The concept of controlled resource sharing and dynamic quality of service (QoS) on the next generation Internet has attracted much attention recently. It is suggested that, by imposing real-time revision of shared resource allocated to individual media streams or data flows according to user/application QoS demand and resource availability, more balanced and efficient multimedia services can be provided. In this paper, we present an Adaptive Control Framework (ACF), which is developed for controlled resource sharing and dynamic QoS in real-time multimedia service. We discuss main elements of ACF including 1) Control schemes applicable in the framework, and 2) Control mechanisms used in ACF. It is clearly shown in this paper that, with control schemes and mechanisms incorporated in ACF and supportive algorithms and protocols for ACF applications on the Internet, more flexible service and better overall performance in terms of packet loss, latency, signal-noise ratio and re-synchronization delay, can be offered.

In a personal communication system (PCS), a scheme for reforwarding call-terminating setup messages (SETUP messages) from a network or a cell station is used to guard against their loss. We have developed a method for evaluating the loss probability of a reforwarding scheme in which the network monitors the response messages from a personal station after forwarding a SETUP message to that personal station and reforwards the SETUP message only if a response message is not received. We started with a stochastic model in which messages registered in the paging-channel queue in a cell station are cyclically forwarded to the wireless area. This model corresponds to the finite-capacity M/D/1/N model with vacation time. We then added a method for calculating the "timeout" probability. Next we expanded the model into one in which the SETUP messages are reforwarded when a response message is not received by the network. This model corresponds to the M/D/1/N model with vacation time and retrials. We then added an approximate method for calculating the loss probability. Finally, using the proposed methods, we clarified the traffic characteristics of PCS call-terminating control.

This paper proposes a new design method of nonlinear filtering and fixed-point smoothing algorithms in discrete-time stochastic systems. The observed value consists of nonlinearly modulated signal and additive white Gaussian observation noise. The filtering and fixed-point smoothing algorithms are designed based on the same idea as the extended Kalman filter derived based on the recursive least-squares Kalman filter in linear discrete-time stochastic systems. The proposed filter and fixed-point smoother necessitate the information of the autocovariance function of the signal, the variance of the observation noise, the nonlinear observation function and its differentiated one with respect to the signal. The estimation accuracy of the proposed extended filter is compared with the extended maximum a posteriori (MAP) filter theoretically. Also, the current estimators are compared in estimation accuracy with the extended MAP estimators, the extended Kalman estimators and the Kalman neuro computing method numerically.

Multimedia mobile communication systems with high-speed radio transmission supported by asynchronous transfer mode (ATM) technologies have been intensively studied over the last few years. Smaller radio zones termed microcells and picocells will be used in this kind of mobile communication systems for the purpose of high-speed radio transmission. When the coverage of a radio zone is smaller, the amount of traffic per radio zone is relatively low. It is not possible to use the cable circuits connecting the switch and base stations in an efficient manner because of the lack of the scale effect of traffic. With smaller radio zones, moreover, handoff occurs frequently as a mobile station moves. The switch is required a large capacity to handle the processing of frequent handoffs. This paper proposes a mobile network architecture controlled by the concentrated grouping of base stations. A special feature of this configuration is the ability of the network's switches to efficiently accommodate numerous base stations that control small radio zones. It can also lighten the handoff control load of switches; the effect of handoff frequency reduction is evaluated with computer simulation.

In this paper, a new digital chaos circuit which can generate multiple-scroll strange attractors is proposed. Being based on the piecewise-linear function which is determined by on-chip supervised learning, the proposed digital chaos circuit can generate multiple-scroll strange attractors. Hence, the proposed circuit can exhibit various bifurcation phenomena. By numerical simulations, the learning dynamics and the quasi-chaos generation of the proposed digital chaos circuit are analyzed in detail. Furthermore, as a design example of the integrated digital chaos circuit, the proposed circuit realizing the nonlinear function with five breakpoints is implemented onto the FPGA (Field Programmable Gate Array). The synthesized FPGA circuit which can generate n-scroll strange attractors (n=1, 2, 4) showed that the proposed circuit is implementable onto a single FPGA except for the SRAM.

This paper proposes a new distributed connection establishment scheme involving several competing network providers in a multimedia telecommunications environment. This connection establishment scheme, which is based on the concept of open competitive bidding, enables mutual selection by users and network providers. By employing this proposed scheme, both network providers and users can pursue their own objectives, according to their own bidding and awarding strategies. In this paper, a simple bidding strategy for network providers is presented, and the effectiveness of this strategy is evaluated by means of computer simulation. It is shown that each network provider can improve its profit by adopting this strategy. In this paper, an example of utility functions for users is presented, and the effectiveness of the mechanism with which users can select a network provider is also evaluated by means of computer simulation. Each user can improve his/her utility by selecting an appropriate network provider based on this utility function.

The current Internet does not offer any quality of service guarantees or support to Internet multimedia applications such as Internet telephony and video-conferencing, due to the best-effort nature of the Internet. Their performance may be adversely affected by network congestion. Also, since these applications commonly employ the UDP transport protocol, which lacks congestion control mechanisms, they may severely overload the network and starve other applications. We present an overview of recent research efforts in developing adaptive delivery models for Internet multimedia applications, which dynamically adjust the transmission rate according to network conditions. We classify the approaches used to develop adaptive delivery models with brief descriptions of representative research work. We then evaluate the approaches based on important design issues and performance criteria, such as the scalability of the control mechanism, responsiveness in detecting and reacting to congestion, and ability to accommodate receiver heterogeniety. Some conclusions are developed regarding the suitability of particular design choices under various conditions.

Low-voltage technique for IC is getting one of the most important matters. It is quite difficult to realize a filter which can operate at 1 V or less because the base-emitter voltage of transistors can hardly be reduced. A design of a low-voltage continuous-time filter is presented in this paper. The basic building block of the filter is a pseudo-differential transconductor which has no tail current source. Therefore, the operating voltage is lower than that of an emitter-coupled pair. However, the common-mode (CM) gain of the transconductor is quite high and the CMRR is low. In order to reduce the CM gain, a CM feedback circuit is employed. The transconductance characteristic is expressed as the function of hyperbolic cosine. The designed filter is a fifth-order gyrator-C filter. The transconductor and the filter which has a fifth-order Butterworth lowpass characteristic are demonstrated by PSpice simulation. Transconductance characteristic, CMRR and stability of the transconductor are confirmed through the simulation. In the analysis of the filter, frequency response and offset voltage are examined. It is shown that the filter which has corner frequency of the order of megahertz can operate at a 1 V supply voltage.

From our previous studies, we derived the worst case cell delay within an ATM switch and thus can find the worst case end-to-end delay for a set of real-time connections. We observed that these delays are sensitive to the priority assignment of the connections. With a better priority assignment scheme within the switch, the worst case delay can be reduced and provide a better network performance. We extend our previous work on the closed form analysis to conduct more experimental study of how different priority assignments and system parameters may affect the performance. Furthermore, from our worst case delay analysis on a regulated ATM switch, network traffic can be smoothed by a leaky bucket at the output controller for each connection. With the appropriate setting on the leaky bucket parameter, the burstiness of the network traffic can be reduced without increasing the delay in the switch. Therefore, fewer buffers will be required for each active connection within the switch. In this paper, our experimental results have shown that the buffer requirement can be reduced up to 5.75% for each connection, which could be significant, when hundreds of connections are passing through the switches within a regulated ATM network.

The full-band Monte Carlo technique is currently the most accurate device simulation method, but its usefulness is limited because it is very CPU intensive. This work describes efficient algorithms in detail, which raise the efficiency of the full-band Monte Carlo method to a level where it becomes applicable in the device design process beyond exemplary simulations. The k-space is discretized with a nonuniform tetrahedral grid, which minimizes the discretization error of the linear energy interpolation and memory requirements. A consistent discretization of the inverse mass tensor is utilized to formulate efficient transport parameter estimators. Particle scattering is modeled in such a way that a very fast rejection technique can be used for the generation of the final state eliminating the main cause of the inefficiency of full-band Monte Carlo simulations. The developed full-band Monte Carlo simulator is highly efficient. For example, in conjunction with the nonself-consistent simulation technique CPU times of a few CPU minutes per bias point are achieved for substrate current calculations. Self-consistent calculations of the drain current of a 60nm-NMOSFET take about a few CPU hours demonstrating the feasibility of full-band Monte Carlo simulations.

In this paper, we propose a new adaptive control system design using internal model principle (IMP) for a bounded polynomial parameters. In this method, we regard time varying parameters as variable disturbance and design an estimating law used the internal model of the disturbance so that the law is able to rejected the effectness of the disturbance. Our method has the features that the tracking error can converge to zero. Furthermore, we give a sufficient condition for the stability based on a small-gain theorem. The condition shows that our proposed method relax the stability condition more than the conventional methods based on a passivity theorem. Finally, we contain a numerical simulation to show an effect of our system.

In this paper, we focus on the evaluation of the Erlang capacity for multimedia DS-CDMA systems supporting the multi-class services with different transmission rates, bit error rates, traffic activity factors in the reverse link. The number of concurrent users of the corresponding service group is modeled as a K-dimension Markov chain. Then, the Erlang capacity for the multimedia system can be found based on a K-dimension M/M/m loss system. For an IS-95 type DS-CDMA system, supporting voice/data services, the capacity bounds are depicted in conjunction with the 2-dimensional Markov chain. Furthermore, it is observed that the Erlang capacity with respect to the each service group should be balanced to enhance the total system Erlang capacity. Finally, the channel reservation scheme is considered to increase the total system Erlang capacity.

This paper considers methods to design multiple-output networks based on decision diagrams (DDs). TDM (time-division multiplexing) systems transmit several signals on a single line. These methods reduce: 1) hardware; 2) logic levels; and 3) pins. In the TDM realizations, we consider three types of DDs: shared binary decision digrams (SBDDs), shared multiple-valued decision diagrams (SMDDs), and shared multi-terminal multiple-valued decision diagrams (SMTMDDs). In the network, each non-terminal node of a DD is realized by a multiplexer (MUX). We propose heuristic algorithms to derive SMTMDDs from SBDDs. We compare the number of non-terminal nodes in SBDDs, SMDDs, and SMTMDDs. For nrm n, log n, and for many other benchmark functions, SMTMDD-based realizations are more economical than other ones, where nrm n is a (2n)-input (n1)-output function computing (X2+Y2)+0.5, log n is an n-input n-output function computing (2n1)log(x1)/nlog2, and a denotes the largest integer not greater than a.

A variable partition duplex scheme on packet reservation multiple access protocol (VPD-PRMA) is analyzed in this paper. We assume a four-state speech model for a conversational pair and successfully obtain performance measures by approximate Markovian analysis. Analytical results show that they quite fit simulation results; and VPD-PRMA can get higher statistical multiplexing gain than fixed partition duplex (FPD)-PRMA, due to the trunking effect. We further investigate the effect of design parameters of permission probability and enlarged reservation duration on system performance by computer simulation. Simulation results shows that it exists appropriate values for these two design parameters so that the packet dropping probability can be minimized. The adjustment of permission probability can greatly improve the performance of uplink traffic with slight deterioration of the performance of downlink traffic; the provision of enlarged reservation duration scheme can enhance the system performance.

This paper proposes a new model which can approximate the delay time distribution in the Internet. It is well known that the delay time in communication links follows the exponential distribution. However, the earlier models cannot explain the distribution when a communication link is heavily overloaded. This paper proposes to use the M/M/S(m) model for the Internet. We have applied our model to the measurement results. This paper deals with one-way delay because it reflects the actual characteristics of communication links. Most measurement statistics in the Internet have been based on round-trip time delay between two end nodes. These characteristics are easily measured by sending sample packets from one node to the other. The receiver side echoes back the packets. However, the results are not always useful. A long distance communication link, such as a leased line, has two different fibers or wires for each direction: an incoming link, and an outgoing link. When the link is overloaded, the traffic in each link is quite different. The measurement of one-way delay is especially important for multimedia communications, because audio and video transmissions are essentially one-way traffic.

Although there are many multimedia presentation systems on the market, they have some shortcomings and most of them only can work on one single computer, and few of them can work on Web. Thus, in the thesis we develop a network multimedia presentation system to let users easily design the multimedia presentation without restriction on technology or presentation time and place. Our system includes 3 main components: User Interface that includes temporal specification editor, spatial specification editor and multimedia object interface, Presentation Interface and Knowledge Base. There is a dynamic homepage generator in our system and we propose a displaying algorithm based on the Allen's theory, that there exist 13 temporal relationships between two intervals, for synchronizing the media objects.

We present a new method for estimation of spectrum transition of nonstationary signals in cases of low signal-to-noise ratio (SNR). Instead of the basic functions employed in the previously proposed time-varying autoregressive (AR) modeling, we introduce a spectrum transition constraint into the cost function described by the partial correlation (PARCORR) coefficients so that the method is applicable to noisy nonstationary signals of which spectrum transition patterns are complex. By applying this method to the analysis of vibration signals on the interventricular septum (IVS) of the heart, noninvasively measured by the novel method developed in our laboratory using ultrasonics, the spectrum transition pattern is clearly obtained during one cardiac cycle for normal subjects and a patient with cardiomyopathy.

Certain relationships between successive packet loss patterns and packet spacing are described. To observe a successive packet loss pattern, one possible method is to investigate test packets that are generated at certain intervals (e. g. , Poisson interval, constant regular interval). However, successive packet loss strongly depends on the interval generated. If test packets are generated with long intervals, then successive loss pattern cannot be shown. Thus, in such a method, where the packet intervals may sometimes be long or short, the successiveness of the packet loss should be considered in terms of the packet spacing. To clarify the relationship between the successive packet loss and the packet spacing, we analyze data based on observation of an actual network with the loss window size as a parameter. We find that when the packet spacing is narrower, i. e. , has a shorter interval, the probability becomes higher that the packet immediately following a single packet loss would also be lost.

As TCP/IP networks develop, various type of applications or services are appearing. Especially, many people want to use real time multicast applications over TCP/IP networks like a TV conference system. Most of the current TCP/IP networks, however, still do not support QoS guarantees using RSVP, so that they provide only a best-effort service. Therefore, such real time applications must control data transmitting rate by the network or receiver's condition. However, it is difficult to control data rate over a multicast session, since every receiver on a multicast network does not necessarily have the same environment. To solve this problem, the authors proposed a locally adaptive control intermediate system. This paper describes a rate-adaptive real-time multicast system with locally adaptive packet flow control.

This paper proposes a method of improving the accuracy of the attenuation constant estimate obtained by using the cross-spectral technique. In the cross-spectral technique, the envelope of the estimated impulse response is deformed due to the use of a time window. As a result, the estimated impulse response decays more rapidly than the real impulse response does, and the attenuation constant obtained by the estimated impulse response becomes larger than the real value. This paper first describes how the attenuation constant changes in the process of impulse response estimation. Next, we propose a method of improving the accuracy of the estimation. The effect of the proposed method is confirmed by computer simulation.