A simple non-destructive depth estimation method for a crack on a metal surface has been proposed. This method is based on our finding that the electromagnetic back scattering from a narrow trough (crack model) on the ground plane causes periodical nulls (dips) as the frequency changes, and the first dip occurs when the depth of the crack becomes nearly one half of the incident wavelength. Dependencies of the crack's aperture and the incident angle have also been studied from rigorous and numerical analyses, and considered as our depth estimation parameters. A simple estimation formula for a crack depth has been derived from these studies. Test measurement has been made to check the accuracy of our estimation formula. Time domain gating process is utilized for isolating the crack scattering spectra buried in the measured frequency RCS data. Tested crack types are a narrow rectangular, a tapered, and a stair approximated crack shapes. It is found that the depth of these cracks can be measured within 3 percent error by our estimation method.

We previously proposed a change-of-measure based performance measurement method which combines active and passive measurement to estimate performance experienced by user packets and applied this to estimate packet delay. In this paper, we apply it to estimating loss rate. Since packets are rarely lost in current networks, rate measurement usually requires a huge number of probe packets, which imposes a non-negligible load on networks. We propose a loss-rate estimation method which requires significantly fewer number of probe packets. In our proposed method, the correlation between delay and loss is measured in advance, and at the time of measurement, the time-averaged loss rate is estimated by using the delay of probe packets and the correlation. We also applied our change-of-measure framework to estimating the loss rate in user packets by using this time-averaged loss rate. We prove that the mean square error in our method is lower than that simple loss measurement, which is estimated by dividing the number of lost packets by the total number of sent packets. We evaluated our method through simulations and actual measurements and found that it can estimate below 10-3 packet loss rate with only 900 probe packets.

The channel noise in OFDM systems affects the accuracy of channel estimation, deteriorating the performance of equalization. We present a novel algorithm with MMSE (Minimum Mean Square Error) channel estimation based on Hadamard Transform, to mitigate the effects of noise. The performance of the proposed algorithm is proved to be better than that with LS (Least Square) estimation, and very close to that with the MMSE estimation based on Fourier Transform, while the computation required is pretty small due to the use of Hadamard Transform.

This paper presents the performance modeling application of SIP-T (Session Initiation Protocol for Telephones) signaling system based on two-class priority queueing process in carrier class VoIP (Voice over IP) network. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of existing PSTN (Public Switched Telephone Network) with carrier class VoIP network. One of the greatest challenges in the migration from PSTN toward NGN (Next Generation Networks) is to build a carrier class VoIP network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new VoIP technology. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP calls in carrier class VoIP network. This paper presents the two class priority queueing model, performance analysis, and simulation of SIP-T signaling system in carrier class VoIP network focused on toll by-pass or tandem by-pass of PSTN. In this paper, we analyze the average queueing length, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class VoIP network. A mathematical model of the M/G/1 queue with two-class non-preemptive priority assignment is proposed to represent SIP-T signaling system. Then, the formulae of average queueing length, queueing delay, and delay variation for the non-preemptive priority queue are expressed respectively. Several significant numerical examples of average queueing length, queueing delay, and delay variation are presented as well. Finally, the two-class priority queueing model and performance analysis of SIP-T signaling system are shown the accuracy and robustness after the comparison between theoretical estimates and simulation results.

The propagator method (PM) belongs to a class of subspace based methods for direction-of-arrival estimation which only requires linear operations but does not involve any eigendecomposition or singular value decomposition as in common subspace techniques. In this paper, we apply the PM for estimating the frequencies of multiple real sinusoids in noise and a computationally simple as well as high resolution multiple frequency estimation algorithm is developed. The estimation accuracy of the proposed method is contrasted with the conventional MUSIC and Cramer-Rao lower bound under different noise conditions.

Recently six conjectures on linear complexities (LC) of some Kronecker sequences of two or three component sequences are proposed by Karkkainen. In, the LC of Kronecker sequences of two component sequences were studied by Uehara and Imamura, their results are true except in the case when eb 2 or when ea = eb = 1. In this paper the LC for Kronecker sequences of two component sequences are determined completely, and it is shown that all the six conjectures are true except in some special cases, which are listed and corrected.

One-dimentional equivalent circuit model of a heterostructure InAlAs/InGaAs/InP metal-semiconductor-metal photodetector is discussed. In this photodetector, InGaAs is used as an optical absorption layer and the InAlAs is used for Schottky barrier enhanement. The measured S11 parameter deviates from equi-resistance lines on the Smith chart, indicating the equivalent circuit is different from the conventional equivalent circuit using a series resistance, a depletion region capacitance and a depletion region resistance. The difference is due to band discontinuity at the heterojunctions, and we propose a equivalent circuit taking account of the band discontinuity. The band discontinuity is expressed by parallel combination of a resistance and a capacitance. The measured S11 parameter can be fitted well with the calculated S11 parameter from the proposed equivalent circuit, and we can successfully extract the device parameters from the fitted curve.

In this paper, we propose a new predictive location management strategy that reduces the update cost while restricting the paging load optimized for mobiles roaming with traceable patterns. Enhanced with directional predictive capabilities offered by Kalman filtering, new update boundaries are assigned to better reflect the movement patterns of individual mobiles upon location registration. Thus, while complying with the required delay constraints, QoS measures such as throughput will not need to be sacrificed as a result of increasing update threshold. The contribution of this paper is two-fold: (1) to propose a distribution model that is capable of describing a wide range of movement patterns with varying correlation between traveling directions and (2) to show the capabilities (in terms of reliable performances) of the Kalman filter in predicting future movement patterns. Simulation results have successfully demonstrated the ability of the Kalman filter in assigning update boundaries capable of reflecting a mobile's roaming characteristics. The performance gains achieved mainly through a significant reduction in the number of updates indicate its potential for promoting better bandwidth conservation.

RF diamond FETs have been realized on a hydrogen-terminated diamond surface conductive layer. By utilizing the self-aligned gate fabrication process which is effective for the reduction of the parasitic resistance, the transconductance of diamond FETs has been greatly improved. Consequently, the high frequency operation of 22 GHz has been realized in 0.2 µ m gate diamond MISFETs with a CaF2 gate insulator. This value is the highest in diamond FETs and is comparable to the maximum value of SiC MESFETs at present.

A TlGaAs/GaAs multiple quantum-well (MQW) structure having four identical well layers was grown on a GaAs (001) substrate by low-temperature molecular-beam epitaxy (MBE) at 190. The (004) X-ray diffraction (XRD) curve of this sample showed satellite peaks up to the 3rd order at least. The measured XRD curve agreed well with the theoretically simulated one with a Tl content of x=7% and a width of 57 for the TlxGa1-xAs well layers. This result indicates that the grown MQW structure has good single-crystalline quality as well as flat and sharp interfaces.

An approach to minimum mean-squared error (MMSE) decoding for vector quantization over channels with memory is presented. The decoder is based on the Gilbert channel model that allows the exploitation of both intra- and inter-block correlation of bit error sequences. We also develop a recursive algorithm for computing the a posteriori probability of a transmitted index sequence, and illustrate its performance in quantization of Gauss-Markov sources under noisy channel conditions.

Karhunen-Loeve (KL) transform is optimal for many signal detection, communication and filtering applications. An explicit solution of the KL integral equation for a practical case when the covariance function of a stationary process is exponentially oscillating is proposed.

In this paper, a simple adaptive notch filter (ANF) scheme for reducing RFI over CAP/QAM-based VDSL systems is proposed. To alleviate the spectral null caused by notch filtering, a null reshaping scheme is introduced between the normal ANF and the decision feedback equalizer (DFE). The proposed filter scheme can control the width and depth of the null. The shallow and narrow null obtained by null reshaping reduces the loss of signal components and consequently improves the mean square error (MSE) at the output of the equalizer. The proposed null reshaping scheme also enables the infinite impulse response (IIR) type constrained ANF to have a smaller pole contraction factor α. This results in a fast convergence property in RFI frequency estimation with a recursive prediction error (RPE) algorithm. The performance variations of the proposed null reshaping are investigated with varying filter parameters. Compared to the conventional ANF, simulation results show that, at the expense of small system complexity, the proposed structure yields a 2-3 dB MSE gain and a fast convergence property for RFI estimation.

We propose multibits/sequence-period optical code division multiple access (MS-OCDMA) systems with double optical hardlimiters (DHL) in the presence of APD noise, thermal noise, and channel interference. We apply Reed-Solomon (RS) codes to MS-OCDMA to further improve the error rate performance. We show that the MS-OCDMA receiver with DHL improves the bit error probability of MS-OCDMA systems when the received laser power is large. We also show that the performance of RS coded MS-OCDMA system is better than that of on-off keying OCDMA (OOK-OCDMA) system at the same bit rate and at the same chip rate, respectively.

DC and RF performances of AlGaN/GaN HEMTs are simulated using a two-dimensional device simulator with the material parameters of GaN and AlGaN. The cut-off frequency is estimated as 205 GHz at the gate length of 0.05 µm and the drain breakdown voltage at this gate length is over 10 V. The values are satisfactory for millimeter wavelength power applications. The use of thin AlGaN layers has key importance to alleviate gate parasitic capacitance effects at this gate length.

Recently, there has been increasing interest in providing high quality and efficient broadband services over wireless and mobile links. Space-time code is designed to exploit multiple-input multiple-output antenna systems and by doing so an enormous increase in the capacity of wireless systems can be achieved. In this letter, a synchronization technique is proposed to improve the performance of multiple-input multiple-output system. The interpolation method is employed to estimate the coarse and fine frequency offset at the same time without additional complexity.

This paper proposes the conditional LZ complexity and analyzes its property. Especially, we show an inequality corresponding to Ziv's inequality concerning a distinct parsing of a pair of sequences. Further, as a byproduct of the result, we show a simple proof of the asymptotical optimality of Ziv's universal source coding algorithm with side information.

Web Services (WS) have received significant attention recently. Delivering Quality of Service (QoS) on the Internet is a critical and significant challenge for WS community. This article proposes a Web Services Testing Framework (WSTF) for WS participates to perform WS testing. WSTF provides three main distributed components: test master, test agents and test monitor. Test master manages scenarios and generates test scripts. It initiates WS testing by sending test scripts to test agents. Test agents dynamically bind and invoke the WS. Test monitors capture synchronous/asynchronous messages sent and received, attach timestamp, and trace state change information. The benefit to use WSTF is that the user only needs to specify system scenarios based on the system requirements without needing to write test code. To validate the proposed approach, this paper used the framework to test a supply-chain system implemented using WS.

Given the importance of the traffic on modern communication networks, advanced error correction methods are needed to overcome the changes expected in channel quality. Conventional countermeasures that use high dimensionality parity codes often fail to provide sufficient error correction capability. We propose a parity code with high dimensionality that is iteratively decoded. It provides better error correcting capability than conventional decoding methods. The proposal uses the steepest descent method to increase code bit reliability and the coherency between parities and code bits gradually. Furthermore, the quantization of the decoding algorithm is discussed. It is found that decoding with quantization can keep the error correcting capability high.

Nishiara and Morita defined an i.i.d. word-valued source which is defined as a pair of an i.i.d. source with a countable alphabet and a function which transforms each symbol into a word over finite alphabet. They showed the asymptotic equipartition property (AEP) of the i.i.d. word-valued source and discussed the relation with source coding algorithm based on a string parsing approach. However, their model is restricted in the i.i.d. case and any universal code for a class of word-valued sources isn't discussed. In this paper, we generalize the i.i.d. word-valued source to the ergodic word-valued source which is defined by an ergodic source with a countable alphabet and a function from each symbol to a word. We show existence of entropy rate of the ergodic word-valued source and its formula. Moreover, we show the recurrence time theorem for the ergodic word-valued source with a finite alphabet. This result clarifies that Ziv-Lempel code (ZL77 code) is universal for the ergodic word-valued source.