A new system, in which a real-time VLBI (very-long-baseline interferometer) is utilized, for real-time monitoring of atmospheric disturbances on a very-long baseline has been developed. In this system, beacon waves from geo-stationary satellites are used for received signals and public communication lines are used for data transmission. Connecting the system to the 6-m Kagoshima and the 10-m Mizusawa radio telescopes enables atmospheric disturbances to be observed. The cross-correlation phase was calculated from the received signals, and the Allan standard deviation of the phase was obtained. It was found that the Allan standard deviation across almost the whole region of the time interval reflects atmospheric disturbances.

Call demand suddenly and greatly increases in the aftermath of a major disaster, because people want to check on their families and friends in the stricken area. Many call attempts in mobile cellular systems are blocked due to the limited radio frequency resources. In this paper, as a solution to this problem, limiting the holding time of calls is investigated and a dynamic holding time limit (DHTL) method, which varies the holding time limit dynamically based on the number of call attempts, is proposed. The effect of limiting the holding time is investigated first using a computer simulation with a constant and heavy traffic load model. This simulation shows that the average holding time of calls is decreased as the holding time limit is reduced. But it also shows limiting the holding time decreases the number of calls blocked and forced call terminations at handover considerably. Next, a simple estimation method for the holding time limit, which reduces the blocking rate to the normal rate for increasing call demand, is described. Finally, results are given of a simulation, which show that the DHTL method keeps good performance for a sudden and great traffic load fluctuation condition.

A generic multilevel quality-of-service (QoS) model for distributed multimedia applications is presented. QoS mapping mechanisms are required to translate the QoS parameters among the hierarchical levels. One QoS mapping mechanism based on the spline functions is proposed, hence two splines are compared. One is natural splines and the other is B-splines. QoS measurement experiments were conducted, and it is found that the B-splines give more accurate mapping results than the natural splines once the knots for the splines are selected appropriately.

Timed token protocols inadequately provide periodic communication service, although this is crucial for hard real time systems. We propose an algorithm to guaranteeing periodic communication service on a timed token protocol network. In this approach, we allocate bandwidth to each node so that the summation of bandwidth allocations is Target Token Rotation Time (TTRT). If a node cannot consume the allocated time, the residual time is made concession to other nodes for non-periodic service using a timer which contains the unused time value and is appended to the token. This algorithm can always guarantee transmission of real-time messages before their deadlines when the network utilization is less than 50%.

The transfer characteristic of an integrator is affected by excess-phase shift caused by the parasitic capacitance. The phase compensation is obtained by introducing zeros to generate phase lead. This paper proposes a phase compensation technique for the differential signal input integrator. The proposed technique is employing feedforward signal current source. The fifth-order leapfrog Chebyshev low-pass filter with 0.5 dB passband ripple is designed using the integrator with the proposed phase compensation. Further, an autotuning phase compensation system using the proposed technique is realized by applying a PLL system. The effectiveness of the proposed technique is confirmed by PSPICE simulation. The simulation results of the integrator with the proposed phase compensation shows that excess-phase cancellation is obtained at various unity gain frequencies. The accurate filter characteristic of the fifth-order leapfrog filter is obtained by using the autotuning phase compensation system. The passband of the filter is improved over wide range of frequencies. The proposed technique is suitable for low voltage application.

Progress in the fields of broadband access networks and information appliances has led to the introduction of a new network domain called Home Network. In 1999, HomePNA 2.0 using phone lines was proposed, and we believe it is one of the most promising solutions, because of its cost-effectiveness. However, it is not able to guarantee the QoS due to the adaptation of the mature IEEE802.3 CSMA/CD technology which is used for Ethernet. In light of this, we propose and evaluate a new MAC protocol for the Home Network called the HomeMAC that provides guaranteed QoS for appliances and PCs. HomeMAC features a hybrid CSMA/CD-Timed Token protocol which combines the CSMA/CD with timed token protocol and transmits real-time traffic based on the QoS Level Table (QLT) for guaranteeing QoS. In the HomeMAC, there are two different transmission modes, namely, the CSMA/CD Mode when there is no real-time traffic, and the Timed Token Mode when there is real-time traffic taking place. By dynamically switching the transmission mode between CSMA/CD Mode and Timed Token Mode in accordance with the different kinds of traffic, the hybrid protocol provides low delay, low jitter, and low loss rate to multimedia appliances such as TVs, DVDs, and PCs. Moreover, by providing flexible bandwidth allocation based on QLT, the HomeMAC can serve high QoS whole covering entire offered load.

This paper presents effective methods to calculate dual frame of the short-time Fourier expansion (STFE) in l2(Z). Based on a relationship between the prototype window used for generating a frame and the dual prototype window used for generating a dual frame in the STFE, two useful numerical methods with a finite frame operator are proposed to obtain finite support dual frames in time domain formulation. The methods can be used to construct the multiple STFE (MSTFE) suitable for a time-frequency analysis, synthesis and coding of discrete-time nonstationary signals. Numerical simulation results are given to verify the effectiveness of the calculation of dual frame.

This paper presents the analysis of integrated voice and data cellular networks with channel borrowing. Our considered system gives higher priority to handoff calls over new calls from users' point of view and reflects each characteristics of voice and data traffic types. Data handoff calls can wait in a queue while they are in handoff areas if there are no channels available. Voice handoff calls can borrow at most l channels from data calls if there are no idle channels upon their arrivals. We mathematically model this system by applying queueing theory. Then, we analyze its performance to derive the forced termination probability of data handoff calls, the blocking probabilities of the new and handoff calls of voice and data, and the Laplace Stieltjes transform for the distribution of waiting time in a queue. In numerical results, the analytical results for the mean waiting time of data handoff calls are compared with the simulation results to validate our analytical approach. Our system is also compared with the system where channel borrowing cannot be allowed (nonborrowing system) with respect to the blocking probabilities of the new and handoff calls of voice and data, the forced termination probability of data handoff calls, the mean and the coefficient of variation of the waiting time of data handoff calls.

This paper considers combining receptions with multiple receive antennas for space time coded MPSK signals over correlated Rayleigh fading channels. For the system with dual-antenna at receiver, a new transform is proposed, which can convert the correlated fading signals into uncorrelated ones. With the results obtained by using the proposed transform, the equivalent selective combining (SC) reception and maximum likelihood (ML) reception are presented. Theoretical analysis shows that ML reception has better performance than SC reception in terms of bit error rate. For the system with triple antenna at receiver, the simulation results are presented by using Monte Carlo method. All the results show that compared to using a receive antenna, a considerable signal to noise ratio gain can be obtained by using multiple receive antennas when the correlation coefficients among the receive antennas is not too high.

In this letter, we present an output feedback controller design technique for uncertain discrete time systems with multiple time-delays. Based on Lyapunov second method, a sufficient condition for the robust stability of the system with a dynamic controller is derived in terms of the linear matrix inequality (LMI) with respect to design variables. The solutions of the LMIs can be easily obtained using existing efficient convex optimization techniques.

In this paper, a time-to-digital converter in which the digital output is obtained without delay time is proposed. The circuit consists of a time-to-voltage converter, voltage-to-frequency converter, and counter. In the time-to-voltage converter, a capacitor is charged with a constant current during the input time interval. The change in the capacitor voltage is proportional to the input time and the capacitor voltage can be converted into a pulse signal with the voltage-to-frequency converter. The frequency of the pulse signal is directly proportional to the peak capacitor voltage and the pulse signals are counted to obtain the digital output. In the proposed circuit, the input time interval can be easily controlled and the resolution of the digital output can be improved by controlling the passive devices such as the capacitor and resistor.

In this paper, new integrated schemes of scheduling real-time traffic and cell loss control in high speed ATM networks are proposed for multiple priorities based on variable queue length thresholds for scheduling and the Partial Buffer Sharing policy for cell loss control. In our schemes, the queues for buffering arriving cells can be constructed in two ways: one individual queue for each user connection, or one physical queue for all user connections. The proposed schemes are considered to provide guaranteed QoS for each connection and cell sequence integrity for virtual channel/path characteristics. Moreover, these new schemes are quite flexible and can realize different scheduling algorithms. This paper also provides the Stochastic Petri Net models of these integrated schemes and an approximate analysis technique, which significantly reduces the complexity of the model solution and can be applied to real ATM switch models. From the numerical results, we can see that our schemes outperform those well-known schemes such as the head-of-line (HOL) priority control and the queue length threshold (QLT) policy.

This paper proposes a quasi-coherent equivalent-time sampling method to acquire repetitive wideband waveform signals with high throughput. We have already proposed a new sampling system which incorporates the pre-trigger ability and the time jitter reduction function for a fluctuated input signal while maintaining the waveform recording efficiency. The quasi-coherent sampling method proposed in this paper can be adopted to it in order to improve its data acquisition throughput significantly. Numerical simulation results show effectiveness of our proposed method.

In this paper, a novel method to reduce additive time-varying noise is proposed. Unlike the previous methods, the proposed method requires neither the assumption about noise nor the estimate of the noise statistics from any pause regions. The enhancement is performed on a band-by-band basis for each time frame. Based on both the decision on whether a particular band in a frame is speech or noise dominant and the masking property of the human auditory system, an appropriate amount of noise is reduced in time-frequency domain using modified spectral subtraction. The proposed method was tested on various noisy conditions: car noise, F16 noise, white Gaussian noise, pink noise, tank noise and babble noise. On the basis of segmental SNR, inspection of spectrograms and MOS tests, the proposed method was found to be more effective than spectral subtraction with and without pause detection in reducing noise while minimizing distortion to speech.

The long-term predictor (LTP) can effectively reduce the redundancy of the quasi-periodicity of speech signals. Traditional pitch predictors in linear predictive coding systems are developed by assuming that pitch gains are constant in each processing subframe. In this paper, we introduce a time-varied prediction gain to achieve a better representation of speech periodicity than the traditional approach. Due to the non-stationary variation of speech, the proposed LTP method can trace the detailed fluctuation of speech amplitude in both transient and stationary periods. Simulation results show that the proposed first-order varied-gain pitch predictor obtains a near speech quality as the fifth-order pitch predictor of the G.723.1 coder does; however, the former requires much lower computation than the latter.

The detection of timing constraint violation is crucial in reactive systems. A method of detecting deadline violation based on Floyd-Warshall shortest path algorithm has been proposed by Chodrow et al. We extend this method to detect the violation of minimum delay time in reactive systems where the repetition of event sequences frequently occurs.

This paper proposes new recursive fixed-point smoother and filter using covariance information in linear discrete-time stochastic systems. In this paper, to be able to treat the estimation of the stochastic signal, a performance criterion, extended from the criterion in the H estimation problem, is newly proposed. The criterion is transformed equivalently into a min-max principle in game theory, and an observation equation in a Krein space is obtained as a result. The estimation accuracy of the proposed estimators are compared with the recursive least-squares (RLS) Wiener estimators, the Kalman filter and the fixed-point smoother based on the state-space model.

The FDTD method needs Fourier analysis to obtain the fields of a single frequency. Furthermore, the frequency spectra of the fields used in the FDTD method ordinarily have wide bands, and all the fields in FDTD are treated as real numbers. Therefore, if the permittivity ε and the permeability µ of the medium depend on frequency, or if the surface impedance used for the surface impedance boundary condition (SIBC) depends on the frequency, the FDTD method becomes very complicated because of convolution integral. In the electromagnetic theory, we usually assume that the fields oscillate sinusoidally, and that the fields and ε and µ are complex numbers. The benefit of introduction of the complex numbers is very extensive. As we do in the usual electromagnetic theory, the authors assume that the fields in FDTD oscillate sinusoidally. In the proposed FDTD, the fields, ε, µ and the surface impedances for SIBC are all treated as the complex numbers. The proposed FDTD method can remove the above-mentioned weak points of the conventional FDTD method.

Scaled models for an anatomical head model and a simple head model are used to investigate the effects of head size on SAR characteristics for a cellular phone exposure at 835 MHz. From the results, we can see that a larger head produces a higher localized SAR and a lower whole-head averaged SAR.

An approach to the enhancement of speech signals corrupted by additive colored noise is proposed and the system architecture to implement the proposed idea in real-time communication is introduced in this paper. A combination of a bandpass FIR filtering technique with wiener filtering is used to improve the SNR for speech signals. The average SNR improvement (between input and output SNR) is 22.48 dB. The additive noises are the sound from a turbo prop aircraft. The system, which shows excellent performance, is designed based on a 16 bits fixed point DSP (ADSP-2181) from Analog Devices. Experiment results demonstrate that the FIR filter leads to a significant gain in SNR, thus visibly improvement for the quality and the intelligibility of the speech.