The unfairness problem among TCP connections has been proved to be very severe in the IEEE 802.11-based wireless ad hoc networks because the hidden station problem still exists and the binary exponential backoff algorithm always favors the latest successful station. In this paper, a novel protocol, neighbor-medium-aware MAC (NEMA-MAC), is proposed to improve the TCP fairness. By adding a medium (channel) state field in the head of the traditional IEEE 802.11 MAC frame, the NEMA-MAC protocol provides a communication mechanism to resolve the hidden station problem. In addition, when a collision occurs, the new backoff algorithm makes the senders cooperatively adjust the contention window according to their local and neighbors' channel usage indexes. The simulation results show that TCP sessions can acquire satisfying fairness and increase the throughput in the NEMA-MAC-based multihop ad hoc networks.

The WDM optical networks currently being deployed are opaque optical networks, in which each link is optically isolated by transponders. To reduce the number of expensive transponders and switching ports, a hierarchical optical architecture consisting of all-optical waveband switching and opaque OEO switching has been proposed. Although this architecture requires fewer transponders and ports, it also requires a large number of wavelength (waveband) multiplexers and demultiplexers. Switching the optical path solely at the fiber level (i.e., by using fiber cross-connects, or FXCs) is desirable as a way to reduce the total node cost. If all the core nodes in an optical network are FXCs, however, the grooming of wavelengths for the optical fibers is only possible at the edge nodes. This leads to poor utilization of wavelength resources when there is only demand for small numbers of wavelengths, and as a result, the link cost increases. This problem can be solved by adding an OEO grooming function to some of the FXCs. In this paper, we propose an algorithm for designing optical cross-connect (OXC) functions on the basis of the FXC, thus minimizing the total network cost.

A digital/analog hybrid system is presented which implements the cardinal polynomial spline interpolation of arbitrary degree. Based on the fact that the (m-1)st derivative of a spline of degree m-1 is a staircase function, this system generates a cardinal spline of degree m-1 by m-1 cascaded integrators with a staircase function input. A given sequence of sampled values are transformed by a digital filter into coefficients for the B-spline representation of the spline interpolating the sampled values. The values of its (m-1)st derivative with respect to time are computed by the recurrence formula interpreting differentiation of the spline as difference of the coefficients. Then a digital-to-analog converter generates a staircase function representing the (m-1)st derivative, which is integrated by a cascade of m-1 analog integrators to make the expected spline. In order to cope with the offset errors involved in the integrators, a dynamical sampled-data control is attached. An analog-to-digital converter is employed to sample the output of the cascaded integrators. Target state of the cascaded integrators at each sampling instance is computed from the coefficients for the B-spline representation. The state error between the target and the estimated is compensated by feeding back a weighted sum of the state error to the staircase input.

This paper presents a novel design procedure for class E oscillator. It is the characteristic of the proposed design procedure that a free-running oscillator is considered as a forced oscillator and the feedback waveform is tuned to the timing of the switching. By using the proposed design procedure, it is possible to design class E oscillator that cannot be designed by the conventional one. By carrying out two circuit experiments, we find that the experimental results agree with the calculated ones quantitatively, and show the validity of the proposed design procedure. One experimental measured power conversion efficiency is 90.7% under 6.8 W output power at an operating frequency 2.02 MHz, the other is 89.7% under 2.8 W output power at an operating frequency 1.97 MHz.

This paper presents a new approach based on current mode circuits for fast and accurate optical level monitoring with wide dynamic range of a gigabit burst-mode laser driver chip. Our proposed solution overcomes the drawbacks that voltage mode implementations show at higher bit rates or in other technologies. The main speed-limiting factor of the level monitoring circuitry is the parasitic capacitance of the back facet monitor photodiode. We propose the use of an active-input current mirror to reduce the impact of this parasitic capacitance. The mirror produces two copies of the photo current, one to be used for the "0" level measurement and another for the "1" level measurement. The mirrored currents are compared to two reference currents by two current comparators. Every reference current needs only one calibration at room temperature. A pattern detection block scans the incoming data for patterns of sufficiently long consecutive 0's or 1's. At the end of such a pattern a valid measurement is present at the output of one of the current comparators. Based on these measurements the digital Automatic Power Control (APC) will adjust the bias (IBIAS) and modulation current (IMOD) setting of the laser driver. Tests show that the chip can stabilize and track the launched optical power with a tolerance of less than 1 dB. In these tests the pattern detection was programmed to sample the current comparators after 5 bytes (32 ns at 1.25 Gbps) of consecutive 1's and 0's. Automatic power control on such short strings of data has not been demonstrated before. Although this laser transmitter was developed for FSAN GPON applications at a speed of 1.25 Gbps upstream, the design concept is generic and can be applied for developing a wide range of burst mode laser transmitters. This chip was developed in a 0.35 µm SiGe BiCMOS process.

Three representative protocols are proposed to support mobility for IPv6 in IETF: Mobile IPv6, Hierarchical Mobile IPv6, and Fast Handovers for Mobile IPv6. Recently, IEEE 802.11 network has been widely deployed in public areas for mobile Internet services. In the near future, IPv6 mobility support over IEEE 802.11 network is expected to be a key function to actualize the pure IP-based mobile multimedia service. The IPv6 mobility support protocols have their characteristics in terms of signaling, handover latency, lost packets, and required buffer size. In this paper, we analyze the performance of the protocols over IEEE 802.11 network. We define a packet-level traffic model and a system and mobility model. Then, we construct a framework for the performance analysis. We also make cost functions to formalize each protocol's performance. Lastly, we investigate the effect of varying parameters used to show diverse numerical results.

A Power on Reset signal generation circuit referencing threshold voltage without standby current consumption has been proposed. The POR signal is generated when supply voltage is larger than the sum of threshold voltages of N- and P-MOSFET.

In this study, a CG animation tool was designed that allows interpolation and extrapolation of two types of repeated motions including finger actions, for quantitative analyses of the relationship between features of human motions and subjective impressions. Three-dimensional human motions are measured with a magnetic motion capture and a pair of data gloves, and then relatively accurate time-series joint data are generated utilizing statistical characteristics. Based on the data thus obtained, time-series angular data of each joint for two dancing motions is transformed into frequency domain by Fourier transform, and spectral shape of each dancing action is interpolated. The interpolation and extrapolation of two motions can be synthesized with simple manner by changing an weight parameter while keeping good harmony of actions. Using this CG animation tool as a motion synthesizer, repeated human motions such as a dancing action that gives particular impressions on the observers can be quantitatively measured and analyzed by the synthesis of actions.

Point-to-point optical and millimeter wave communication has recently been of interest, especially in urban areas. Its benefits include simpler and easier installation compared with a land-based line. However, this technology suffers when adverse weather conditions are present, such as rain, fog and clouds, which induce scattering and absorption of the optical wave. The effects of scattering and absorption degrade the quality of the communication link resulting in increase of bit-error-rate. Therefore, there exists a need for accurate channel characterization in order to understand and mitigate the problem. In this paper, radiative transfer theory is employed to study the behavior of amplitude modulated signal propagating through a random medium. We show the effect of the medium to a modulated signal and relate the outcome on the quality of the communication link.

With the explosive growth of the Internet system, demands for broadband communication networks have rapidly increased to provide high quality network services. For this purpose, the IEEE 802.6 MAC standard protocol defines the distributed-queue dual bus (DQDB) for metropolitan area networks (MANs). The isochronous channel reuse problem (ICRP) has been studied for efficient use of DQDB by finding proper channel assignments to incoming connection requests. In this paper, we first define the generalized isochronous channel reuse problem (GICRP) as a generalization of ICRP, to afford demands of simultaneously satisfying plural connection requests such as for multicast applications, where certain sets of connection requests must be assigned channels simultaneously. We prove the NP-completeness of its decision problem. Then, we propose a minimum dead space (MDS) algorithm as a heuristic approach to GICRP. The extensive simulation results show that with shorter computation time, our MDS algorithm can always find better channel assignments reducing the waiting time for packet transmissions than the best existing algorithm for conventional ICRP.

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.

Decision-directed, pilot-symbol-aided channel estimation (PSACE) for coded orthogonal frequency division multiplexing (COFDM) systems has structurally unavoidable processing delay owing to the generation of new reference data. In a fast fading environment, the channel condition which varies during the delay induces channel estimation error. This paper proposes a method of reducing this estimation error. In this method, channel equalization is performed for the received signal twice. One is done as pre-equalization with the delayed estimates of channel frequency response in order to update them periodically. At the same moment, the other is done as post-equalization for the received signal that is delayed by the processing delay time, with the same estimates as the pre-equalization. By the proposed method, more accurate channel estimation can be realized without significant output delay. Computer simulations are performed by utilizing the IEEE 802.11a packet structure of 24 Mbit/s. The result shows that the proposed OFDM transmission scheme having the delay time of 20 µs offers 2.5 dB improvement in the required Eb/N0 at PER = 10-2 in the ESTI-BRAN model C Rayleigh fading channel with fd = 500 Hz.

This paper deals with a design problem of an adaptive robust control system for linear systems with structured uncertainties. The control law consists of a state feedback with a fixed gain designed by using the nominal system, a state feedback with an adaptive gain tuned by a parameter adjustment law and a compensation input. We show the parameter adjustment law and that sufficient conditions for the existence of the compensation input are given in terms of linear matrix inequalities (LMIs). Finally, a numerical example is included.

This paper describes a noise robustness technique that normalizes the cepstral amplitude range in order to remove the influence of additive noise. Additive noise causes speech feature mismatches between testing and training environments and it degrades recognition accuracy in noisy environments. We presume an approximate model that expresses the influence by changing the amplitude range and the DC component in the log-spectra. According to this model, we propose a cepstral amplitude range normalization (CARN) that normalizes the cepstral distance between maximum and minimum values. It can estimate noise robust features without prior knowledge or adaptation. We evaluated its performance in an isolated word recognition task by using the Noisex92 database. Compared with the combinations of conventional methods, the CARN could improve recognition accuracy under various SNR conditions.

Various effective and draft legislations and rules in Europe (WEEE--Waste Electrical and Electronic Equipment ROHS--Restrictions on the use of certain substances and ELV--End of life of vehicles) and Japan (Recycling Law for Home Electric Appliances) have either targeted restrictions or fully banned on the use of lead, to be enforced from 2001, 2003 and 2006 onwards. Up to now, mainly tin-lead alloys have been used in electronics. The process temperatures usually applied have been in the range of 230. All currently discussed lead-free alternatives for professional electronics need process temperatures which are at least 20 higher. In addition, the process duration is significantly longer. The combination of higher process temperatures and longer duration results in significant thermal stress on electromechanical devices. In particular the precision mechanics of electromechanical relays must withstand the solder process with maximum process temperatures of 255 without dimensional changes. During the transition from tin-lead to lead-free solder processes all combinations of component surfaces and solder must be possible. The selection of pure Sn100 or SnCu0.7 as terminal surface allows mixed assemblies with tin-lead as well as lead-free solders. All tested combinations of terminal surface, PCB surface and solder showed good results. From these results it can be concluded that mixed assemblies are possible during the transition time without any negative impact on the reliability of the electronic devices.

The purpose of this study is to examine the impact of the opening speed on a breaking arc. The opening speeds are 10, 20 and 30 mm/s. The breaking arc is generated in a D.C. 42 V/10.5 A circuit, and the arc voltage, the arc current, the gap length and the arc spectrum intensity are measured. Arc temperature is calculated by using a Boltzmann plot. Even if the opening speed is changed, the arc temperature starts from a high temperature, and falls gradually to 4650-4750 K with time. Namely, the opening speed has no influence on the arc temperature.

We propose a new dynamic routing algorithm that uses traffic and network state information to minimize the blocking rate and link congestion level. Our scheme uses the currently available link capacity in calculating the link weight by modifying Wang's approach, and computes the shortest path when a new call comes into the network. We consider the blocking count based update mechanism and the timer based mechanism, and conclude that the former is better than the latter in terms of efficiency and complexity.

This letter describes consecutive zero and one bits detection circuits designed for a 1.25 Gbit/s burst mode laser driver realized in a SiGe 0.35 µm BiCMOS technology with 3.3 V power supply. The architecture is based on a frequency divider and a delay line counting per four consecutive zero or one bits. The detector was designed with high-speed split-output stage flip-flops modified to have a reset input. Experimental results validate the design of the detector.

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.

Multi-Input Multi-Output (MIMO) systems, which utilize multiple antennas at both the receiver and transmitter, promise very high data rates in a rich scattering environment. It was proven in literature that with optimal power allocation, MIMO eigenmode transmission system (EMTS) is optimal because MIMO capacity is maximized. However, the performance of MIMO EMTS is very sensitive to the accuracy of channel state information and thus it is of practical importance to analyze its performance when channel state information is corrupted under realistic system and propagation conditions. In this paper, we lower bound the mutual information of MIMO EMTS with imperfect channel estimation and delayed quantized feedback in a spatially correlated continuous fading channel. Our results showed that this lower bound is tight and can serve as a comprehensive guide to the actual performance of MIMO EMTS under practical operating conditions.