• Journal of the Korean Society of Manufacturing Technology Engineers
• /
• v.5 no.4
• /
• pp.26-37
• /
• 1996

This paper presents a new approach to the design of adaptive control system using DSPs(TMS320C30) for robotic manipulators to achieve trajectory tracking by the joint angles Digital signal processors are used in implementing real time adaptive control algorithms to provide an enhanced motion control for robotic manipulators. In the proposed control scheme adaptation laws are derived from the improved Lyapunov second stability analysis method based on the adaptive model reference control theory. The adaptive controller consists of an adaptive feedforward controller. feedback controller. and PID type time-varying auxiliary control elements. The proposed adaptive control scheme is simple in structure, fast in computation, and suitable for implementation of real-time control. Moreover, this scheme does not require a an accurate dynamic modeling, nor values of manipulator parameters and payload. Performance of the adaptive controller is illustrated by simulation and experimental results for a SCARA robot.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.25 no.12
• /
• pp.888-894
• /
• 2015

A new method for the simulation of the vehicle's interior road noise is proposed in the present study. The road noise model can synthesize road noise of a vehicle for varying driving speed within a range. In the proposed method, interior road noise is considered as a stochastic time-series, and is modeled by a nonstationary parametric model via two steps. First, each interior road noise signal, obtained from constant speed driving tests performed within a range of speed, is modeled as an autoregressive model whose parameters are estimated by using a standard method. Finally, the parameters obtained for different driving speeds are interpolated based on the varying driving speed to yield a time-varying autoregressive model. To model a full band road noise, audible frequency range is divided into an octave band using a wavelet filter bank, and the road noise in each octave band is modeled.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.8A
• /
• pp.664-670
• /
• 2005

OFDM is a very attractive technique for achieving high-bit-rate data transmission and high spectrum efficiency in fading environment. However, the reliable detection of an OFDM signal in time-varying multipath fading channels is a challenging problem. Accordingly, various channel estimation methods have been proposed for performance improvement. But, conventional pilot patterns for channel estimation in OFDM systems have not robust characteristics relating to various mobile speed. To solve this drawback in conventional patterns, we propose the pilot patterns modified from conventional patterns to have a good error performance in time-varying fading channel. Simulation results show that the performance of the proposed pilot patterns is better than conventional patterns in fast time-varying channel.

• Journal of Institute of Control, Robotics and Systems
• /
• v.20 no.1
• /
• pp.1-5
• /
• 2014

In this paper, we propose an optimal fixed-lag FIR (Finite-Impulse-Response) smoother for a class of discrete time-varying state-space signal models. The proposed fixed-lag FIR smoother is linear with respect to inputs and outputs on the recent finite horizon and estimates the delayed state so that the variance of the estimation error is minimized with the unbiased constraint. Since the proposed smoother is derived with system inputs, it can be adapted to feedback control system. Additionally, the proposed smoother can give more general solution than the optimal FIR filter, because it reduced to the optimal FIR filter by setting the fixed-lag size as zero. A numerical example is presented to illustrate the performance of the proposed smoother by comparing with an optimal FIR filter and a conventional fixed-lag Kalman smoother.

• The Journal of the Acoustical Society of Korea
• /
• v.16 no.4
• /
• pp.5-10
• /
• 1997

In this paper, a new time delay estimation algorithm, WTD-LMSTDE was proposed. This method has great improvement in convergence rate relative to the time domain approach by decreasing the eigen value spread of input signal autocorrelation matrix. The performance of the algorithm was evaluated for the cases of time invariant time delay and time varying time delay. In the case of time invariant time delay, the estimation accuracy of WTD-LMSTDE was better than that of LMSTDE from 3.3% to 12.5% with respect to SNR. In the case of time varying time delay, the mean error power of WTD-LMSTDE in linear increased delay environment was decreased about 2.4dB compared to that of LMSTDE under noise-free condition. As a result, we showed that the performance of WTD-LMSTDE is better than of LMSTDE.

• Journal of the Institute of Convergence Signal Processing
• /
• v.12 no.1
• /
• pp.44-49
• /
• 2011

The need of data transmission in oceans and other underwater mediums are increasing day by day, so as the research. The underwater medium is very different from that of air. Propagation of electromagnetic wave in water or underground is very difficult because of the conductivity of the propagation materials. In this case, we usually use acoustic signals as ultrasonic but, they are not easy to transfer long distance with coherent method because of time varying multipaths, Doppler effects and attenuations. So, we use non-coherent methods such as FSK or ASK to communicate between long distances. But, as the propagation speed of acoustic wave is very slow, BW of the channel is narrow. It is very hard to guaranty the enough speed for the transmission of digital image data. In previous studies, we proposed this data communication protocol theoretically. In this paper, an underwater channel is modeled and this protocol is tested in this channel condition. The results show that the protocol is 4-6 times faster than ASK. Some relations and results are shown depending on the data length, channel length, bit rate etc.

• Journal of the Korean Institute of Telematics and Electronics C
• /
• v.34C no.9
• /
• pp.33-42
• /
• 1997

Fixed-point optimization utility software that can aid scaling and wordlength determination of digital signal processign algorithms written in C or C$\^$++/ language is developed. This utility consists of two programs: the range estimator and the fixed-point simulator. The former estimates the ranges of floating-point variables for automatic scaling purpose, and the latter translates floating-point programs into fixed-point equivalents for evaluating te fixed-point performance by simulation. By exploiting the operator overloading characteristics of C$\^$++/ language, the range estimation and the fixed-point simulation can be conducted just by modifying the variable declaration of the original program. This utility is easily applicable to nearly all types of digital signal processing programs including non-linear, time-varying, multi-rate, and multi-dimensional signal processing algorithms. In addition, this software can be used for comparing the fixed-point characteristics of different implementation architectures.

• Journal of electromagnetic engineering and science
• /
• v.16 no.1
• /
• pp.1-6
• /
• 2016

Computer simulations were conducted to demonstrate the generation of microwave-induced thermoacoustic signal. The simulations began with modelling an object with a biological tissue characteristic and irradiating it with a microwave pulse. The time-varying heating function data at every particular point on the illuminated object were obtained from absorbed electric field data from the simulation result. The thermoacoustic signal received at a point transducer at a particular distance from the object was generated by applying heating function data to the thermoacoustic equation. These simulations can be used as a foundation for understanding how thermoacoustic signal is generated and can be applied as a basis for thermoacoustic imaging simulations and experiments in future research.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2001.11b
• /
• pp.1044-1049
• /
• 2001

Classical deconvolution methods for source identification following linear filtering can only be used if the transfer function of the system is known. For many practical situations, however, this information is not accessible and/or is time varying. The problem addressed here is that of reconstruction of the original input from only the measured signal. This is known as 'blind deconvolution'. By using Higher Order Statistics (HOS), the restoration of the input signal is established through the maximisation of higher order moments (cumulants) with respect to the characteristics of the signals concerned. This restoration is achieved by constructing an inverse filter considering the choice of the initial inverse filter type. As a practical application, an experimental verification is carried out for the restoration of our impacting signal arising in the response of a cantilever beam with an end stop when randomly excited.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.1E
• /
• pp.7-13
• /
• 2001

In this paper we propose a new subspace tracking algorithm based on the PASTd (Projection Approximation Subspace Tracking with deflation). The algorithm is obtained via introducing the variable forgetting factor which adapts itself to the time-varying subspace environments. The tracking capability of the proposed algorithm is demonstrated by computer simulations in an abruptly changing DOA scenario. The estimation results of the variable forgetting factor PASTd(VFF-PASTd) outperform those of the PASTd in the nonstationary case as well as in the stationary case.