To improve the throughput efficiencies of ARQ protocols over a high random packet-error channel, contiguous multiple copy-transmission (CMCT) strategy for which each packet is (re-)transmitted by sending its multiple copies in contiguous slots has been used so far. However, in burst error environments, all copies may be damaged in an error burst resultting the performance degradation of CMCT. To cope with this situation, we propose, in this paper, a new strategy called intermittent multiple copy-transmission (IMCT) whereby multiple copies are sent at a fixed interval. The throughput efficiency of go-back-N ARQ using CMCT or IMCT is analyzed and considered under a two-state Markov channel model expressing burst error property of a channel. As a result, it is shown that (i) the degree of improvement of throughput efficiency by CMCT or IMCT depends on the degree of error burst and (ii) the proposed IMCT can improve the throughput efficiency of go-back-N ARQ for high and burst error channels.

Efficient query processing in multi-dimensional indexing structures is an important issue for multimedia data applications. In this paper, we propose incremental k-nearest neighbor query (k-NNQ) and range query algorithms for R-tree based structures. The novel aspect of these algorithms is that they make use of the notion of VP filtering, a concept borrowed from the MVP-tree. The filtering notion allows for delaying of computational overhead until absolutely necessary. By so doing, we attain considerable performance benefits while paying insignificant overhead during the construction of the index structure. We implemented our algorithms and carried out experiments to demonstrate the capability and usefulness of our method. Results show that improvements range from 8% to 23% in response time for the experimental environment that we considered.

A new IDDQ testable design method is proposed for static CMOS PLA circuits. A testable PLA circuit of NOR-NOR type is designed using this method. It is shown that all bridging faults in NOR planes of the testable designed PLA circuit can be detected by IDDQ testing with 4 sets of test input vectors. The test input vectors are independent of the logical functions to be realized in the PLA circuit. PLA circuits are designed using this method so that the quiescent supply current generated when they are tested will be zero. Thus, high resolution of IDDQ tests for the PLA circuits can be obtained by using the testable design method. Results of IDDQ tests of PLA circuits designed using this testable design method confirm not that the expected output can be generated from the circuits but that the circuits are fabricated without bridging faults in NOR planes. Since bridging faults often occur in state-of-the-art IC fabrication, the testable design is indispensable for realizing highly reliable logic systems.

This paper focuses on the comparison of OFDM system channel estimation using time domain techniques and using frequency domain techniques. The channel model is based on the Taiwan DTV field-testing results, with static and dynamic multipath distortion. The simulation results prove that the channel estimation performance of the OFDM system in the time domain is better than in the frequency domain.

Over the future Internet, the real time communication generating such as CBR (Constant Bit Rate) traffic will widely spread, whereas the current Internet has no ability to provide QoS (Quality of Service) assurance for real time communication so far. In QoS networks, CBR traffic will have priority for its stringent QoS requirement over non-real time traffic such as TCP connections, which use the unused bandwidth left by CBR connections. Therefore, there is possibility that CBR traffic with priority causes TCP throughput degradation in QoS networks. For this reason, the performance of Tahoe TCP has been examined in that context, but other TCP variants such as Reno TCP, NewReno TCP and TCP with SACK option, which are now very common, have not yet been investigated clearly. In the present research, we will clarify how these TCP variants behave in QoS networks by means of simulations and compare their performance. From the results, SACK TCP can adapt very well to the changing bandwidth available and is very robust against the fluctuation, i.e., burstness, of CBR packet arrival process.

The differentiated services (diffserv) architecture has been proposed for implementing scalable service differentiation in the Internet. Expedited forwarding and assured forwarding have been standardized as Per-Hop Behaviors (PHB) in diffserv. Assured forwarding can be utilized to realize the service, which provides each user with a minimum guaranteed rate and a fair share of the residual bandwidth. We call it guaranteed rate (GR) service. With GR service, each packet for flow i is marked in or out based on comparison between the sending rate and the minimum guaranteed rate. When congestion occurs in networks, out packets are dropped more aggressively than in packets. Recently, several fair queuing schemes have been proposed for core stateless networks. They can achieve fairer bandwidth allocation than random early detection (RED). However, there have not been any studies that consider in/out bit usage to support GR service. This paper proposes how to extend the schemes that have been proposed for core stateless networks to allow the support of in/out bit usage. We present the performance of one of the extended schemes and compare the scheme to RED with in/out bit (RIO) in terms of fair bandwidth allocation.

We have developed a new scheme to provide Diffserv-based QoS over ATM access networks. Well-known Diffserv over ATM scheme requires some extension for conventional routers with ATM interfaces. The routers must map their Diffserv classes of services into ATM QoS classes and forward IP packets into prioritized VCs based on DSCP (DiffServ Code Point). The purpose of this work is to provide Diffserv-based QoS over ATM network using conventional IP over ATM interfaces on routers. We propose DSCP snooping at ATM edge nodes, which differentiates services over a single VC between two IP domains. A prototype circuit was used to evaluate this scheme.

In wireless ad hoc mobile network, a host desires to communicate with another host may need some intermediate nodes to relay data packets. To maximize the channel resource utilization and minimize the network transfer delay along the path, the shortest path with minimum hops approach is often adapted. However, by considering the employing medium access control (MAC) protocol, the minimum transfer delay from source to destination may be achieved by choosing a longer path but with less contention delay. In this paper, we will propose an efficient delay-oriented routing protocol for mobile ad hoc wireless networks. The expected access contention delay of IEEE 802.11 protocol is analyzed to support the routing decision. Simulation results show that the derived path length in proposed delay-oriented routing protocol is slightly higher than that of conventional shortest path with minimum hops approach but it can significantly reduce both average transfer delay and packet loss rate.

Sequence estimation (SE) of narrow-band digital FM signals, such as CPFSK and GMSK, with non-coherent limiter/discriminator (L/D) and integrate and dump (I&D) detection is investigated in detail using both analysis and simulation. The BER is studied from approximate upper and lower bounds obtained through Chernoff bounding techniques and minimum error event path probability along with a Gaussian noise assumption for high SNR. Various IF filters and the dependence of the error probability upon modulation index are considered. The results show an optimum modulation index around h 0.55, and clearly demonstrate the effectiveness and limitations of sequence estimation.

Recently, traffic of real-time communication such as video or IP telephony is increasing in Internet. Low delay and low jitter communication is required to perform these kinds of services smoothly. CoS (Class of Services) scheme which controls traffic by the class unit such as application is currently being paid attention. It is considered to be efficient control mechanism which utilizes characteristics of Internet traffic. One of effective methods for queueing control to realize CoS is CBQ (Class Based Queueing). Efficient and dynamic bandwidth allocation can be realized by using CBQ. However, CBQ mechanism is not necessarily efficient for real-time communication. In this paper, We clarify the effect of dynamic CBQ control scheme on real-time communication by experiment. In addition, we show that low delay and low jitter communication is possible by reducing unnecessary control overhead which is caused by traffic change. We propose new mechanism which changes bandwidth allocation of CBQ dynamically suppressing non real-time traffic and giving priority to real-time traffic on using bandwidth. We also show its efficiency by using real system.

In this paper we propose a new algorithm to approximate the solution of Hamilton-Jacobi-Bellman equation by using a three layer neural network for affine and general nonlinear systems, and the state feedback controller can be obtained which make the closed-loop systems be suboptimal within a restrictive training domain. Matrix calculus theory is used to get the gradients of training error with respect to the weight parameter matrices in neural networks. By using pattern mode learning algorithm, many examples show the effectiveness of the proposed method.

In order to increase the flexibility and the extensibility for packet transmission, the active network approach, that makes network nodes programmable, is very promising. Every packet includes a program specifying behavior of the packet at network nodes, and the network nodes only have the functions to execute programs of the packet. In an active network, network congestion management is achieved more flexibly and intelligently than that in the conventional internet. A number of mechanisms for congestion management are easily developed because various network functions such as QoS routing and congestion detection are easily combined in the same network nodes. By executing different routing protocols that use different levels of the link information, a congestion management scheme can be customized by users. This paper shows how QoS routing and congestion detection achieve a congestion management over an active internetwork system.

One drawback of Integrated service architecture is the scaling problem. Therefore, flow aggregation is an important solution for supporting quality of service in large-scale network. In advance reservation, priori information of advance-reserved requests can be used for flow aggregation before their initiation time. However, an impolitic aggregation can lead to violate admission control. In this paper, we propose an effective algorithm to aggregate advance-reserved requests with guaranteed delay. The proposed algorithm not only can reduce the amount of state in core network but also minimize the bandwidth consumption. The simulation result indicates that the state in the core network can be reduced as low as 17.3% even in the worst case.

This paper describes a realtime 3D sound localization algorithm to be implemented with the use of a low power embedded DSP. A distinctive feature of this implementation approach is that the audible frequency band is divided into three, in accordance with the analysis of the sound reflection and diffraction effects through different media from a certain sound source to human ears. In the low, intermediate, and high frequency subbands, different schemes of the 3D sound localization are devised by means of an IIR filter, parametric equalizers, and a comb filter, respectively, so as to be run realtime on a low power embedded DSP. This algorithm aims at providing a listener with the 3D sound effects through headphones at low cost and low power consumption.

A new scheduling algorithm called the Adaptive Weighted Round Robin with Delay Tolerance (AWRR/DT) is presented. This scheme can adapt to the traffic fluctuation of networks with a small processing burden. The proposed scheme incorporates a cell discarding method to reduce the QoS degradation in high-loaded (or congested) period. Simulation results show that the proposed scheme can reduce the average delay of the non-real-time (NRT) class, especially in high-loaded conditions, while maintaining the QoS of real-time (RT) classes. Our scheme with the discarding method can also reduce both the mean waiting time and cell loss ratio of RT classes.

In QoS networks, routing algorithms for QoS traffic have to provide the transmission path satisfying its QoS requirement while achieving high utilization of network resources. Therefore, server-based QoS routing algorithms would be more effective than distributed routing ones which are very common on the Internet. Furthermore, we believe that rerouting function enhances the advantage of their algorithms in which an already accepted flow with established path is replaced on some other path in order to accept newly arriving transmission request if it can not be accepted without doing so. Thus in this paper, we will propose a rerouting algorithm with the server-based QoS routing and evaluate its performance in terms of the blocking probability by computer simulation. In addition, we will investigate the impact of the amount of traffic with high-priority on the performance in some network topologies. Through some simulation results, we also discuss some issues arising in improving the effectiveness of rerouting.

A completely symmetric dual-level spiral inductor structure is proposed. The symmetry, area efficiency, the size dependence of the coupling factor, and the quality factors of the dual-level inductors are evaluated and compared with that of the single-level. This work demonstrates that, with most RF applications, the dual-level inductors are the better choice than the single-level.

This paper proposes a bandwidth allocation scheme which improves degradation of communication quality due to handoffs in mobile multimedia networks. In general, a multimedia call consists of several component calls. For example, a video phone call consists of a voice call and a video call. In realistic environments, each component call included in one multimedia call may have different requirements for quality-of-service (QoS) from each other, and priorities among these component calls often exist with respect to importance for communications. When the available bandwidth is not enough for a handoff call, the proposed scheme eliminates a low priority component call and defers bandwidth allocation for a component call whose delay related QoS is not strict. Moreover, in the allocation, the scheme gives priority to new calls and handoff calls over a deferred call and also performs bandwidth reallocation to eliminated component calls. By computer simulation, we evaluate the performance such as call dropping probability and show effectiveness of the proposed scheme.

In erbium doped optical fiber amplifiers (EDFAs) used in modern high-capacity wavelength division multiplexing (WDM) systems, the gain flatness of EDFA is very important in wide-band long-haul systems. In the EDFAs using the passive gain equalizers, the gain flatness deteriorates due to gain-tilt when the operating condition of the EDFA changes, while the EDFAs should maintain the gain flatness even if the operating condition has changed. To solve this problem, we have developed an active gain-slope compensation technique of an EDFA using a thulium-doped optical fiber (TDF) as a saturable absorber. The actively gain-slope compensated EDFA with the TDF compensator keeps the gain profile constant for the wide gain dynamic range more than 8 dB with the low noise figure less than 6 dB in the wavelength range of 1539-1564 nm.

Based on filtering ground clutter power directly in the frequency domain, a new non-coefficient Adaptive MTI (AMTI) scheme is presented in this letter. The results of simulation example show that this scheme has smaller signal-to-noise ratio loss than the classical AMTI based on spectral estimation, as well as high improvement factor.