We report a flexible and lightweight wearable microstrip antenna that can be sewn into clothing and hats. This antenna is composed of felt and a conductive woven fabric. Experimental results clearly show that this antenna operates normally as a conventional microstrip antenna, and is practical and feasible for personal satellite communications.

The characteristics of the spreading sequence significantly affect the signal-to-interference power ratio (SIR) of the received signal in direct sequence code division multiple access (DS-CDMA) system. In this paper, we analyze the receiver performance of the forward link of a DS-CDMA system in terms of the SIR and bit error rate (BER) when pseudo noise (PN) codes and concatenated orthogonal/PN (OPN) codes are used as the spreading sequence. The use of OPN spreading codes can cancel out the intra-cell interference signals with equal path delay, but the use of PN spreading codes cannot, significantly degrading the performance. As a result, the BER performance of the OPN spreading system is better than that of the PN spreading system. The use of OPN spreading sequences can provide the system capacity at least two times larger than the use of PN spreading sequences in the single-cell environment even when the channel has a large number of multipaths. The two spreading systems also show significant difference in the user capacity even in a multi-cell environment.

In this paper, we propose a resolution enhancement method for estimating direction-of-arrival (DOA) of narrowband incoherent signals incident on a general array. The resolution of DOA algorithm is dependent on the aperture size of antenna array. But it is very impractical to increase the physical size of antenna array in real environment. We propose the method that improves resolution performance by virtually expanding the sensor spacing of original antenna array and then averaging the spatial spectrum of each virtual array which has a different aperture size. Superior resolution capabilities achieved with this method are shown by simulation results in comparison with the standard MUSIC for incoherent signals incident on a uniform circular array.

In this paper we propose an MPLS switch model with a GPS-based scheduler to provide guaranteed QoS in the Internet. Important applications of this model include IP VPN (MPLS-VPN) services and MPLS networks. The considered scenarios include cases with sufficient labels or with limited number of labels under competing traffic streams. Due to tractability issues in direct analysis, we use a worst case equivalent system model to obtain the theoretical upper bounds of delays and backlogs of the original system model. Simulation results are used to verify the analytical results. We can obtain the worst case delay bounds and the maximum backlogs. Furthermore, the trade-off among the number of labels, the worst delay bounds, and the maximum backlogs can be observed. With our model, one can determine the number of required labels according to the allowable delay and the available buffer size.

In the optimum design of AC contactors, it is necessary to analyze the dynamic behavior. Moreover, movable contacts and core bounce have remarkable effect on the lifetime of contactors. A set of differential equations describes the coupling of the electric circuit, electromagnetic field and mechanical system taking account into bounce and the influence of friction. With virtual prototyping technology, the dynamic behavior, especially for contacts bounce, has been investigated according to different electrical circuit parameters. Two approaches are introduced to solve electromagnetic parameters. Based on 3D finite element static nonlinear analysis, the flux linkage and electromagnetic force can be evaluated with different air gap and exciting current for larger gap. In addition, concerning to the shading coil for smaller gap, magnetic circuit can facilitate the calculation. The validity of the proposed method is confirmed by experiments.

Montgomery-form elliptic curves have the advantage of faster arithmetic than Weierstrass-form elliptic curves. The dominant operation of the Elliptic Curve Cryptosystem (ECC) is scalar multiplication of points on an elliptic curve, and it usually includes scalar multiplication of a fixed base point of ECC. For Weierstrass-form elliptic curves, accelerating methods of scalar multiplication by using a pre-computed table of the fixed point have been widely studied. However, such methods cannot naturally expand to Montgomery-form elliptic curves. In this paper, we propose a fast scalar multiplication method on Montgomery-form elliptic curves by using a pre-computed table for the first time. Our method is 1.6 times as fast as the known method for Montgomery-form elliptic curves under the practical conditions that the size of the definition field is 160 bits and the memory size used for the pre-computed table is 3.2 KB.

This paper proposes a learning classifier based on Support Vector Domain Description (SVDD) for two-class problem. First, by the description of the training samples from one class, a sphere boundary containing these samples is obtained; then, this boundary is used to classify the test samples. In addition, instead of the traditional quadratic programming, multiplicative updates is used to solve the Lagrange multiplier in optimizing the solution of the sphere boundary. The experiment on CBCL face database illustrates the effectiveness of this learning algorithm in comparison with Support Vector Machine (SVM) and Sequential Minimal Optimization (SMO).

A simple millimeter-wave quasi-maximal-ratio-combin-ing antenna diversity system based on the millimeter-wave self-heterodyne transmission technique is described. The millimeter-wave self-heterodyne transmission technique is useful for developing millimeter-wave systems with enhanced characteristics in regard to system miniaturization, development and fabrication cost, and the frequency stability of the signal transmission. We also show that applying this technique with an antenna diversity receiver configuration can easily solve a problem peculiar to millimeter-wave systems--the fact that the transmission link always requires a line-of-sight path--without requiring hardware designed with millimeter-scale precision. In this paper, we theoretically analyze the operating principle of a combining antenna diversity system based on the millimeter-wave self-heterodyne transmission technique. We further prove that we can obtain a diversity gain in accordance with that of a maximal-ratio combining diversity system without resorting to any complicated control of the received signal envelope and phase. Our experiments using the simplest two-branch diversity structure have validated the operating principle derived in our theoretical analysis. Our results show that a received CNR improvement of 3 dB is obtained as a diversity gain. We also demonstrate that circuit precision corresponding to the wavelength of the intermediate frequency, rather than to the millimeter wavelength, is sufficient to obtain the diversity effect when we control the signal phase or delay in combining the received signals.

In terms of the formulation of the optimality, image restoration filters can be divided into two streams. One is formulated as an optimization problem in which the fidelity of a restored image is indirectly evaluated, and the other is formulated as an optimization problem based on a direct evaluation. Originally, the formulation of the optimality and the solutions derived from the formulation are identical each other. However in many studies adopting the former stream, an arbitrary choice of a solution without a mathematical ground passes unremarked. In this paper, we discuss the relation between the formulation of the optimality and the solution derived from the formulation from a mathematical point of view, and investigate the relation between a direct style formulation and an indirect one. Through these analyses, we show that the both formulations yield the identical filter in practical situations.

The Camera sensor network is a new advent of technology in which each sensor node can capture video signal, process and communicate with other nodes. We have investigated a dense node configuration. The requested processing task in this network is arbitrary view generation among nodes view. To avoid unnecessary communication between nodes in this network and to speed up the processing time, we propose a distributed processing architecture where the number of nodes sharing image data are optimized. Therefore, each sensor node processes part of the interpolation algorithm with local communication between sensor nodes. Two processing methods are used based on the image size shared. These two methods are F-DP (Fully image shared Distributed Processing) and P-DP (Partially image shared Distributed Processing). In this research, the network processing time has been theoretically analyzed for one user. The theoretical results are compatible with the experimental results. In addition, the performance of proposed DP methods were compared with Centralized Processing (CP). As a result, the best processing method for optimum number of nodes can be chosen based on (i) communication delay of the network, (ii) whether the network has one or more channels for communication among nodes and (iii) the processing ability of nodes.

In uplink multicarrier code division multiple access (MC-CDMA), the inter-code interference (ICI) caused by the independent and frequency-selective fading channel of each user and the inter-carrier interference caused by the asynchronous reception of each user's OFDM symbols result in multiple access interference (MAI). This paper evaluates the ICI in frequency-selective Rayleigh fading channels for uplink MC-CDMA. We derive theoretical expressions for the desired-to-undesired signal power ratio (DUR) as a quantitative representation of ICI, and validate them by comparison with computer simulations using a Walsh-Hadamard (WH) code. Based on the analytical results, we obtain the optimum spreading sequence that minimizes the ICI (in short, maximizes the multiplexing performance); this sequence appears to be orthogonal. Three equalization combining methods are examined; equal gain combining (EGC), orthogonality restoring combining (ORC), and maximum ratio combining (MRC).

This paper proposes a new implementation of an adaptive noise canceller based upon a parallel block structure, which aims to raise the processing and convergence rates and to improve the steady-state performance. The procedure is as follows: First, an IIR bandpass filter with a variable center angular frequency using adaptive Q-factor control and two adaptive control signal generators are realized by the parallel block structure. Secondly, a new algorithm for adaptive Q-factor control with parallel block structure is proposed to improve the convergence characteristic. In addition, the steady-state performance of the filter is stabilized by using the variable step size parameter in adaptive control of the center frequency and the speed up of the convergence rate is achieved by adopting a normalized gradient algorithm for adaptive control. Finally, simulation results are given to demonstrate the convergence performance.

Call Admission Control (CAC) is a very important issue in CDMA systems to guarantee a required quality of service (QoS) and to increase system capacity. In this paper, we proposed and analyzed the CAC scheme using multiple criterions (MCAC), which can provide a quicker processing time and better performance. One is based on the number of active users with the minimum/maximum threshold by considering the spillover ratio, and the other is based on the signal to interference ratio (SIR). If active users are lower/higher than the minimum/maximum number of users threshold (N_min )/(N_max ), we accept/reject the new call without any other considerations based on the first criterion. And if the number of active users is between the N_min and N_max, we consider the current SIR to guarantee QoS based on the second criterion. Then the system accepts the new call when the SIR satisfies the system requirements, otherwise, the call is rejected. The multiple criterions scheme is investigated and its performance is compared with the number of user based CAC and power based CAC.

This paper addresses the utility-based radio resource allocation problem in DS-CDMA systems carrying multimedia traffic. The proposed scheme, aiming at achieving optimal resource allocation, considers the joint power and data rate allocation. To avoid high computational complexity of nonlinear optimization, we reformulate the radio resource allocation problem as a market model, where resource is regarded as a commodity. Since the market model satisfies the incentive-compatible constraint, the optimal resource allocation can be obtained at the market equilibrium in a distributed manner. According to whether to allocate a minimal transmission data rate to each user, two algorithms, UCA and FCA, are proposed. UCA emphasize on maximizing system overall utilities, while FCA guarantees fairness to users. Simulation results show that the proposed radio resource allocation scheme and algorithms are flexible and efficient for multimedia DS-CDMA systems.

A prototype modem with unequal error protection (UEP) capability was developed using multiple block coded modulation (MBCM). Benefited from its unique structure, MBCM can be used to provide UEP straightforward. We propose a new method to increase the robustness of the carrier recovery process by taking advantage of the MBCM code structure. We also use a frame format to facilitate the synchronization operation. This modem was developed in preparation for the mobile satellite communication experiments using the Engineering Test Satellite of VIII-type (ETS-VIII). In addition of MBCM, some other types of modulation schemes have been implemented in the same modem to enable a range of communication experiments to be performed. The modem can operate at variable data rates. The results of laboratory measurements agreed well with computer simulation results. Typical link budgets based on the parameters of ETS-VIII are also presented.

Fading in mobile satellite communications severely degrades the performance of data transmission. It is commonly modeled with non-frequency selective Rayleigh fading. For this type of channel, a new structure for a bit-interleaved coded modulation (BICM) scheme is presented and evaluated to determine its effectiveness compared to previously proposed schemes. This scheme is referred to as rate-compatible punctured BICM (RCP-BICM), in that its BICM encoder is able to yield a wide range of data rates by using a punctured convolutional code obtained by periodically perforating parity bits from the output of a low-rate-1/2 systematic convolutional code. A trellis-coded modulation (TCM) scheme and a turbo TCM (TTCM) scheme are discussed and evaluated for comparison with the RCP-BICM scheme. Simulation results demonstrate that the RCP-BICM scheme with hard-decision iterative decoding is superior to the TCM scheme by 3 dB at a bit error rate (BER) of 10-5 over an Rayleigh fading channel, and comes at a BER of 10-5 within 1 dB of the TCM scheme over an additive white Gaussian noise (AWGN) channel.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

In finding the optimal solution of virtual-path bandwidth allocation for large-scale networks, existing searching algorithms frequently call the process which calculate the bandwidth for given call blocking probability (CBP) and traffic loads. This is an inverse process of calculating CBP for given traffic loads and bandwidth. Because there is no analytic expression of calculating CBP, the process of calculating bandwidth with given CBP and traffic adopts an iteration algorithm. It leads to a tedious computation process. In this letter, a fast bandwidth evaluation algorithm is proposed and applied to the field of virtual path bandwidth allocation that aims at minimizing the worst call blocking probabilities in the network. The algorithm is proved to be accurate and fast. Finally, we provide comparison curves for the exact optimal CBPs obtained in the case of using OPBM against that of DCLPBM aided by the fast bandwidth evaluation algorithm.

A round-ended wide straight slot cut in the broad wall of a rectangular waveguide is analyzed by the Method of Moments (MoM) using numerical eigenmode basis functions derived by the edge-based finite element method (FEM), referred to as MoM/FEM. The frequency characteristics of the calculated transmission coefficients are compared with the measured ones, and good agreement is observed for a wide variety of antenna parameters. For simpler analysis that does not use MoM/FEM, an equivalent rectangular slot approximation for a round-ended slot is discussed. The resonant frequencies of empirically introduced "equal-area" and "equal-perimeter" slots are compared with those of round-ended slots for a wide variety of parameters such as slot width, wall thickness and dielectric constant inside the waveguide. Based upon MoM/FEM, which can be a reliable reference, it is found that the equal-area slot always gives a better approximation of the order of 1% over that of the equal-perimeter one which is of the order of 5%. For higher accuracy, a new rectangular slot approximation of a round-ended slot is proposed to be a linear combination of equal-area and equal perimeter approximation. The error is around 0.25% for a wide variety of parameters such as slot width-to-length ratio, wall thickness and dielectric constant of the filling material inside the waveguide.

In this paper, we present a group communication protocol that achieves total ordering message delivery for mobile computing systems with multiple overlapping groups. Our mechanism is an efficient adaptation of the propagation-tree technique to the mobile computing environments. It takes advantages of the capability of stationary mobile support stations to overcome the deficiencies associated with mobile devices. We construct the propagation tree based on the stationary stations, rather than the mobile hosts. As a result, mobile hosts are relieved of the excessive load of forwarding messages and communications on wireless channels are confined to transmitting messages to destination processes. This is important considering that the bandwidth of the wireless channels is limited. Moreover, the proposed protocol employs a mechanism to synchronize transmissions within a wireless cell. This serves to avoid redundant transmissions of a message in a wireless network in an attempt to achieve better utilization of the network bandwidth. Our mechanism relies on a handoff operation to deal with mobility of mobile devices. The handoff procedure ensures a smooth integration of a mobile host into a new cell, while preserving reliability of communication and the total ordering property of message delivery.