This paper deals with a study of a problem for finding the minimum-cost spanning tree with a response-time bound. The relation of cost and response-time is given as a monotonous decreasing and convex function. Regarding communication bandwidth as cost in an information network, this problem means a minimum-cost tree shaped routing for response-time constrained broadcasting, where any response-time from a root vertex to other vertex is less than a given time bound. This problem is proven to be NP-hard and consists of the minimum-cost assignment to a rooted tree and the minimum-cost tree finding. A nonlinear programming algorithm solves the former problem for the globally optimal solution. For the latter problem, different types of heuristic algorithms evaluate to find a near optimal solution experimentally.

A timer-based scheme is proposed to manage information within terminal and service profiles for both incall registration/deregistration of UPT users and incall registration resets of terminal owners. In the timer-based scheme, information related to incall registration for a UPT user in a terminal profile is deleted due to a timer expiration without accessing the terminal profile. The performance of the timer-based scheme is compared with the previously proposed request-based scheme in terms of; 1) total cost and, 2) the number of terminal profile accesses per unit time for a terminal. Even though provision of the timer-based scheme requires the modification of incoming call delivery procedure, the timer-based scheme can reduce both the total cost and the number of terminal profile accesses compared to the previously proposed request-based scheme.

Method of measuring disturbances at telecommunication ports has been published by IEC/CISPR. A method using both disturbance voltage and current probes is useful because it does not require any impedance stabilization networks (ISNs). In this paper, the values measured using this method are theoretically and experimentally compared with those measured using ISNs. An experiment using a simple circuit model presents that the value obtained by using this method is lower than that by using ISNs in some cases. A theoretical analysis however derives that the estimated value by adding the margin to the measured value is always guaranteed to be large compared with the value measured by ISNs. The analysis also indicates that the margin is dependent on the deviation of phase angle of ISN and can be calculated by a simple equation. The experiment using actual equipment shows that the estimated results including the margin is always larger than those measured by ISNs. The results of the study show that the method using both disturbance voltage and current probes can be used for measuring the disturbances by taking the margin into account, and this margin can be reduced by improving the phase angle characteristics of the common mode impedance of ISNs.

The authors propose and experimentally demonstrate an all-optical exclusive OR (XOR) logic gate based on self-phase modulation (SPM) of a semiconductor optical amplifier (SOA). The scheme is insensitive to the polarization of the input signal and requires no additional synchronized clock. The output of the XOR gate showed the contrast ratio of more than 17 dB for the input signal at 2.5 GHz.

EL (Electroluminescent) displays have been applied to automobiles, as their images are very clear and bright. High voltage, high integration and low power dissipation ICs are needed to drive these devices. To meet this, high voltage CMOS ICs using SOI (Silicon On Insulator) substrates are chosen as the driving devices. In this paper, an isolation structure between the output CMOS devices, of high density and high voltage is proposed. Conventional trench dielectric isolation shows degradation of a break down voltage with short distance from trench to source. In this work, the authors make clear the electric field distribution near the isolation, and offer a novel structure of "Field-plate Trench Isolation," which enables to relax the electric field on the silicon surface by shifting a part of electric field into surface oxide. Finally, operation of high voltage and high density, a 200-volt and 32-channel, EL display driver for automotive display panel is confirmed.

In this paper, we investigate the performance of Fast Reservation Protocols (FRP) for burst-level bandwidth allocation in ATM networks. FRP schemes can be classified into delayed transmission (DT) and immediate transmission (IT) methods according to reservation procedure. Moreover, according to the responsibility for negative acknowledgment (NAK) cell generations when burst blocking occurs, FRP schemes can be further classified into blocking node NAK (BNAK) and destination node NAK (DNAK) schemes. We analize the FRP schemes with different reservation and NAK methods for single node and multihop network models, respectively. We then discuss the dependence of performance for each FRP scheme on propagation delay, peak rate, and the number of hops.

It is well known that based on the structure of a transversal filter, the RLS equaliser provides the fastest convergence in stationary environments. This paper addresses an adaptive transversal equaliser which has the potential to provide more faster convergence than the RLS equaliser. A comparison is made with respect to computational complexity required for each update of equaliser coefficients, and computer simulations are demonstrated to show the superiority of the proposed equaliser.

We consider a dynamically reconfigureable network where dynamically changing traffic is offered. Rearrangement and adjustment of network capacity can be performed to maintain Quality of Service (QoS) requirements for different traffic classes in the dynamic traffic environment. In this work, we consider the case of a single, dynamic traffic class scenario in a loss mode environment. We have developed a numerical, analytical tool which models the dynamically changing network traffic environment using a time-varying, fluid-flow, differential equation; of which we can use to study the impact of adaptive capacity adjustment control schemes. We present several capacity adjustment control schemes including schemes which use blocking and system utilization as means to calculate when and how much adjustment should be made. Through numerical studies, we show that a purely blocking-based capacity adjustment control scheme with a preset adjustment value can be very sensitive to capacity changes and can lead to network instability. We also show that schemes, that uses system utilization as a means to calculate the amount of capacity adjustment needed, is consistently stable for the load scenarios considered. Finally, we introduce a minimum time interval threshold between adjustments, which can avoid network instability, in the cases where the results showed that capacity adjustment had been performed too often.

One major breakthrough on the communication society recently is the extension of networking from wired to wireless networks. This has made possible creating a mobile distributed computing environment and has brought us several new challenges in distributed protocol design. Obviously, wireless networks do have some fundamental differences from wired networks that need to be paid special attention of, such as lower communication bandwidth compared to wired networks, limited electrical power due to battery capacity, and mobility of processes. These new issues make traditional recovery algorithm unsuitable. In this paper, we propose an efficient algorithm with O(nr) message complexity where O(nr) is the total number of mobile hosts (MHs) related to the failed MH. In addition, these MHs only need to rollback once and can immediately resume its operation without waiting for any coordination message from other MHs. During normal operation, the application message needs O(1) additional information when it transmitted between MHs and mobile support stations (MSSs). Each MSS must keep an ntotal_h*n cell_h dependency matrix, where O(ntotal_h) is the total number of MHs in the system and ncell_h is the total number of MHs in its cell. Finally, one related issue of resending lost messages is also considered.

In this paper, a novel cumulant-based adaptive notch filtering technique for the enhancement and tracking of a single sinusoid in additive noise is presented. In this technique, the enhanced signal is obtained as the output of a narrow bandpass filter implemented using a second-order pole-zero constraint IIR adaptive notch filter, which needs only one coefficient to be updated. The filter coefficient, which leads to identifying and tracking the sinusoidal frequency, is updated using a suggested adaptive algorithm employing a recursive estimate of the kurtosis and only one-sample-lag point of a selected one-dimensional fourth-order cumulant slice of the input signal. Therefore, the proposed technique provides automatically resistance to additive Gaussian noise. It is also shown that the presented technique outperforms the correlation-based counterpart in handling additive non-Gaussian noise. Simulation results are provided to show the effectiveness of the proposed algorithm in comparison with the correlation-based lattice algorithm.

Recently, in the modem, the spread spectrum communication system and the software radio, Digital Signal Processor type Squaring Loop (DSP-squaring-loop) is employed in the demodulation of Binary Phase Shift Keying (BPSK) signal. The DSP-squaring-loop extracts the carrier signal that is used for the coherent detection. However, in case the Signal to Noise Ratio (SNR) is low, the DSP-Phase Locked Loop (DSP-PLL) can not pull in the frequency offset and the phase offset. In this paper, we propose a DSP-squaring-loop that is robust against noise and which uses the adaptive notch filter type frequency estimator and the adaptive Band Pass Filter (BPF). The proposed method can extract the carrier signal in the low SNR environment. The effectiveness of the proposed method is confirmed by the computer simulation results.

For evaluating the output response fluctuation of the actual environmental acoustic system excited by arbitrary random inputs, it is important to predict a whole probability distribution form closely connected with many noise evaluation indexes Lx, Leq and so on. In this paper, a new type evaluation method is proposed by introducing lower and higher order type functional models matched to the prediction of the response probability distribution form especially from a problem-oriented viewpoint. Because of the non-negative property of the sound intensity variable, the response probability density function can be reasonably expressed in advance theoretically by a statistical Laguerre expansion series form. The system characteristic between input and output can be described by the regression relationship between the distribution parameters (containing expansion coefficients of this expression) and the stochastic input. These regression functions can be expressed in terms of the orthogonal series expansion. Since, in the actual environment, the observed output is inevitably contaminated by the background noise, the above regression functions can not be directly employed as the models for the actual environment. Fortunately, the observed output can be given by the sum of the system output and the background noise on the basis of additivity of intensity quantity and the statistical moments of the background noise can be obtained in advance. So, the models relating the regression functions to the function of the observed output can be derived. Next, the parameters of the regression functions are determined based on the least-squares error criteria and the measure of statistical independency according to the level of non-Gaussian property of the function of the observed output. Thus, by using the regression functions obtained by the proposed identification method, the probability distribution of the output reducing the background noise can be predicted. Finally, the effectiveness of the proposed method is confirmed experimentally too by applying it to an actual indoor-outdoor acoustic system.

This paper proposes a low-distortion noise canceller and its learning algorithm which is robust against crosstalk and is applicable for continuous sounds. The proposed canceller consists of two stages: cancellation of the crosstalk and cancellation of the noise. A recursive filter reduces the number of computations for noise cancellation stage. Separate filters for the adaptation and the filtering are introduced for crosstalk cancellation. Computer simulations show 10 dB improvement of the error power.

Voltage follower is one of the most useful building blocks in analog circuits. This paper proposes a voltage follower composed of a complementary pair of p-channel MOS(PMOS) and n-channel MOS (NMOS) differential amplifiers which operates under low power supply. The proposed circuit has a rail-to-rail dynamic range by combining complementary differential amplifiers.

Loss analysis in bent graded-index optical slab waveguides is given using the modal-matching method. The conformal mapping replaces curved structure by an equivalent straight waveguide with a modified index profile. For this planar waveguide structure, the normal modes are calculated using a multilayer approximation method. The wave incident on the bend is expanded initially into a finite set of normal modes of the equivalent straight structure, and the transverse fields are matched across the junction. The numerical results show the loss formation in the graded-index waveguides and its dependence of the effective index of the corresponding straight waveguide.

A merged analog-digital circuit architecture is proposed for implementing intelligence in SoC systems. Pulse modulation signals are introduced for time-domain massively parallel analog signal processing, and also for interfacing analog and digital worlds naturally within the SoC VLSI chip. Principles and applications of pulse-domain linear arithmetic processing are explored, and the results are expanded to the nonlinear signal processing, including an arbitrary chaos generation and continuous-time dynamical systems with nonlinear oscillation. Silicon implementations of the circuits employing the proposed architecture are fully described.

In this paper, a new expression is derived for the bit error rate (BER) performance of Gray-encoded MDPSK for M=2 and 4 in orthogonal frequency division multiplexing (OFDM) systems over time-variant and frequency-selective Rayleigh fading channels. We assume that the guard time is sufficiently larger than the delay spread to solve the intersymbol interference (ISI) problem on the demodulated OFDM signal. In this case, the performance depends on the Doppler spread of fading channel. The closed form expression for the bit error probability of MDPSK/OFDM extended from the result in [5] shows that the BER performance of MDPSK is determined by (N + NG ) fD Ts where N is the number of subchannels, NG the length of the guard interval, fD the maximum Doppler frequency, and Ts the sampling period. The theoretical analysis results are confirmed by computer simulations for DPSK and QDPSK signals.

The paper proposes a transmitter diversity scheme with a desired signal selection for the mobile communication systems in which the severe cochannel interference (CCI) is assumed to occur at the base station. The feature of the proposed scheme is that the criterion of the downlink branch selection is based on the desired signal power estimated by the correlation between the received signal and the unique word at the matched filter. Moreover, the unique word length control method according to the instantaneous SIR is applied to the proposed scheme, taking account of the uplink transmission efficiency. Computer simulation results show that the proposed scheme provides the better performance than the conventional transmitter diversity in the severe CCI environments, and that the unique word length control method applied to the proposed scheme decreases the unique word length without the degradation of the transmission quality, comparing with the fixed unique word length method.

In this letter, we introduce new parameters for classifying digitally modulated unknown QAM and PSK signals. Our two parameters for the classification are the variance of magnitude ratios and the mean of mod 2π phase differences. The gain adjustments of amplitudes are not required for the classification. Five different types of QAM constellations and three different types of PSK constellations are tested and the characteristics of our classification parameters are investigated in various SNR environments. Simulation results demonstrate the effectiveness of our proposed technique.

This paper proposes two novel Multi-bit Delta-Sigma Modulator (Δ Σ M) architectures based on a Dual-Quantization architecture. By using multi-bit quantization with single-bit feedback, Both eliminate the need for a multi-bit digital-to-analog converter (DAC) in the feedback loop. The first is a Digital quantization-Error Canceling Multi-bit (DECM)-Δ Σ M architecture that is able to achieve high resolution at a low oversampling ratio (OSR) because, by adjusting the coefficients of both analog and digital circuits, it is able to cancel completely the quantization error injected into the single-bit quantizer. Simulation results show that a signal-to-quantization-noise ratio of 90 dB is obtained with 3rd order 5-bit quantization DECM-Δ Σ M at an OSR of 32. The second architecture, an analog-to-digital mixed (ADM)-Δ Σ M architecture, uses digital integrators in place of the analog integrator circuits used in the Δ Σ M. This architecture reduces both die area and power dissipation. We estimate that a (2+2)-th order ADM-Δ Σ M with two analog-integrators and two digital-integrators will reduce the area of a 4-th order Δ Σ M by 15%.