The hardware implementation of a neural network model using stochastic logic has been able to integrate numerous neuron units on a chip. However, the limitation of applications occurred since the stochastic neurosystem could execute only discrete-time dynamics. We have contrived a neuron model with continuous-time dynamics by using stochastic calculations. In this paper, we propose the circuit design of a new neuron circuit, and show the fabricated neurochip comprising 64 neurons with experimental results. Furthermore, a new asynchronous updating method and a new activation function circuit are proposed. These improvements enhance the performance of the neurochip greatly.

Single electron devices are ultra low power and extremely small devices, and suitable for implementation of large scale integrated circuits. Artificial neural networks (ANNs), which require a large number of transistors for being to be applied to practical use, is one of the possible applications of single electron devices. In order to simplify a single electron circuit configuration, we apply stochastic logic in which various complex operations can be done with basic logic gates. We design basic subcircuits of a single electron stochastic neural network, and confirm that backgate bias control and a redundant configuration are necessary for a feedback loop configuration by computer simulation based on Monte Carlo method. The proposed single electron circuit is well-suited for hardware implementation of a stochastic neural network because we can save circuit area and power consumption by using a single electron random number generator (RNG) instead of a conventional complementary metal oxide semiconductor (CMOS) RNG.

We study the statistical multiplexing performance of self-similar traffic. We consider that input streams have different QoS (Quality of Service) requirements such as loss and delay jitter. By applying the FBM (fractal Brownian motion) model, we present methods of estimating the effective bandwidth of aggregated traffic. We performed simulations to evaluate the QoS performances and the bandwidths required to satisfy them. The comparison between the estimation and the simulation confirms that the estimation could give rough data of the effective bandwidth. Finally, we analyze the bandwidth gain with priority multiplexing against non-prioritized multiplexing and suggest how to get better performance with the right configuration of QoS parameters.

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.

In this paper, we propose a set of broadcast streaming protocols designed for unidirectional radio channels. Considering the limited size and implementation overhead on a mobile terminal, the proposed protocol set is almost compliant with the current mobile streaming protocols, i.e. 3GPP PSS (Packet-switched Streaming Service), except for that the proposed protocols are designed to work on a unidirectional downlink channel. This protocol set enables flexible layout rendering by SMIL (Synchronized Multimedia Integration Language) in combination with SDP (Session Description Protocol), and reliable and synchronized static media (including still image and text) delivery by RTP (Real-time Transport Protocol) carousel. We present the prototype of this protocol set and measure its performance of video quality and waiting time for video presentation through a W-CDMA radio channel emulator and header compression nodes. From the experimental results, we show 1) trade-off between video quality and waiting time, 2) advantage and disadvantage of header compression, 3) effectiveness of synchronized transmission of SDP, SMIL, and I-frames of video objects, and 4) reliability of RTP-carousel. This protocol set is applicable to 3G MBMS (Multimedia Broadcast/Multicast Service) streaming service.

In this letter, we present an efficient stream authentication scheme that is an improvement of SAIDA. It is shown that under the same communication overhead its verification probability is higher than that of SAIDA. Moreover, its computation cost is lower than that of SAIDA.

In this paper, we propose the Synchronized Mobile Multicast (SMM) scheme for the real-time multimedia service to achieve three most important characteristics that the traditional Home Subscription (HS) and Remote Subscription (RS) mobile schemes cannot support. First, the SMM scheme supports the scalable one-to-many and many-to-many synchronized multimedia multicast on mobile IP networks to achieves seamless playback of continuous media streams even when both the mobile sender and receivers handoff simultaneously. Second, it analyzes the minimal buffer requirements of the mobile sender, the core router, the foreign agents and the mobile receivers in the multicast tree and formulates the initial playback delay within a handoff Guarantee Region (GR). Further, combined with the fine granularity scalability (FGS) encoding approach in the MPEG-4 standard, the SMM scheme achieves superior multimedia QoS guarantees and unlimited numbers of handoffs of the mobile sender and receivers only at the cost of degraded video quality for a short period after handoff with minimal extra bandwidth.

From the bifurcation viewpoint, this study examines a boost PFC converter with average-current-mode control. The boost PFC converter is considered to be a nonlinear circuit because of its use of a multiplier and its large duty cycle variation for input current control. However, most previous studies have implemented linear analysis, which ignores the effects of nonlinearity. Therefore, those studies were unable to detect instability phenomena. Nonlinearity produces bifurcations and chaos when circuit parameters change. The classical PFC design is based on a stable periodic orbit that has desired characteristics. This paper describes the main bifurcations that this orbit may undergo when the parameters of the circuit change. In addition, the instability regions in the PFC converter are delimited. That fact is of practical interest for the design process. Moreover, a prototype PFC circuit is introduced to examine these instability phenomena experimentally. Then, a special numerical program is developed. Bifurcation maps are provided based on this numerical study. They give a comprehensive outstanding for stability conditions and identify stable regions in the parameter space. Moreover, these maps indicate PFC converter dynamics, power factors, and regulation. Finally, numerical analyses and experimentation show good agreement.

Using daily infection data for Hong Kong we explore the validity of a variety of models of disease propagation when applied to the SARS epidemic. Surrogate data methods show that simple random models are insufficient and that the standard epidemic susceptible-infected-removed model does not fully account for the underlying variability in the observed data. As an alternative, we consider a more complex small world network model and show that such a structure can be applied to reliably produce simulations quantitative similar to the true data. The small world network model not only captures the apparently random fluctuation in the reported data, but can also reproduce mini-outbreaks such as those caused by so-called "super-spreaders" and in the Hong Kong housing estate of Amoy Gardens.

The authors present a multiparty signature generation (MSG) scheme of the Digital Signature Algorithm (FIPS 186-1). The scheme is based on a simple idea, however, it is much more convenient in usability in the real world than existing MSGs. The scheme has the following properties: (1) valid signatures are generated with odd n split private keys, (2) broadcast messages between the key holders are hidden from them, so that the n key holders do not need to process signature generation simultaneously, (3) even if up to t (= ) split keys are stolen, the adversary can get no information on the private key, (4) the scheme is as secure as the original signature algorithm against chosen message attack, and (5) the scheme is efficient in the sense that an implementation on smart card has demonstrated practical performance for interactive use with human user.

We propose a new algorithm for overdetermined blind source separation (BSS) based on multistage independent component analysis (MSICA). To improve the separation performance, we have proposed MSICA in which frequency-domain ICA and time-domain ICA are cascaded. In the original MSICA, the specific mixing model, where the number of microphones is equal to that of sources, was assumed. However, additional microphones are required to achieve an improved separation performance under reverberant environments. This leads to alternative problems, e.g., a complication of the permutation problem. In order to solve them, we propose a new extended MSICA using subarray processing, where the number of microphones and that of sources are set to be the same in every subarray. The experimental results obtained under the real environment reveal that the separation performance of the proposed MSICA is improved as the number of microphones is increased.

The AMR wideband speech codec was recently developed for high-quality wideband speech communications. Although it has an excellent performance due to expanded bandwidth of speech signal, it requires a huge amount of computation especially in codebook search. To solve this problem, this paper proposes an efficient codebook search method for AMR wideband codec. Starting from a poorly performing initial codevector, the proposed method enhances the performance of the codevector iteratively by exchanging the worst pulse in the codevector with a better one after evaluating the role of each pulse. Simulations show that the AMR wideband codec adopting the proposed codebook search method provides better performance with much less computational load than that using the standard method.

In order to produce precise enclosures from a multi-dimensional interval vector, we introduce a sharp interval sub-dividing condition for optimization algorithms. By utilizing interval inclusion properties, we also enhance the sampling of an upper bound for effective use in the interval quadratic method. This has resulted in an improvement in the algorithm for the unconstrained optimization problem by Hansen in 1992.

This letter describes the design and performance of a new ATM Adaptation Layer (AAL-UDP) for time-critical traffic over wireless ATM networks. The key ideas in the design consist of no discard at the AAL level and header protection with sequence number mechanism. The UDP/IP header is repeated for reliability, because it contains the most important information such as address and port number. The simulation results show that the AAL-UDP provides significant improvement in throughput as well as in application-level performance compared to the conventional AAL 5 case.

Computer simulation of cellular process is one of the most important applications in bioinformatics. Since such simulators need huge computational resources, many biologists must use expensive PC/WS clusters. ReCSiP is an FPGA-based, reconfigurable accelerator which aims to realize economical high-performance simulation environment on desktop computers. It can exploit fine-grain parallelism in the target applications by small hardware modules in the FPGA which work in parallel manner. As the first step to implement a simulator of cellular process on ReCSiP, a solver to perform a basic simulation of metabolism was implemented. The throughput of the solver was about 29 times faster than the software on Intel's PentiumIII operating at 1.13 GHz.

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.

Software rejuvenation is a preventive and proactive solution that is particularly useful for counteracting the phenomenon of software aging. In this paper, we consider both the periodic and non-periodic software rejuvenation policies under different dependability measures. As is well known, the steady-state system availability is the probability that the software system is operating in the steady state and, at the same time, is often regarded as the mean up rate in the system operation period. We show that the mean up rate should be defined as the mean value of up rate, but not as the mean up time per mean operation time. We derive numerically the optimal software rejuvenation policies which maximize the steady-state system availability and the mean up rate, respectively, for each periodic or non-periodic model. Numerical examples show that the real mean up rate is always smaller than the system availability in the steady state and that the availability overestimates the ratio of operative time of the software system.

This paper describes the design of a programmable QAM transceiver for VDSL applications. A 12-b DAC with 64-dB spurious-free dynamic range (SFDR) at 75-MS/s and an 11-b ADC with 72.3-dB SFDR at 70-MS/s are integrated in this complete physical layer IC. A digital IIR notch filter is included in order to not interrupt existing amateur radio bands. The proposed dual loop AGC adjusts the gain of a variable gain amplifier (VGA) to obtain maximum SNR while avoiding saturation. Using several low power techniques, the total power consumption is reduced to 300-mW at 1.8-V core and 3.3-V I/O supplies. The transceiver is fabricated in a 0.18-µm CMOS process and the chip size is 5-mm 5-mm. This VDSL transceiver supports 13-Mbps data rate over a 9000-ft channel with a BER < 10-7.

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.

The effect of the structure with difference on cross-section for the enlarged models that simulates signal transmission line (STL) in the magnetic head of HDD is discussed. The experimental results suggested that strip and shield structure are effective for suppression of EMI.