This paper presents a novel digit-level algorithm for motion estimation (ME) and its hardware implementations. It uses the most-significant-digit-first (MSD-first) processing and on-line arithmetic ME components. A dedicated array architecture is also proposed for applications with high-throughput ME. Various fast search algorithms were presented in literatures to reduce the complexity but sacrifice the motion vector (MV) quality. Our MSD-first ME decomposes the summation of absolute differences (SAD) and comparison operations to digit level with MSD-plane first. These comparisons are interleaved into SADs to distinguish the MV as soon as possible. The algorithm precisely extracts the impossible candidates and removes their rest operations. It saves 47.4 % to 64.3 % of SAD computations in full search block matching (FSBM) ME. In the past, the high implementation cost of redundant number system prevented the practical use of on-line arithmetic. Besides, the redundant SAD removal results in irregular data flow in system-level integration. All these problems are solved by our novel architecture design. In this paper, we propose novel architecture designs to solve these problems. Besides, the architecture requires only one memory access per pixel to lower memory bandwidth by extensive data parallelism and a particular memory addressing while keeping the controller simple. A 4 4 array processor is implemented in 0.35 µm 1P4M CMOS cell library, with 2.84 ns cycle time and 1510 gates. It can support 83 M FSBM operations per second. After normalization, our implementation can support 2.67 times SAD operations per unit area (estimated in gate count) of the conventional two's complement ones. MSD-first ME can realize with other ME algorithms to improve the performance as well.

A high-speed 3-D camera has a future possibility of wide variety of application fields such as quick inspection of industrial components, observation of motion/destruction of a target object, and fast collision prevention. In this paper, a row-parallel position detector for a high-speed 3-D camera based on a light-section method is presented. In our row-parallel search method, the positions of activated pixels are quickly detected by a row-parallel search circuit in pixel and a row-parallel address acquisition of O(log N) cycles in N-pixel horizontal resolution. The architecture keeps high-speed position detection in high pixel resolution. We have designed and fabricated the prototype position sensor with a 12816 pixel array in 0.35 µm CMOS process. The measurement results show it achieves quick activated-position acquisition of 450 ns for "beyond-real-time" 3-D imaging and visual feedback. The high-speed position detection of the scanning sheet beam is demonstrated.

A special group of voice application services (VASs) are promising contents for wireless as well as wireline networks. Without a designated call admission policy, VAS calls are expected to suffer from relatively high probability of blocking since they normally require better signal quality than ordinary voice calls. In this letter, we consider a prioritized call admission design in order to reduce the blocking probability of VAS calls, which makes the users feel the newly-provided VAS in belief. The VAS calls are given a priority by reserving a number of channel-processing hardwares. With the reservation, the blocking probability of prioritized VAS calls can be evidently reduced. That of ordinary calls, however, is increasing instead. This letter provides a system model that counts the blocking probabilities of VAS and ordinary calls simultaneously, and numerically examines an adequate level of the prioritization for VAS calls.

For the reception of MC-CDMA signals in a frequency-selective fading channel, frequency-domain equalization is necessary before despreading. In this paper, joint antenna diversity combining and one-tap frequency-domain equalization is considered (simply referred to as the joint antenna diversity & equalization, in this paper). A receiver structure for joint antenna diversity & equalization is presented and the unified weights based on minimum mean square error (MMSE) criterion are found in the presence of multi-users with different spreading factors and transmit powers. For comparison, antenna diversity combining after despreading using MMSE combining (MMSEC) is also considered. The achievable bit error rate (BER) performances with joint antenna diversity & equalization and with antenna diversity after MMSEC despreading in a frequency-selective Rayleigh fading channel are evaluated by computer simulations and compared.

This letter presents a lower bound and approximation for the coverage probability of the pilot channel that can be used for a CDMA downlink design. The approximation of a compound truncated Poisson distribution is used to obtain a closed form equation for the coverage probability of the pilot channel. Computer simulations show that our lower bound curve is truly less than the empirical curve, and our proposed approximation agrees well with the empirical result.

We propose and demonstrate a directly modulated AM-VSB CATV system employing split-band and optical vestigial sideband (VSB) modulation techniques. Systems' performance can be improved by using optical VSB modulation and split-band techniques simultaneously. Our proposed directly modulated transmission system is simpler and more cost-effective than conventional externally modulated transmission system because of external modulator and complicated stimulated Brillouin scattering (SBS)-suppression method are not required.

The explosive growth of the Web contents has led to increasing attention on two major challenges: scalability and high availability of network file system. In this paper, based on our previous reports, we introduced the concept of intermediate file handle to cover the heterogeneity of file system and proposed a new data consistency scheme to reduce the overhead of write request in the reliable network file system. In addition, we also proposed a simple load-sharing mechanism for NFS client to switch to a lightly-load server in order to improve the response time of READ requests. With such an approach, NFS Clients are always active. They issued their requests and waited the corresponding replies. Finally, we analyzed the new data consistency scheme. It shows truly that the scheme is able to improve the performance of our network file system

Making up an electromagnetic wave simulator based on the FEM is tried, which may run on some widely used platforms by use of Java and a single commercial tool. Since the codes and configuration files to be created for this simulator are common, one can construct the simulator running on the platforms at the same time. Using this simulator, the transmission properties of two- and three-dimensional waveguide discontinuities in optical and microwave waveguides are analyzed, the inverse problem in material constant measurement is solved, and the computed results are presented including plots of the electric field distribution.

This paper presents measurement and analysis of various networks and applications using a high-speed IP meter. The authors have developed a high-speed IP meter with a GPS timestamp component, which enables precise measurement of packet delay and jitter in various networks. Measurement of the following networks was performed, 1) Measurement of traffic in a commercial IPv6 access service on ADSL, as a typical broadband access service network. 2) Measurement of traffic in the 54th IETF meeting in Yokohama, as a typical high-speed Internet backbone network. This paper reports the characteristics identified in these networks, e.g. asymmetricities of one-way packet delay over an ADSL access network and the difference in TCP/UDP packet delay over a high-speed backbone network. It also presents the analysis results of some multimedia applications in the Internet, and discusses the quality of service on Internet access service networks.

A two stage non-scan design for testability method is proposed. The first stage selects test points based on an earlier testability measure conflict. A new design for testability algorithm is proposed to select test points by a fault-oriented testability measure conflict+ in the second stage. Test points are selected in the second stage based on the hard faults after the initial ATPG run of the design for testability circuit in the preliminary stage. The new testability measure conflict+ based on conflict analysis of hard-faults in the process of test generation is introduced, which emulates most general features of sequential ATPG. The new testability measure reduces testability of a fault to the minimum D or controllability of the primary outputs, and therefore, does not need observability measure any more. Effective approximate schemes are adopted to get reasonable estimation of the testability measure. A couple of effective techniques are also adopted to accelerate the process of the proposed design for testability algorithm. Experimental results show that the proposed method gets even better results than two of the recent non-scan design for testability methods nscan and lcdft.

In this paper, we introduce an analytical model of CDMA reverse link power control based on the asymptotic theory. The effect of the SIR threshold value on the transient behavior is addressed and the optimal value of the threshold to achieve the fast convergence without oscillation is derived. In addition, the usefulness of the asymptotic analysis is confirmed by simulations.

This paper proposes a simulation framework suitable for holonic manufacturing systems, or HMS, based on the concept of distributed self-simulation. HMS is a distributed system that comprises autonomous and cooperative elements called holons, for the flexible and agile manufacturing. The simulation framework proposed here capitalizes on this distributed nature, where each holon functions similar to an independent simulator with self-simulation capabilities to maintain its own clock, handle events, and detect inter-holon state inconsistencies and perform rollback actions. This paper discusses the detailed architecture and design issues of such a simulator and reports on the results of a prototype.

Files of 3D mesh models are often large and hence time-consuming to retrieve from a storage device or to download through the network. Most 3D viewing applications need to obtain the entire file of a 3D model in order to display the model, even when the user is interested only in a small part, or a low-resolution version, of the model. Therefore, coding that enables multiresolution and ROI (Region Of Interest) transmission of 3D models is desired. In this paper, we propose a coding algorithm of 3D models based on partitioning schemes. The algorithm actually partitions the 3D meshes into some small sub-meshes according to some geometric criteria (such as curvatures), and then codes each small sub-meshes separately to transmit it progressively to users on demand. The key idea of this paper lies in the mesh partitioning procedure prior to its LOD control, which enables good compression ratio of the mesh data as well as some other good capable properties through network transmission such as ROI coding, view-adaptive transmission, error resilient coding, etc.

The performance of a GMPLS switching architecture with or without the flush capability is studied. For this switching architecture, we propose a queueing model that includes the control plane, the switch buffer mechanism, and the flush mechanism. The flush capability is included to reduce the out-of-sequence problem due to dynamic path changes. The behavior of aggregated streams, the label-setup and release policies, and the mechanisms for efficient resource allocation are all covered. With the proposed model, one can select appropriate parameters for the label-setup policy and the label-release policy to match the traffic load and network environment. Key performance metrics, such as the label-setup rate, the switching ratio, the bandwidth utilization of the label switched path, the average delay, and the average packet loss rate, can all be evaluated via this mathematical model. Numerical results and simulations are used to verify the accuracy of the proposed queueing model. Furthermore, we adopt trace simulation to certify the applicability the proposed model. The trade-off among these performance metrics can be observed as well.

A mathematical expression for the received signal power in a severe frequency-selective fading channel is derived. Using the derived expression, the signal power distributions are obtained by Monte-Carlo simulation and compared with the Nakagami m-power distribution. It is found that the power distribution matches well with the Nakagami m-power distribution when the multipath channel has a uniform power delay profile.

This paper presents a distinct approach to the robustness stability analysis and design of linear uncertain systems. Based on the extension version of the projection method, the specific stability issue, which ensures the poles within a specific region, can be efficiently analyzed. Furthermore, we derive a simple design scheme for a class of uncertain systems. By the proposed numerical algorithm, some examples are given to demonstrate the validity and effectiveness.

The transient phenomenon of electromagnetic waves caused by a time dependent resistive screen in a waveguide is treated. A boundary-value problem is formulated to describe the phenomena, in which the resistivity of the screen varies from one steady state to another in dependence on time. Application of Fourier analysis derives an integral equation, which is approximately solved by the method of least-squares. From the solution of the equation, the transient field is obtained by the inverse Fourier transform. By the use of the incomplete Lipschitz-Hankel integral for the computation of the field, numerical examples showing typical transient phenomenon are attached.

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 auction is a popular way of trading. Despite of the popularity of the auction, only a small number of papers have addressed the protocol which realize the double auction. In this paper, we propose a new method of double auction which improves the algorithm of the existing double auction protocol. Our new method is based on the idea of number comparison which is realized by homomorphic encryption. The new method solves the problem of the privacy of losing bids found in the existing algorithm. The buyers and the sellers can embed a random number in their bidding information by the use of the homomorphic encryption. The players in an auction cannot get anyone else's bidding information. The new method is more efficient than the existing ones. Our new method satisfies the criteria for the auction protocol.

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.