This paper considers the design of quadrature amplitude modulation (QAM) transceivers for fixed wireless communications. We propose the use of power control in the QAM transmitter (Tx) to obtain BER performance robust to fading. The gain of the Tx is adaptively adjusted to keep the power of the received signal nearly constant despite of the short term fading and the second multipath. The BER performance of the proposed scheme is analytically evaluated in fixed wireless channels with flat fading and frequency selective fading. Analytic and simulation results show that the use of power control in the Tx can provide the BER performance only about 1 dB inferior to that in additive white Gaussian noise (AWGN) channel.

ATM networks are connection-oriented. Making a call requires first sending a message to do an admission control to guarantee the connections' QoS (quality of service) in the network. In this paper, we focus on the problem of translating a global QoS requirement into a set of local QoS requirements in ATM networks. Usually, an end-user is only concerned with the QoS requirements on end-to-end basis and does not care about the local switching node QoS. Most of recent research efforts only focus on worst-case end-to-end delay bound but pay no attention to the problem of distributing the end-to-end delay bound to local switching node. After admission control, when the new connection is admitted to enter the network, they equally allocate the excess delay and reserve the same bandwidth at each switch along the path. But, this can not improve network utilization efficiently. It motivates us to design a novel local QoS requirement allocation scheme to get better performance. Using the number of maximum supportable connections as the performance index, we derive an optimal delay allocation (OPT) policy. In addition, we also proposed an analysis model to evaluate the proposed allocation scheme and equal allocation (EQ) scheme in a series of switching nodes with the Rate-controlled scheduling architecture, including a traffic shaper and a non-preemptive earliest-deadline-first scheduler. From the numerical results, we have shown the importance of allocation policy and explored the factors that affect the performance index.

This paper addresses a technique for multiplexing voice streams into an IP packet for efficient VoIP transportation. The principles and distinctive schemes are presented and a comparative performance evaluation of the schemes based on a prototype is provided. In this paper, a scheme using a dynamic packet length threshold is proposed. As a conclusion, we show the scheme is beneficial for shortening packetization delay compared with the scheme in which multiple coded blocks in one voice stream are carried by an IP packet and superior with respect to reducing the header overhead and the number of generated packets.

The available bit rate (ABR) is an ATM service category that provides an economical support of connections having vague requirements. An ABR session may specify its peak cell rate (PCR) and minimum cell rate (MCR), and available bandwidth is allocated to competing sessions based on the max-min policy. In this paper, we investigate the ABR traffic control from a different point of view: Based on the decentralized bandwidth allocation model studied in [9], we prove that the max-min rate vector is the equilibrium of a certain system of noncooperative optimizations. This interpretation suggests a new framework for ABR traffic control that allows the max-min optimality to be achieved and maintained by end-systems, and not by network switches. Moreover, in the discussion, we consider the constrained version of max-min fairness and develop an efficient algorithm with theoretical justification to determine the optimal rate vector.

This paper discusses recognition up to intensities of mix of primary facial expressions in real time. The proposed recognition method is compatible with the MPEG-4 high level expression Facial Animation Parameter (FAP). In our method, the whole facial image is considered as a single pattern without any block segmentation. As model features, an expression vector, viz. low global frequency coefficient (DCT) changes relative to neutral facial image of a person is used. These features are robust and good enough to deal with real time processing. To construct a person specific model, apex images of primary facial expression categories are utilized as references. Personal facial expression space (PFES) is constructed by using multidimensional scaling. PFES with its generalization capability maps an unknown input image relative to known reference images. As PFES possesses linear mapping characteristics, MPEG-4 high level expression FAP can be easily calculated by the location of the input face on PFES. Also, temporal variations of facial expressions can be seen on PFES as trajectories. Experimental results are shown to demonstrate the effectiveness of the proposed method.

Detection of nonlinearly distorted signals is an essential problem in telecommunications. Recently, neural network combined conventional equalizer has been used to improve the performance especially in compensating for nonlinear distortions. In this paper, the self-organizing map (SOM) combined with the conventional symbol-by-symbol detector is used as an adaptive detector after the output of the decision feedback equalizer (DFE), which updates the decision levels to follow up the nonlinear distortions. In the proposed scheme, we use the box distance to define the neighborhood of the winning neuron of the SOM algorithm. The error performance has been investigated in both 16 QAM and 64 QAM systems with nonlinear distortions. Simulation results have shown that the system performance is remarkably improved by using SOM detector compared with the conventional DFE scheme.

Si single-electron transistors with a high voltage gain at a considerably high temperature have been fabricated by vertical pattern-dependent oxidation. The method enables the automatic formation of very small tunnel junctions having capacitances of less than 1 aF. In addition, the use of a thin (a few ten nanometers thick) gate oxide allows a strong coupling of the island to the gate, which results in a gate capacitance larger than the junction capacitances. It is demonstrated at 27 K that an inverting voltage gain, which is governed by the ratio of the gate capacitance to the drain tunnel capacitance, exceeds 3 under constant drain current conditions.

We analyze electron transport of silicon single-electron transistors (Si SETs) with an ultra-small quantum dot using a master-equation model taking into account the discreteness of quantum levels and the finiteness of scattering rates. In the simulated SET characteristics, aperiodic Coulomb blockade oscillations, fine structures and negative differential conductances due to the quantum mechanical effects are superimposed on the usual Coulomb blockade diagram. These features are consistent with the previously measured results. Large peak-to-valley current ratio of negative differential conductances at room temperature is predicted for Si SETs with an ultra-small dot whose size is smaller than 3 nm.

Improvements of Internet technology during the last decade have shifted the technical focus from reachability to the quality of communication. There are many technical frameworks, such as Integrated Service and Differentiated Services, which have been standardized to assure the quality of communication. QoS routing is also one of such frameworks. It changes or fixes a route that IP datagrams take, and is also indispensable to put a variety of services into practice. Nevertheless, experiment reports of QoS routing on operational network are quite few, especially with dynamic SLA. Therefore, we still do not know much about the important factors for QoS-enabled network to be realized, such as users' behavior, suitable services to offer, and configuration parameters. In this paper, we carried out field-trial with pseudo QoS routing and dynamic SLA in an actual network built at the WIDE retreat in autumn 2000. In this field-trial, we provided two different types of links to attendees. Attendees chose one of the links, through which their flows go, with our dynamic SLA. We describe the details and the results of this experiment. Our results could help to understand the customers' behavior for differentiated services, and therefore be useful for designing and deploying various QoS technologies.

In this paper we study the scalability issue in the design of a centralized bandwidth broker model for dynamic control and management of QoS provisioning. We propose and develop a path-oriented, quota-based dynamic bandwidth allocation mechanism for efficient admission control operations under the centralized bandwidth broker model. We demonstrate that this dynamic bandwidth allocation mechanism can significantly reduce the overall number of QoS state accesses/updates, thereby increasing the overall call processing capability of the bandwidth broker. Based on the proposed dynamic bandwidth allocation mechanism, we also extend the centralized architecture with a single bandwidth broker to a hierarchically distributed architecture with multiple bandwidth brokers to further improve its scalability. Our study demonstrates that the bandwidth broker architecture can be designed in such a manner that it scales with the increase in the network capacity.

This paper proposes a novel file transfer scheme named "Jam-packing file transfer" that consists of a call admission control and a packet scheduling mechanism. This combination can efficiently multiplex the traffic of file transfer and provides a guaranteed delivery time. Simulation results show the highness of extreme the multiplexing efficiency as the improvement in call blocking probabilities compared with the conventional rate-based reservation schemes. Furthermore, simulations of the packet scheduling indicate that file deliveries are done at the predicted delivery time.

In the past, a number of scheduling algorithms that approximate GPS, such as WFQ, have been proposed and have received much attention. This class of algorithms provides per-flow QoS guarantees in terms of the bounded delay and minimum bandwidth guarantee. However, with O(log N) computational cost for each new arrival scheduling, where N is the number of backlogged flows, these algorithms are expensive to implement (e.g., in terms of scalability). Moreover, none of them addresses the issues of delay distribution and jitter. In this paper, we propose a new traffic scheduling discipline called Jitter Control Frame-based Queueing (JCFQ) that provides an upper bound for delay jitter in the case of rate-controlled connections, such as packet video streams and IP telephony, while guaranteeing bounded delay and worst-case fair weighted fairness, such as in the WF2Q algorithm, but with O(1) complexity in selecting the next packet to serve, assuming that the number of flows is fixed. Three different algorithms for slot or service order assignment between flows are proposed: Earliest Jitter Deadline First (EJDF), Rate Monotonic (RM) and Maximum Jitter First (MJF). In these algorithms, delay jitter is formulated into the virtual finish time calculation. We compare the fairness, delay and jitter performance of the JCFQ with that of the MJF algorithm with WF2Q via simulation. The results show that with proper choice of the slot size, JCFQ can achieve better flow isolation in delay distribution than can WF2Q.

Recently, the Guaranteed Frame Rate (GFR) service was proposed as a new service category of ATM to support non-realtime data applications and to provide the minimum rate guarantee. To keep the simplicity of GFR as much as possible and overcome defects of FIFO-based mechanisms, we propose a FIFO-based algorithm extending DFBA one to improve the fairness and provide the minimum rate guarantee for a wider range of Minimum Cell Rate (MCR). The key idea is controlling the number of CLP1 cells which are occupying more buffer space than the fair share even when the queue length is below Low Buffer Occupancy (LBO).

We believe the quantum functional device to be a future perspective device, if we solve the problems that it has nowadays. We will summarize such problems with several discussions from the viewpoint of circuit and system.

The adaptive system design by 16-bit fixed point processing enables to employ an inexpensive digital signal processor (DSP). The narrow dynamic range of such 16 bits, however, does not guarantee the same performance that is confirmed beforehand by computer simulations. A cause of degrading the performance originates in the operation halving the word length doubled by multiplication. This operation rounds off small signals staying in the lower half of the doubled word length to zero. This problem can be solved by limiting the multiplier to only its sign () like the signed regressor algorithm, named 'bi-quantized-x' algorithm in this paper, for the convenience mentioned below. This paper first derives the equation describing the convergence property provided by a type of signed regressor algorithms, the bi-quantized-x normalized least mean square (NLMS) algorithm, and then formulates its convergence condition and the step size maximizing the convergence rate. This paper second presents a technique to improve the convergence property. The bi-qiantized-x NLMS algorithm quantizes the reference signal to 1 according to the sign of the reference signal, whereas the technique moreover assigns zero to the reference signal whose amplitude is less than a predetermined level. This paper explains the principle that the 'tri-qunatized-x' NLMS algorithm employing the technique can improve the convergence property, and confirms the improvement effect by computer simulations.

We have investigated the feasibility of the 16QAM-OFDM transmission scheme by calculating its transmission performance by means of some evaluation models. These models have been proposed by standardization projects such as ETSI-BRAN or ARIB-MMAC project. We evaluated the effects of the following four factors on transmission performance: (1) modulation scheme of each OFDM sub-carrier-channel; (2) FEC scheme, especially coding rate and coding scheme; (3) multipath fading environments; and (4) phase noise and nonlinear amplifier. By these evaluations, we have compiled a fundamental data in order to realize multimedia mobile access communication system based on 16QAM-OFDM transmission scheme.

This paper presents the characterization and validation of a time-domain physical model for illuminated high-frequency active devices and shows the possibility of use of the electromagnetic analysis of FDTD not only for electromagnetic interaction and scattering but also for the device simulation as a good candidate for a microwave simulator. The model is based on Boltzmann's Transport Equation, which accurately accounts for carrier transport in microwave and millimeter wave devices with sub-micrometer gate lengths. Illumination effects are accommodated in the model to represent carrier density changes inside the illuminated device. The simulation results are compared to available experimental records for a typical MESFET for validation purposes. Simulation results show that the microscopic as well as the macroscopic characteristics of the active device are altered by the light energy. This fact makes the model an important tool for the active device design method under illumination control.

Mobile communication channels always suffer serious frequency-selective fading due to multipath effect. Traditional spreading codes are characterized by the fact that their time-domain orthogonality is based on fixed chip width across a code period. They often fail to perform well under frequency-selective fading. This paper proposes a new class of CDMA codes, wavelet-packet orthogonal codes capable to retain time-domain orthogonality as well as to offer intra-code subband diversity to mitigate frequency-selective fading. The new codes are constructed by congregating several wavelets with various dilations and shifts. The combination of the wavelets in different nodes in a wavelet-packet full binary tree enables frequency diversity capability. Owing to the even code length, they can be easily used in mobile communications for multi-rate streaming and multi-code spreading. The performance study is carried out using correlation statistics distribution convolution algorithm and the results reveal that wavelet-packet codes, combined with RAKE receiver, perform better than traditional spreading codes in frequency-selective fading channels.

Emerging multimedia technologies introduce the prevalent multicast transmission, and the multicast tree is determined using the time-invariant network parameters. This paper addresses the time-varying multicast tree problem and presents path selection heuristics for multicast routing to determine an alternative path for real-time applications. A network is partitioned into the optimal region, the disjoint region, and the edge cutset if a branch of the multicast tree meets the un-guaranteed QoS condition. The path selection heuristics operate during the multicast session phase to efficiently select an alternative routing path containing an edge in the edge cutset to connect the multicast tree again. The source-based heuristics PS-SPT finds the path for minimal source-to-destination delay and the sharing-based heuristics PS-DDMC for minimal total cost. These path selection heuristics can efficiently provide solutions to keep the multicast transmission reliable. Simulation results also show that the proposed heuristics can provide effective good solutions for real-time multicast transmission. PS-SPT can select a path with optimal source-to-destination delay and PS-DDMC can select a path with optimal total cost.

Fairness as defined in wireline network cannot be achieved in wireless packet networks due to the bursty and location-dependent channel errors of wireless link. Channel-state dependent scheduling and compensation mechanism for errored flows are generally employed to improve the fairness in wireless packet scheduling algorithms. Most of the wireless scheduling algorithms, however, have two common significant problems. One problem is that they operate incorrectly unless all flows have the same packet size. This is due to the incorrect lead-and-lag model and the swapping-based rough compensation mechanism of the algorithms. The other problem is the degradation of error-free flow during compensation. The root of the degradation is that the bandwidth for compensation cannot be reserved since it is very difficult to predict. In this paper, we introduce WGPS (Wireless General Processor Sharing) to define fairness in wireless networks and present PWGPS as a packetized algorithm of WGPS. In our method, lead and lag of all flows are defined in terms of virtual time which is flexible enough to deal with packet size differences. Further, compensation is done by increasing the service share of the errored flow to a predetermined amount Δ. This enables the maximum bandwidth for compensation can be estimated and thus reserved. Our method can be proved to achieve improved fairness as compared with the previous scheduling algorithms.