact polynomial representation can be derived. Together with the developed folding scheme and simplified boundary cell, the resulting design effectively reduces the hardware complexity while meeting the throughput requirements of optical communication systems. Experimental results demonstrate that the developed RS(255, 239) decoder, implemented in the TSMC 0.18 µm process, can operate at up to 425 MHz and achieve a throughput rate of 3.4 Gbps with a total gate count of 11,759. Compared to related works, the proposed decoder has the lowest area requirement and the smallest area-time complexity.

The VoIP-based Internet Phonesystem is now seen as one of the killer applications in the high speed and broadband internet environment. Given the wide-spread use of the Internet Phone, it is necessary to provide security services for guaranteeing users' privacy. However, providing security service in Internet Phone has the possibility of incurring additional overheads such as call setup delay time. In this paper, we present a one-way key agreement model based on VoIP in order to reduce call setup time as well as protecting user privacy. The proposed approach decreases the delay time of the call setup in comparison with the previous models because our model enables the key generation in caller side without waiting the response from the receiver.