This paper proposes a novel method to correct numerical phase velocity errors in FDTD meshes with nonuniform step size. It enables the complete compensation of the phase velocity errors introduced by the mesh grading for one frequency and one arbitrary direction of propagation independently of the mesh grading. This permits the usage of the Total-Field-Scattered-Field formulation in connection with electrically large nonuniform FDTD meshes and allows a general reduction of the grid dispersion errors. The capabilities of the proposed method are demonstrated with the help of two examples: (1) the fields in a dielectric sphere illuminated by a plane wave are calculated and (2) a patch antenna simulation demonstrates that the uncertainty in determining its resonance frequency can be reduced by about 50%.

This paper describes special-purpose hardware for large-scale logic simulation, called SP2, which executes an event driven algorithm and can simulate up to sixteen million gates. SP2 was developed, in 1992, for system verification of large-scale computer designs as a successor to SP1, which was developed in 1987. SP2 provides enhanced performance, throughput, and delay accuracy over SP1. Since 1992, SP2 has been widely used for system-level simulation of mainframes, super computers, UNIX servers and microprocessors. It is used as a powerful simulator, in all stages of design verification, or in early stages, before regression testing, by using emulators.

This paper presents a novel system-level design methodology, called quality-driven design, by which application-specific optimization can be achieved; furthermore the entire functionality can be shared to maximize design reuse. As a case of study, this paper focuses on quality-driven design for video applications and introduces an output quality adaptive approach based on variable bitwidth optimization to explore a new design space. MPEG2 video is used as the driver application to illustrate the potential of the presented methodology. Experimental results show the effectiveness of the methodology.

We clarify the effectiveness of receiver-side compensation in offsetting fiber Bragg grating (FBG) dispersion induced-electrical signal-to-noise ratio (SNR) degradation in a 10 Gb/s 8-channel wavelength-division multiplexing (WDM) 6,400 km transmission system. The receiver-side compensation greatly improves the SNR degradation. The allowable accumulated FBG dispersion is -400 1000ps/nm for the worst arrangement, a single FBG at the transmitter, which is about half the accumulated fiber dispersion permissible with receiver-side compensation.

A wireless communications system with a transmission speed of 18 Mbit/s is presented using the 2.4 GHz ISM band. This system employs the Carrier Frequency Offset-Spread Spectrum (CFO-SS) scheme and the Dual-Polarization Staggered Transmission (DPST) scheme. The 18 Mbit/s CFO-SS system (named CFO-SS18) was developed and its performance evaluated in fields. In this paper, the detailed operating principle of CFO-SS and DPST schemes, together with the specifications and structures of CFO-SS18, are presented. Results of indoor and field tests obtained by using CFO-SS18 are also presented.

In this paper, we propose and describe a new synchronizer for the FFT timing applicable to multi-carrier spread-spectrum (MC-SS) communication systems. The performance of the synchronizer is evaluated in terms of false- and miss-detection probabilities in the presence of additive white Gaussian noise (AWGN) and Rayleigh fading.

In this paper, we clarify the properties of the contract net protocol based on its formal specification. To specify the contract net protocol, we propose a formal specification method for an agent system. In this method, agents are modeled as communicating finite state machines. To deal with the behavior of agents and its time passage explicitly, we incorporate the concept of time into the communicating finite state machine. The contract net protocol is specified based on the specification method proposed in this paper. From the specification, we analyze the possibility about agent deadlock and its avoidance solution.

This letter presents text-independent speaker identification results for telephone speech. A speaker identification system based on Karhunen-Loeve transform (KLT) derived from codebook design using genetic algorithm (GA) is proposed. We have combined genetic algorithm (GA) and the vector quantization (VQ) algorithm to avoid typical local minima for speaker data compression. Identification accuracies of 91% were achieved for 100 Mandarin speakers.

The development of efficient quality of service (QoS) routing algorithms in a high-speed network environment is a very important and at the same time very difficult task due to the need to provide divergent services with multiple QoS requirements. Recently heuristic algorithms based on Lagrange relaxation techniques have been proposed to resolve the contradiction between the time complexity and the quality of solution. In this paper, we investigate the performance of two heuristic algorithms, LR_DCLC and NR_DCLC, for the delay-constrained least-cost (DCLC) routing problem. Algorithm LR_DCLC is based on linear relaxation, while algorithm NR_DCLC, which is proposed in this paper, is based on nonlinear relaxation. A large number of simulations demonstrate that even though both algorithms have very good performance, NR_DCLC can obtain much better solutions than LR_DCLC by running Dijkstra's algorithm on average a few more times, especially in the case when the optimal solutions are hard to find.

Automatic traffic surveillance based on visual tracking techniques has been desired for many years. This paper proposes a basic highway surveillance system using an HMM-based segmentation method. The presented system meets the essential requirement of ITS: real-time running. Its another advantage is robustness to the shadows of moving objects, which have been recognized as one of main obstacles to robust car tracking. At present, using the system we can estimate velocity of vehicles with high accuracy. For acquiring metric information in the real world, the system does not require a precise calibration but only needs four point correspondences between the image plane and ground plane.

An n-detection testing for stuck-at faults can be used not only for delay fault testing but also for detection of unmodeled faults. We have developed a hybrid BIST circuit; that is, a method consisting of a shift register with partial rotation and a procedure that selects test vectors from ATPG ones. This testing method can perform at-speed testing with high stuck-at fault coverage. During the at-speed testing, a subset of the ATPG vectors is input by using a low-speed tester. Computer simulations on ISCAS'85, ISCAS'89, and ITC'99 circuits are conducted for n = 1, 2, 3, 5, 10, and 15. The simulation results show that the amount of test vectors can be reduced to ranging from 52.3% to 0.9% in comparison with that of the ATPG vectors. As a result, the proposed method can reduce the cost of at-speed testing.

This paper addresses the single channel speech enhancement method which utilizes the mean value and variance of the logarithmic noise power spectra. An important issue for single channel speech enhancement algorithm is to determine the trade-off point for the spectral distortion and residual noise. Thus the accurate discrimination between speech spectral and noise components is required. The conventional methods determine the trade-off point using parameters obtained experimentally. As a result spectral discrimination is not adequate. And the enhanced speech is deteriorated by spectral distortion or residual noise. Therefore, a criteria to determine the point is necessary. The proposed method determines the trade-off point of spectral distortion and residual noise level by discrimination between speech spectral and noise components based on statistical criteria. The spectral discrimination is performed using hypothesis testing that utilizes means and variances of the logarithmic power spectra. The discriminated spectral components are divided into speech-dominant spectral components and noise-dominant ones. For the speech-dominant ones, spectral subtraction is performed to minimize the spectral distortion. For the noise-dominant ones, attenuation is performed to reduce the noise level. The performance of the method is confirmed in terms of waveform, spectrogram, noise reduction level and speech recognition task. As a result, the noise reduction level and speech recognition rate are improved so that the method reduces the musical noise effectively and improves the enhanced speech quality.

In this paper, we propose and evaluate a new channel assignment scheme for a micro-cellular network integrating data and conversational services. The channel assignment scheme combines handover processing depending on terminal speed with a preemptive scheme. High-speed terminals take over the channels of data terminals upon entering a full cell, while the data terminals are put in a queue until new resources are available. Simulating several variations of the scheme, allowing both fast moving data and voice terminals to preempt data terminals yielded the best result. Suspension time for fast moving data terminals was reduced dramatically, reducing the disadvantage caused by a high number of handovers. The cost was a small increase in blocking probability for new terminals.

One of the reasons of slow convergence in Backpropagation learning is the diminishing value of the derivative of the commonly used activation functions as the nodes approach extreme values, namely, 0 or 1. In this letter, we propose arctangent activation function to accelerate learning speed. Simulation results indicate considerable improvement in convergence performance.

This paper investigates the effect of fast cell selection (FCS) associated with fast packet scheduling methods and hybrid automatic repeat request (HARQ) with Chase combining, in which the optimum cell (or sector) transmitting a slot-assigned downlink shared channel (DSCH) is selected based on the received signal-to-interference power ratio (SIR), in high-speed downlink packet access (HSDPA). The Round robin (RR), Proportional fairness (PF) and Maximum carrier-to-interference power ratio (CIR) schedulers are used as the scheduling algorithm. The simulation results elucidate that although almost no additional diversity gain through FCS is obtained for the PF and Maximum CIR schedulers, the improvement in throughput by FCS coupled with the RR scheduler is achieved. Furthermore, we elucidate that the effect of FCS is small when only inter-sector FCS is performed; however, inter-cell FCS is effective in improving the radio link throughput for the access users with a lower received SIR near the cell edge. The radio link throughput at the cumulative distribution of 20% of soft handover users when both inter-sector and inter-cell FCS are performed is increased by approximately 20% and 60% for PF and RR schedulers, respectively, compared to that without FCS, i.e. with hard handover. We also show that when a traffic model such as the modified ETSI WWW browsing model is taken into account, the effect of FCS associated with the decreasing effect of fast packet scheduling is greater than that assuming continuous packet transmission. The user throughput at the cumulative distribution of 20% employing both inter-sector and inter-cell FCS is increased by approximately 60% compared to that without FCS.

A performance comparison is developed between a chaotic communication system and a spread spectrum system with similar features in terms of bandwidth and transceiver structure but based on more conventional Gold sequences. Comparison is made in the presence of noise and multipath contributions which degrade the channel quality. It is shown that, because of its more favourable correlation properties, the chaotic scheme exhibits lower error rates, at a parity of the bandwidth expansion factor. The same favourable correlation properties are also used to explain and show, through a numerical example, the benefits of chaotic segments in a multi-user environment.

The increase of test time of embedded DRAMs (e-DRAM) is one of the key issues of System-on-chip (SOC) device test. This paper proposes to put the repair analysis function on chip as Built In Self Repair (BISR). BISR is performed at 166 MHz as at-speed of e-DRAM with using low cost automatic test equipment (ATE). The area of the BISR is 1.7 mm2. Using error storage table form contributes to realize small area penalty of repair analysis function. e-DRAM function test time by BISR was about 20% less than the conventional method at wafer level testing. Moreover, representative samples are produced to confirm repair analysis ability. The results show that all of the samples are actually repaired by repair information generated by BISR.

This paper presents the transmission performance of a class-IV partial-response signaling SSB system and proposes a method that can improve its power efficiency. A line code that has no dc component has been used in the SSB transmission of digital signals. The type of line code, such as a partial-response signaling, increases the modulation states, and as a result, decreases the power efficiency. To overcome this obstacle, a new demodulation method called "re-filtering and combining" is proposed on the assumption of orthogonal phase detection. The demodulated quadrature channel is re-filtered by a Hilbert filter and is combined with the in-phase channel. It is confirmed by computer simulations that the new demodulation method improves the BER performance and a 3 dB improvement of the power efficiency is obtained.

A duration modeling technique is proposed for the HMM based connected digit recognizer. The proposed duration modeling technique uses a cumulative duration probability. The cumulative duration probability is defined as the partial sum of the duration probabilities which can be estimated from the training speech data. Two approaches of using it are presented. First, the cumulative duration probability is used as a weighting factor to the state transition probability of HMM. Second, it replaces the conventional state transition probability. In both approaches, the cumulative duration probability is combined directly to the Viterbi decoding procedure. A modified Viterbi decoding procedure is also presented. One of the advantages of the proposed duration modeling technique is that the cumulative duration probability rules the transitions of states and words at each frame. Therefore, an additional post-procedure is not required. The proposed technique was examined by recognition experiments on Korean connected digit. Experimental results showed that two approach achieved almost same performances and that the average recognition accuracy was enhanced from 83.60% to 93.12%.

This paper presents a detailed simulation method to estimate Doppler power spectrum and mean blood velocity using real CW Doppler transducers with twin-crystal arrangement. The method is based on dividing the sample volume into small cells and using the statistics of the Doppler power spectrum with the same Doppler shift frequency, which predicts the mean blood velocity. The acoustic fields of semicircular transducers across blood vessels were calculated and the effects of acoustical and physiological factors on Doppler power spectrum and mean blood velocity were analyzed. Results show that nonuniformity of the acoustic field of the ultrasonic beam in the blood vessel and blood velocity profiles significantly affect Doppler power spectrum and mean blood velocity. However, Doppler angle, vessel depth, and sample volume length are not sensitive functions. Comparisons between simulation and experimental results illustrated a good agreement for parabolic flow profile. These results will contribute to a better understanding of Doppler power spectrum and mean blood velocity in medical ultrasound diagnostics.