In this paper are presented kinematic and dynamic modeling and path-tracking of fourwhelled mobile robot with 2 d.o.f. having the limited drivetorques. Controllability of wheeled-mobile robot is revealed by using the kinematic model. Instantaneously coincident coordinate cystem, force/torques generated by inverse dynamics exceed the limitation, we make wheeled-mobile robot follow the reference path by modifying the planned reference trajectory with time-scaling. The controller is introduced to compensate for error owing to modeling uncertainty and measurement noise. And simulation results prove that method proposed by this paper is efficient.

This paper proposes a deadzone estimator for robust system identification. In order to cope with the drift phenomena occurred in where system inputs are not sufficiently excited in adaptive control, we introduce a novel and flexible bound condition against a fixed constant. It is derived from a forgetting factor and a rational value of the traces of the covariance matrices between step k and k-1. The key feature of this is that it does not require a priori for the bound. Also, the calculation of it is more simple than the one of literatures. The simulation results are examined for showing the practical performance of this algorithm.

A submerged vehicle which is a nonlinear multivariable system must be designed to be roubst against inner-outer perturbations and hydrodynamic disturbances induces maneuvering operation. But a practical design of motion controller is limited by both mathematical modeling error and linearization errors. Performance of a motion controller based on traditional design method is very poor when the vehicle motion is under wave force distrubacnes near sea surface. Therefore, this ppaer proposes a design method of $^{\infty}$ controller under model uncertainty and sea wave disturbances. performance of the controllers by both computer simulation and HILS (hardwave in the loop simulation) shows that $H^{\infty}$ controller is more robust than PID controller under model uncertainty and high sea state...

This paper proposed a kinematic modeling methodlogy and feedback control system based on kinematics for 2 degrees of freedom of 4-wheeled mobile robot. We assigned coordinate systems to specify the transformation matirx and write the kinematic equation of motion. We derived the actuated inverse and sensed forwared solution for the calculation of actual robot orientation and the desired robot orientation. It is the most significant error and has the largest impact on the motion accuracy. To calculate the WMR position in real time, we introduced the dead-reckoning algorithm and composed two feedback control system that is based on kinematics. Through the simulation result, we compare with the ffedback control system for position control.

This paper presents a fuzzy data fusion scheme which can analyze the sensor condition, the strength and location of a force applied to a test material. These can be realized by the modelling and fusioning of sensor signals and sensor properties. The technique uses, as the inference variables, relative magnitude of data (RMD), absolute magnitude of data (AMD) initial state (IS), synchronized relational function (SRF) and asynchronized relational function (ARF). To show the usefulness of this scheme, an experiment on the cantilever bar and six strain gages is carried out. The location of the force is inferred from SRF and ARF and the strength from RMD and AMD. In particular, the strength is compared with the measurement data of the force sensor.

In this paper, a new continuous-time current-mode integrator as basic building block of the low-voltage analog current-mode active filters was proposed. Compared to the current-mode integrator which was proposed by Zele, the proposed current-mode integrator had higher unity gain frequency and output impedance in addition to lower power dissipation. And also, a current-mode third-order lowpass active filter was designed with the proposed current-mode integrator. The designed circuits were fabricated using the ORBIT's 1.2.mu.m double-poly double-metal CMOS n-well process. The experimental resutls of the active filter designed and fabricated for this research have shown that it has the performance of 44.5MHz cutoff frequency, 3.3mW power dissipation and the third-order active filter area was 0.12mm$^{2}$.

In this paper, a variable length block code for real time compression and decompression of hanguel and chinese character bit map font is proposed. The proposed code shows a good compression ratio in complete form of hangeul myoungjo and godik style and chinese batang and doddum style bit map font. Besides, a compression and decompression ASIC is designed and simulated on CAD. The 0.8 micron CMOS sea of gate is used to implement the ASIC in amount of 5,200 gates, and it runs at simple hardware and compress and decompress at 33M bit/sec at maximum, which is ideal for real time applications.

In this paper, variable windowing mean filter to remove speckle noise and a measure to detect thin edge in ultrasonic images are proposed. Because ultrasonic images are corrupted by speckle noise showing a granular appearance, good edge detection is difficult. As a result, noise removing filter is needed in preprocessing stage. The speckle noise removing filter is based on mean filter whose window size is changed by the ratio of standard deviation to mean for image signal and noise signal in local area. And the measure expressed the difference of means between tow windows is used for detecting thin edge in filtered image. Results show that variable windowing mean filter removes speckle noise effectively, and proposed measure is useful in detecting thin edge.

In this paper, we implemented a computerized pulse diagnosis system for the developement of pulse diagnosis algorithms and the generalization of the pulse diagnosis. The system consists of the hardware and software. The hardware detects pulse waves and inputs the waves into the computer system, while the software not only manages and analyzes the input pulse wave data but also privides the database. In order to clinically test the developed pulse diagnosis system, we applied the carotid-radial pulse diagnosis algorithm to the system. As the results, it is found that the genralizatio of the pulse diagnosis is possible.

A hybrid approach employing both the genetic algorithm and the newton-raphson method is proposed for the electrical impedance computed tomography (EICT) image reconstruction. Computational experiments based on the new concept have shown promising results for several noise-free models. In particular, the resistance distribution of the tested models having resistivity ratio up to 100:1 has been reconstructed sucessfully. Using the proposed mehtod, it is also possible to get the reconstruction by the conventional iterative approaches be difficult to vonverge to a robust solution. If the compution power is enhanced further, the proposed method is expected to stimulate the practical applications of the EICT technology in the near future.

Color image quantization is a process of selecting a set of colors to display an image with some representative colors without noticeable perceived difference. It is very important in many applications to display a true color image in a low cost color monitor or printer. the basic problem is how to display 224 colors with 256 or less colors, called color palette. In this paper, we propose an algorithm to design the 256 or less size color palette by using spatial maskin geffect of HVS and subjective distortion measure weighted by color palette by using spatial masking effect of HVS and subjective distortion measure weighted by color activity in 4*4 local region in any color image. The proposed algorithm consists of octal prequantization and subdivision quantization processing step using the distortion measure and modified Otsu's between class variance maximization method. The experimental results show that the proposed algorithm has higher visual quality and needs less consuming time than conventional algorithms.

In this paper, a new edge-based segmentation algorithm for range image using pseudo reflectance images (PRIs) is proposed. A model of pseudo reflectance which is useful in analyzing three dimensional scene and objects is introduced and then three PRIs are generated by the model. For generating three PRIs, bels and jain's differential window operator is selected and three different light source directions are determined. Three edge images are extracted from each PRI and a fused (logical ORing) edge image is constructed for the benefit of enhanced edge formation. The final segmentation results of the proposed algoritm are obtained after the processing of thinning, labeling and correcting erroeneous regions with the fused edge image. The good performance of edge detection and segmentation is confirmed via computer simulation with synthetic and real range images.

A compensation method of color error resulting from dynamic range transformation of color image is propsoed. The color error is measured using the CIE L a b uniform color space, and the color image is compensated to minimize it. The color error is significant in the dark region of image, which is caused by the lunimance scaling factor is relatively large in that case. In the proposed method, we weight the luminance scaling factor corresponding to luminnce to minimize the error. Because the weighting factor depends on luminace distribution of image, a decisio method of weighting factor using histogram is also proposed.

In this paper, we propose the method that extracts the speech feature using the hearing model through signal precessing techniques. The proposed method includes following procedure ; normalization of the short-time speech block by its maximum value, multi-resolution analysis using the discrete wavelet transformation and re-synthesize using thediscrete inverse wavelet transformation, differentiation after analysis and synthesis, full wave rectification and integration. In order to verify the performance of the proposed speech feature in the speech recognition task, korean digita recognition experiments were carried out using both the dTW and the VQ-HMM. The results showed that, in case of using dTW, the recognition rates were 99.79% and 90.33% for speaker-dependent and speaker-independent task respectively and, in case of using VQ-HMM, the rate were 96.5% and 81.5% respectively. And it indicates that the proposed speech feature has the potentials to use as a simple and efficient feature for recognition task.

In this paper an improved morphological algorithm for directional defect detection is proposed, where the defect is parallel to the texture image. The algorithm is based on obtaining the background image while removing the defect by comparing every directional morphological result with max or min except that of defect. The defect can of defect and the background image. For a computer simulation, it is shown that the proposed method has better performance than the conventional algorithm.

A loop structure optimization algorithm is proposed for generting a set of efficient codes for loop structure in order to optimize RISC compiler codes. Since there are so many loop structure in the program, most of the execution time is used to process looping codes. Thus, reduction of loop instructions is more effective than optimizing codes outside the loop. The proposed algorithm presents a method to combine several different loops into a simple loop. Therefore, rather than executing each loop independently, loops in the program are serached, analyzed, and finally created some relative informtion such as dependency and range. In doing so, the loops in the program can efficiently be recombined and restructured. As a result, the overall execution time for the program of the sequential programming language is reduced.

In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

This paper presents an alternative technique for DOA (direction-of-arrival) estimation. For generating a weight vector orthogonal to the signal subspace, a modified version of MES (minimum eigenvalue searching ) method is introduced. The performance of the proposed technique is compared to that of the conventional ED (eigen decomposition) method in terms of angle resolution for a number of snapshots during agiven observation period as well as various SNR's. In addition, the superiority of the suggested technique is shown, by analyzing the required computational load of the proposed MES and conventional ED method. A novel procedure of simplifying the MES proposed in [1] is presented on that purpose. Another advnatage of the proposed technique is that it is performed independently of the detection of the number of signal components, which makes it possible to estimate the DOA's of clusters consisting of infinite number of inseparable signal components.

In continuous speech recognition system, it is possible to implement the system which can handle unlimited number of words by using limited number of phonetic units such as phonemes. Dividing continuous speech into the string of tems of phonemes prior to recognition process can lower the complexity of the system. But because of the coarticulations between neiboring phonemes, it is very difficult ot extract exactly their boundaries. In this paper, we propose the algorithm ot extract short terms which can represent each phonemes instead of extracting their boundaries. The short terms of lower spectral change and higher spectral chang eare detcted. Then phoneme changes are detected using distance measure with this lower spectral change terms, and hgher spectral change terms are regarded as transition terms or short phoneme terms. Finally lower spectral change terms and the mid-term of higher spectral change terms are regarded s the represent each phonemes. The cepstral coefficients and weighted cepstral distance are used for speech feature and measuring the distance because of less computational complexity, and the speech data used in this experimetn was recoreded at silent and ordinary in-dorr environment. Through the experimental results, the proposed algorithm showed higher performance with less computational complexity comparing with the conventional segmetnation algorithms and it can be applied usefully in phoneme-based continuous speech recognition.

The purpose of this paper is to show that the solutions of the problem for the anaphor ocured in korean senstence, by means of one-direction activated chart parsing leaded by a head. This is the phenomenon frequently occured in the conversation of natural language and the part necessarily required in the construction of natural language processing system for the practical use. To solve the problem of anaphor in the korean language, we have computerized definition and the management conditions necessary in the semantic classification between the anaphor and its antecedent and index are added in the feature structure in lexicon. To deal with anaphor in parser and algorithm is proposed to solve the problem for anaphor. The range of management of pareser is extended to solve the problem for anaphor of the indeclinable parts of speech in korean occured in all the sentences the parser HPSG developed previously manages.

This paper presetns a new fuzzy neural network for fuzzy modeling.The fuzzy neural network is composed of 4 layers and then odes of each layer represent the each step of the if-then fuzzy inference. A heuristic based on the back-propagation algorithm is proposed to ajdust the parameters of the fuzzy nerual network. We prove the feasibility of the network using the experiments on modeling a nonlinear mathematical system and the comparison with previous research.

This paper proposes an efficient method for improving the training performance of the neural network using a hybrid of backpropagation algorithm and dynamic tunneling system.The backpropagation algorithm, which is the fast gradient descent method, is applied for high-speed optimization. The dynamic tunneling system, which is the deterministic method iwth a tunneling phenomenone, is applied for blobal optimization. Converging to the local minima by using the backpropagation algorithm, the approximate initial point for escaping the local minima is estimated by the pattern classification, and the simulation results show that the performance of proposed method is superior th that of backpropagation algorithm with randomized initial point settings.

In this paper, we propose a learning method of multi-layer perceptrons (MLPs) with 8-bit data precision. The suggested method uses the cross-entropy cost function to remove the slope term of error signal in output layer. To decrease the possibility of overflows, we use 16-bit weighted sum results into the 8-bit data with appropriate range. In the forwared propagation, the range for bit-conversion is determined using the saturation property of sigmoid function. In the backwared propagation, the range for bit-conversion is derived using the probability density function of back-propagated signal. In a simulation study to classify hadwritten digits in the CEDAR database, our method shows similar generalization performance to the error back-propagation learning with 16-bit precision.

This paper describes a high power resonant inverter for diagnostic X-ray generators using zero-voltage soft-switching technology. The system consists of a step-down chopper, a resonant phase-shift PWM inverter, a hihg-voltage diode, and high voltage cables a smoothing DC capacitor. The inverter makes use the leakage inductance of the hihg-voltage transformer and external capacitor as resonant components. The rectified input voltage is controlled by a step-down chopper with input voltage compensator. The output regualtion is obtained by a resonant phase-shift PWM inverter with the digital feedback controller using DSP (digital signal processor), resulting in fast rising time and wide output voltage variation. The theoretical results are correlated with results from an experimental prototype of a 7-kVp, 300mA (21kW).