We consider transmission beamforming techniques for frequency-division-duplex (FDD) wireless communication systems using adaptive arrays to improve the signal quality of the array transmission link. We develop a simple effective transmission beamforming technique based on an approximated frequency tranlsation (AFT) to derive the tranmsiion beamforming weights from the uplink channel vector. This technique exploits the invariance of the short-time averaged fast fading statistics to small frequency translations. A simple approximate relationship that relates the transmission channel vector to the reception channel vector is derived. We have developed its practical alternative in which the frequency translation of the channel vector is performed at the principal angle of arrival (AOA) of the u;link synthestic angular spectrum instead of the mean AOA. To analyze the performance of the proposed methods, we consider the power loss incurred by applying the estimated channel vector instead of the true downlink channel vector. The performance is analyzed as a function of the mean AOA, the angular spread, the number of elements, frequncy difference between the uplink and the downlink, and the angle distribution. Their performance is also compared with that of the direct weight reuse method and the AOA based methods.

We propose a tunable add/drop filter in a form of an all-fiber mach-zehnder interferometer iwth one bragg grating at each arm. This device can be tuned by inducing a strain in the bragg grating. We also theoretically analyze the outut characteristics of the tunable add/drop filter. As a result of simulation, we know that the proposed tunable add/drop filter works well. When 2*10$^{-3}$ of strain is induced, the reflected spectrum shifts about 3nm. And its reflected spectral width is about 0.3nm. In this case roughly 5 channels can be tuned, assuming the channel spacing is 0.3nm. When the pathlengths of the both arms are not the same, the transmissivities at the add and output ports and the reflectivity at the tap port varies sinusoidally with the pthlength difference. To maintain the transmissivities above 90% in the wavelength tuning range of 20nm the pathlength difference less than 16.mu.m is required.

Relatively slow tuning speed of optical device causes the unwanted delay in high speed single-hop multiwavelength networks. To lessen the overhead, we present a channel access protocol with dynamjic message scheduling. Th eframe structure of proposed protocol adopts hybrid multiaccess scheme in which WDMA is used as a basic multiaccess technique and TDMA is used to provide subchannels within a wavelength band. This architecture has two merits : the network extention is not limited by available number of wavelengths, and the transmission delay caused by optical device tuning time can to minimize the number of tunings. It schedules messages that require same wavelength channels sequencely, so the total transmission delay is reduced by decreasement of wavelength changes. The performance of proposed protocol is evaluated through numerical analysis based on probability and queueing theory. The peformance of proposed protocol is evaluated through numerical analysis based on probability and queueing theory. The numeric results show that the peformance of proposed protocol is better than that of previous one.

In this paper, a an algorithm that detects the endocardial boundary, expanding the region from endocardial cavity using fuzzy inference, is proposed. This algorithm decides the ventricular cavity by fuzzy inference in process of searching each pixel from the inside of left ventricle in echocardial image and expands it. Uncertainty and fuzziness exists in decision of endocardial boundary. Therefore, we convert the lingustic representation of mean, standard deviation, and threshold value that are characteristic variables of endocardial boundary to fuzzy input and output variables. And, we extract proposed method is robuster to noise than radial searching method that is highly dependent on center position. To prove the similarity of detected boundary by fuzzy nference, we used the measures of SIZE, correlation coefficient, MSD, and RMSE and had acquired reasonable results.

To trnsmit moving image signals by using an object vased coding technique at the rate of 8kbps~11kbps, it is very important to minimize the bit rates used for the compression of the color information of MF-objects. This paper proposes a hybrid coding method which uses the shpae adaptive coding method and the interframe reference method selectively. Gilge's shape adaptive orthogonal coidng method is utilized for shpae adaptive coding. The interfarame reference method approximates the low-passed signals of the image by gilge's shpae adaptive orthogonalization method and then refers the approximation error signals from the high frequency signal components of th eprevious frame. The proposed method achives the bit rates reduction of 17% compared to the gilge's shape adaptive orthogonalization method and 30% rduction compared to the shape adaptive DCT in average.

For the degradation of severe noise and ill-conditioned blur the optimization function has the solution spaces which have many local optima around global solution. General restoration methods such as inverse filtering or gradient methods are mainly dependent on the properties of degradation model and tend to be isolated into a local optima because their convergences are determined in the convex space. Hence we introduce genetic algorithm as a searching method which will search solutions beyond the convex spaces including local solutins. In this paper we introudce improved evaluation square error) and fitness value for gray scaled images. Finally we also proposed the local fine tunign of window size and visit number for delicate searching mechanism in the vicinity of th global solution. Through the experiental results we verified the effectiveness of the proposed genetic operators and evaluation function on noise reduction over the conventional ones, as well as the improved performance of local fine tuning.

In this paper, we propose a method improving the accuracy of th eregistration coefficients calculated form two range images considering the measurement error. The employed range finder is based on triangulation and the depth measurement error is described with an error covariance matrix, which is based on thesensitivity of the range measurements. Experimental results demonstrate that the registration coefficients obtained with the proposed method are better than the results when the measurment errors are neglected.

In a drop-on-demand thermal ink-jet printer, the dot size of an ink droplet expelled from a printer depends on the absorption of the paper. This causes severe differences between output images on the different paper materials. In this paper, the color matching algorithm for different papers is proposed. To achieve corresponding color reproduction, dot gain compensatio nbased on saturation was applied to predict color reproduction on a printer. If the dot gain of pigment increases, the white portion of paper decreases while the saturation value increases monotonically. As the results of dot gain compensation, intensity change may appear. Therefore, an intensity compensation without any hue variation is followed to match the colors of different subtrates.

With recent development of speech recognition technology and multimedia computer systems, more potential applications of voice will become a reality. In this paper, we implement speech interface on the windows95 environment for practical use fo multimedia computers with voice. Speech interface is made up of three modules, that is, speech input and detection module, speech recognition module, and application module. The speech input and etection module handles th elow-level audio service of win32 API to input speech data on real time. The recognition module processes the incoming speech data, and then recognizes the spoken command. DTW pattern matching method is used for speech recognition. The application module executes the voice command properly on PC. Each module of the speech interface is designed and examined on windows95 environments. Implemented speech interface and experimental results are explained and discussed.

It is necesary that analog buffer circuit should drive an external load in the VLSI design such as switched capacitor efilter (SCF), D/A converter, A/d converter, telecommunicatin circuit, etc. The conventional CMOS buffer circuit have many probvlems according as CMOS technique. Firstly, Capacity of large load ar enot able to opeate well. The problem can be solve to use class AB stages. But large load are operated a difficult, because an element of existing CMOS has a quadratic functional relation with inptu and outut voltage versus output current. Secondly, whole circuit of dynamic rang edecrease, because a range of inpt and output voltages go down according as increasing of intergration rate drop supply voltage. In this paper suggests that new differential CMOS line driver make out of operating an external of large load. In telecommunication's chip case transmission line could be a load. It is necessary that a load operate line driver. The proposal circuit is planned to hav ea high generation power rnage of voltage with preservin linearity. And circuit of capability is inspected through simulation program (HSPICE).