The ATM-ABR service category provides minimum cell rate (MCR) guarantees and robust connections even with insufficient network resources. Recently proposed rate-management algorithms for supporting multimedia applications over ABR mainly aim at minimizing the cell loss and delay. However, jitter is also an important element of QoS for multimedia applications. In this paper, we focus our attention on the arrival point of the critical cell corresponding to the end of data packet and propose a simple cell scheduling scheme for source node to reduce the jitter on application level over the ATM-ABR service class. In our proposed method, critical cells are delayed intentionally and the packet stream at application level becomes smooth. We verify the effectiveness of our proposed algorithm by an analytical model and simulation. From those results, we find that our proposed scheduling algorithm is effective in reducing the application level jitter even when the tagged cell stream is transmitted along the path with multiple nodes.

This paper presents a framework for modeling and mixed-mode simulation of circuits/interconnects and electromagnetic (EM-) radiations. The proposed framework investigates the signal integrity in VLSI chips, packages and wiring boards at the GHz-band level, and verifies the electromagnetic interference (EMI) and the electromagnetic compatibility (EMC) of high-speed systems. In our framework, the frequency characteristics of interconnects and EM-radiations are extracted by the full-wave FDTD simulation. The macromodels of interconnects are synthesized as SPICE subcircuits, and the impulse responses of EM-radiations are stored in the database. Once the macromodels are synthesized, the circuit simulation with the consideration of EM-effects can be performed by using SPICE. The EM-field distributions can be also easily calculated by taking convolutions of pre-simulated EM impulse responses and the SPICE results.

This paper presents a new method to evaluate time stamping schemes from three viewpoints: integrity of a time stamp, cost of issuing and verifying a time stamp and availability of the schemes. The main advantage of the proposed evaluation method is to clarify whether or not a certain scheme is optimal under certain prioritized requirements. Therefore, the proposed method can help potential users of time stamping services select an appropriate one which meets their prioritized requirements. In this paper, we explain the basic idea of the evaluation method and show how to use it by applying it to seven existing schemes.

This paper presents a beamspace-time transmit diversity scheme that uses a space-time block code (STBC) and a fixed multi-beam transmit array with low sidelobes for time-domain spreading orthogonal frequency-division multiplexing code-division multiplexing (OFDM-CDM) downlink transmission. The scheme assigns space-time-coded signals to a pair of neighboring beams via closed-loop beam selection. Time-domain spreading provides non-frequency selectivity in each spreading region, which makes it possible for multiple STBCs to share any beam and to be decoded after despreading. Simulation results demonstrated that multiple transmit beams and multiple receive antennas provide large beam gains and/or a high order of diversity gains. In addition, the proposed scheme spatially separates users by beam and thus alleviates multi-user interference.

A key problem under imperfect power control in multimedia DS-CDMA networks is how to guarantee the differentiated outage probabilities of different traffic classes resulted from the uncertainty of received powers. In addition, in order to utilize the scarce wireless resource efficiently, as many users as possible should be admitted into the network while providing guaranteed quality-of-service support for them. In this work, a call admission control scheme, Differentiated Outage Probabilities CAC or DOP-CAC, is proposed to achieve the above goals for imperfectly power-controlled multimedia CDMA networks. Two important features of CDMA systems are considered in our scheme: one is the power multiplexing among bursty traffics and the other is the power allocation scheme employed at the physical layer. The validity and efficiency of DOP-CAC are verified by numerical examples. Two power allocation schemes, Limited Optimal Power Allocation (LOPA) proposed in [3] and Quasi-Optimal Power Allocation (QOPA) we proposed in [6], are considered respectively and compared in the performance evaluation of DOP-CAC. The results show that DOP-CAC achieves much better resource utilization under QOPA than it does under LOPA. By employing QOPA at the physical layer and DOP-CAC at the link layer, our work suggests a high efficiency solution for QoS support of multimedia traffic under imperfect power control environment.

In this paper, the separate coding scheme is applied to space-time turbo-coded modulations (ST-TuCM). The separate coding for ST-TuCM uses the plural number of component encoders, each of which is a binary turbo encoder in the transmitter. The receiver has component decoders corresponding to the component encoders. The likelihood values derived by the component decoders are employed as a-priori information of transmitted signal from other transmit antennas in iterative demodulation-decoding. Simulation results under the fast Rayleigh fading channel show that separate coding and iterative demodulation-decoding improve the bit error rate performance.

This letter presents parallel algorithms for matrix multiplication and the fast Fourier transform (FFT) that are significant problems arising in engineering and scientific applications. The proposed algorithms are designed on a 3-dimensional processor array with separable buses (PASb). We show that a PASb consisting of N N h processors can compute matrix multiplication of size N N and the FFT of size N in O(N/h+log N) time, respectively. In order to examine ease of hardware implementation, we also evaluate the VLSI complexity of the algorithms. A result obtained achieves an optimal bound on area-time complexity when h=O(N/log N).

There is a strong demand to expand captioned broadcasting for TV news programs in Japan. However, keyboard entry of captioned manuscripts for news program cannot keep pace with the speed of speech, because in the case of Japanese it takes time to select the correct characters from among homonyms. In order to implement simultaneous subtitled broadcasting for Japanese news programs, a simultaneous subtitling system by speech recognition has been developed. This system consists of a real-time speech recognition system to handle broadcast news transcription and a recognition-error correction system that manually corrects mistakes in the recognition result with short delay time. NHK started simultaneous subtitled broadcasting for the news program "News 7" on the evening of March 27, 2000.

This paper proposes a high-level energy-optimizing algorithm which can synthesize low energy system VLSIs. Given an initial system hardware obtained from an abstract behavioral description, the proposed algorithm applies to it the three energy reduction techniques, 1) reducing supply voltage, 2) selecting lower energy modules, and 3) applying gated clocks. By incorporating our area/delay/power estimation, the proposed algorithm can obtain low energy system VLSIs meeting the constraints of area, delay, and execution time. The proposed algorithm has been incorporated into a high-level synthesis system and experimental results demonstrate effectiveness and efficiency of the algorithm.

Most network systems provide an authentication mechanism based on a user identification number and a password. In such systems, it is easy to obtain a user's password using a sniffer program with illegal eavesdropping. The one-time password method and the challenge-response method are useful authentication schemes that protect a user's password against eavesdropping. In client/server environments, the one-time password scheme using time is especially useful because it solves the synchronization problem. However, it has a problem of time-slippage, and this problem causes the authentication to be failed. In this paper, we propose an effective one-time password algorithm, which solves the time-slippage problem through the use of 1-bit information, which denotes the duration in which the authentication could be failed because of time-slippage. This algorithm can be added easily and quickly to current one-time password systems using time without requiring any change of protocols.

The problem of the super-resolution time delay estimation of the real stationary signals is addressed in this paper. The time delay estimation is first converted into a frequency estimation problem. Then a MUSIC-type algorithm to estimate the subsequent frequency from the single-experiment data is proposed, which not only avoids the mathematical model mismatching but also utilizes the advantages of the subspace-based methods. The mean square errors (MSEs) of the time delay estimate of the MUSIC-type method for varying signal-to-noise (SNR) and separation of two received signal components are shown to illustrate that they approximately coincide with the corresponding Cramer-Rao bound (CRB). Finally, the comparison between the MUSIC-type method and the other conventional methods is presented to show the advantages of the proposed method in this paper.

In this paper, we consider the similar software cost models with periodic rejuvenation to Garg, Puliafito, Telek and Trivedi (1995) under the cost effectiveness criteria. First, an alternative model as well as the original one are analyzed by Markov regenerative processes. We derive analytically the optimal periodic software rejuvenation policies which maximize the cost-effectiveness in the steady state for two models. Further, we develop statistical non-parametric algorithms to estimate the optimal software rejuvenation policies, provided that the sample data to characterize the system failure times are given. Then, the total time on test (TTT) concept is used. In numerical examples, we compare the periodic software rejuvenation policy with the non-periodic one, and investigate the asymptotic properties of the non-parametric estimators for the optimal software rejuvenation policies through a simulation experiment.

In this paper, a new approach is proposed to improve the channel estimation accuracy with channel tracking capability for adaptive multicarrier equalization systems under time-variant multipath fading channel. The improvement is carried out based on the assumption that the channel is static over a transmitted block period, and slowly linearly changing over several block periods. By applying IFFT to the concatenated channel transfer function derived from different blocks, the noise-averaging improvement is achieved, and a better estimation of the channel coefficients with some delay can be obtained. A multi-step channel predictor and a smoothing filter is utilized to compensate for the delay and make the system more robust in terms of channel tracking performance. Adaptive time domain equalization is jointly performed with this approach to avoid the channel invertibility problem found in the frequency domain approach. A short period of training sequences is utilized resulting in more efficient use of available communication capacity. The effectiveness of the proposed approach is evaluated through simulation for multicarrier systems in time-variant multipath fading channels. Results show improvement over previous channel estimation schemes.

In this paper, we propose a medium access control (MAC) protocol for multimedia code division multiple access (CDMA) communications. In the proposed protocol, a base station (BS) estimates the instantaneous number of simultaneously transmitted packets in the future slots with exploiting a stochastic property of traffic. In order to carry out this estimation, we employ an adaptive algorithm. We evaluate the performance of the proposed protocol by comparing that with two different cases. One is no estimation case and the other is perfect estimation case. From these results, we clarify the advantage of the proposed MAC protocol.

For real-time systems, multiprocessor support is indispensable to handle the large number of requests. Existing on-line scheduling algorithms such as Earliest Deadline First Algorithm (EDF) and Least Laxity Algorithm (LLA) may not be suitable for scheduling hard real-time tasks on multiprocessors. Although EDF has a low context switching overhead, it can produce arbitrarily low processor utilization. LLA has been shown as suboptimal, but has the potential for higher context switching overhead. We propose new on-line scheduling algorithms Earliest Deadline/Least Laxity (ED/LL) and Earliest Deadline Zero Laxity (EDZL) for identical multiprocessors. We show that ED/LL is suboptimal for multiprocessors and EDZL is suboptimal for two processors. Experimental results show that ED/LL and EDZL have low context switching overhead and low deadline miss rate.

This paper develops a modular synthesis algorithm for timed circuits that is dramatically accelerated by partial order reduction. This algorithm synthesizes each module in a hierarchical design individually. It utilizes partial order reduction to reduce the state space explored for the other modules by considering a single interleaving of concurrently enabled transitions. This approach better manages the state explosion problem resulting in a more than 2 order of magnitude reduction in synthesis time. The improved synthesis time enables the synthesis of a larger class of timed circuits than was previously possible.

In this paper, a VLSI architecture for lifting-based discrete wavelet transform (LDWT) is presented. Our architecture folds the computations of all resolution levels into the same low-pass and high-pass units to achieve higher hardware utilization. Due to the regular and flexible structure of the design, its area is independent of the length of the 1-D input sequence, and its latency is independent of the number of resolution levels. For the computations of analysis process of N-sample 1-D 3-level LDWT, our design takes about N clock cycles and requires 2 multipliers, 4 adders, and 22 registers. It is fabricated with TSMC 0.35-µm cell library and has a die size of 1.21.2 mm2. The power dissipation of the chip is about 0.4 W at the clock rate of 80 MHz.

A new approach to build up a real-time multiprocessing system that is configuration flexible for evaluating space-time (ST) equalizers is described. The core of the system consists of fully programmable devices such as digital signal processors (DSPs), field-programmable gate arrays (FPGAs), and reduced instruction set computers (RISCs) with a real-time operating system (RTOS). The RTOS facilitates flexibility in the multi-processor configuration for the system conforming with ST processing algorithms. Timing jitter synchronization caused by use of the RTOS-embedded system is shown, and an adjustable frame format for a transmission system is described as a measure to avoid the jitter problem. Bit error rate (BER) performances measured in uncorrelated frequency-selective fading channels show that an ST equalizer provides a significantly lower BER than an array processor does.

Asymmetric networks, which provide asymmetric bandwidth or delay for upstream and downstream transfer, have recently gained much attention since they support popular applications such as the World Wide Web (WWW). HTTP (Hypertext Transfer Protocol) is the basis of most WWW services so, evaluating the performance of HTTP on asymmetric networks is increasingly important, particularly real-world networks. However, the performance of HTTP on the asymmetric networks composed of satellite and terrestrial links has not sufficiently evaluated. This paper proposes new formulas to evaluate the performance of both HTTP1.0 and HTTP1.1 on asymmetric networks. Using these formulas, we calculate the time taken to transfer web data by HTTP1.0/1.1. The calculation results are compared to the results of an existing theoretical formula and experimental results gained from a system that combines a VSAT (Very Small Aperture Terminal) satellite communication system for satellite links (downstream) and the Internet for terrestrial links (upstream). The comparison shows that the proposed formulas yield more accurate results (compared to the measured values) than the existing formula. Furthermore, this paper proposes an evaluation formula for pipelined HTTP1.1, and shows that the values output by the proposed formula agree with those obtained by experiments (on the VSAT system) and simulations.

Recently, information display equipment such as a navigation system has often come to be installed in a vehicle, and a variety of useful information has been offered to the driver by voice and images while driving. The necessity of improving safety when the driver receives such information has come to be stressed. As one of the means of solving this problem, we can develop a system that presents the driving and road conditions information such as a lane changing car to the driver by using a warning sound. The purpose of our study is to clarify the effectiveness of an auditory display that uses spatial sounds on such a system. An experiment for measuring the driver's reaction time and eye movements to LED lighting during actual driving has been carried out to investigate whether the spatial sound can quicken the driver's operation and decrease human error. We evaluated the effectiveness by two measures, average reaction time and the number of largely delayed reactions. We considered that the average reaction time corresponds to the quickness of the driver's operation, and the number of largely delayed reactions corresponds to the probability of human error. As a result of the experiment, the use of directional sound clearly showed better performance than the use of monaural sound and no sound in the number of largely delayed reactions. Moreover, we analyzed the factors involved in delay of the reaction by the results of eye movement measurements. Consequently, it has been found that directional sound can decrease the number of the largely delayed reactions, which lead to an accident during actual driving.