• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.5C
• /
• pp.317-325
• /
• 2011

For OFDM (Orthogonal Frequency Division Multiplexing) systems, 2-D MMSE (2-Dimensional Minimum Mean Square Error) channel estimation provides optimal performance in frequency/time selective fading channel environment. However, the 2-D MMSE channel estimation has high computational complexity due to the large matrix size, because the 2-D MMSE channel estimation considers time as well as frequency axis for channel estimation. To reduce the computational complexity, we propose a modified 2-D MMSE channel estimator which is based on 1-D MMSE channel estimation with weighted sum. Furthermore, we consider RMS delay spread and Doppler frequency estimation for 2-D MMSE channel estimation. We show that the proposed method can significantly reduce computational complexity as well as that it can perform close to 2-D MMSE channel estimation.

• Journal of Korean Society for Quality Management
• /
• v.49 no.3
• /
• pp.245-254
• /
• 2021

Purpose: In order to accurately reach an underwater projectile to a target point, reliable INS and accurate arrangement of INS between master and slave INS is paramount. Unlike terrestrial and aerial environments, underwater projectile will operates in a restricted environment where location information cannot be received or sent through satellites. In this report, we review the factors affecting the transfer alignment of master and slave INS, as well as how to improve the positional error between INS through improved transfer alignment algorithms. Methods: In this work, we propose an improvement algorithm and verify it through simulation and driving test. The simulation confirmed the difference in the transfer alignment azimuth by fitting the MINS and SINS indoors, displacement in posture, and the process of transfer alignment between MINS and SINS through a driving test to confirm algorithm can improve the arrangement. Results: According to this study, reason for the error in the transfer alignment between MINS/SINS is the factors of the system where movements such as roll, pitch, yaw are not inter locked in real time due to the delay in transmit/receive system. And confirm that the improved algorithm has a desirable effect on accuracy. Conclusion: Through this work, it is possible to identify ways to improve the accuracy of underwater projectiles to reach their target points under various underwater environments and launch condition.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1996.06a
• /
• pp.55-59
• /
• 1996

In this paper, we first analyze the influence of interference due to the time variation and delay spread of the mobile channel on an orthogonal frequency division multiplexing (OFDM) system. With the result, we obtain the bit error rate performance of the 16-QAM OFDM system. Second, we investigate the performance of the Reed-Solomon (RS) coded 16-QAM OFDM system when the number of subcarriers varies. In the investigation, we assume that the information transmission rate and the total bandwidth expansion due to coding, guard interval, and the number of subcarriers are fixed. Under this condition, it is observed that there are optimum numbers of subcarriers that minimize the post decoding symbol error probability of RS code for various channel states.

• Journal of KSNVE
• /
• v.8 no.6
• /
• pp.1037-1042
• /
• 1998

This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. We also Propose a novel control algorithm (AFC/CAFC) for tonal noise cancellation. The proposed algorithm estimates the magnitude and phase of the tonal noise. The algorithm uses the steepest descent method for the phase/magnitude estimation. Performances of tile CAFC algorithm are presented in comparison with those by the AFC algorithm based on computer simulations and experiments.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.5 no.6
• /
• pp.1089-1093
• /
• 2001

A H$\infty$ control theory is applied to the design problem of synthesis filters in a mutirate filter bank. We select a desired pure time-delay system as reference model, and then consider the error system between the mutirate filter bank and the reference model. 1'he synthesis filters minimize the ι$_2$-induced norm of the error system.

• Journal of the Korea Society of Computer and Information
• /
• v.9 no.2
• /
• pp.89-96
• /
• 2004

In this paper, an improved method is proposed by supplementing multi-path gain for detection. This method is proposed to complement the shortcomings of the conventional detection method which is used for multiuser detection in STBC(Space-Time Block Code) CDMA system. We analyzed the improved method in bit error probability viewpoint and compared the result with that of the conventional method. From this result, we showed that bit error probability of the improved method is superior to that of the conventional method when the parameters such as delay, number of user and SNR are increased.

• Journal of Astronomy and Space Sciences
• /
• v.9 no.1
• /
• pp.52-68
• /
• 1992

The ionosphere in accordance with solar activity can affect the transmission of radio waves. The effect of the ionosphere on the radio wave propagation are scattering of radio waves, attenuation, angle error, ranging error, and time delay. The present study is based on the Korean ionospheirc data obtained at the AnYang Radio Research Laboratory from January 1985 through October 1989. The data are analyzed to show the daily and the annual variations of the ionosphere. The data are also used to simulate the density distribution of the Korean ionosphere following the Chapman law.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2001.10a
• /
• pp.275-278
• /
• 2001

A H$_{\infty}$ control theory is applied to the design problem of synthesis filters in a mutirate filter bank. We select a desired pure time-delay system as reference model, and then consider the error system between the mutilate filter bank and the reference model. The synthesis filters minimize the ι$_2$-induced norm of the error system.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.20 no.10
• /
• pp.1061-1070
• /
• 2009

In this paper, we have designed TDL(True-time Delay Line) for eliminating beam-squint occurring in active phased array antenna with large electrical size operated in wide bandwidth, and have tested its electrical performance. The proposed TDL device is composed of 4-bit microstrip delay line structure and MMIC amplifier for compensation of the delay-line loss. The measured results of gain and phase versus delay state satisfy the electrical requirements, also P1dB output power and noise figure meet the requirement. To verify the performance of fabricated TDL, we have simulated the beam patterns of wide-band active phased array antenna using the measured results and have certified the beam pattern compensation performance. As a result of simulated beam pattern compensation with respect to the 675.8 mm size antenna which is operated in X-band, 800 MHz bandwidth, we have reduced the beam squint error of ${\pm}1^{\circ}$ with ${\pm}0.1^{\circ}$. So this TDL module is able to be applied to active phase array antenna system.

• The Journal of the Acoustical Society of Korea
• /
• v.17 no.2E
• /
• pp.38-46
• /
• 1998

In this paper, we propose a remote speech input device, a new method of user-friendly speech input in spontaneous speech recognition system. We focus the user friendliness on hands-free and microphone independence in speech recognition applications. Our method adopts two algorithms, the automatic speech detection and the microphone array delay-and-sum beamforming (DSBF)-based speech enhancement. The automatic speech detection algorithm is composed of two stages; the detection of speech and nonspeech using the pitch information for the detected speech portion candidate. The DSBF algorithm adopts the time domain cross-correlation method as its time delay estimation. In the performance evaluation, the speech detection algorithm shows within-200 ms start point accuracy of 93%, 99% under 15dB, 20dB, and 25dB signal-to-noise ratio (SNR) environments, respectively and those for the end point are 72%, 89%, and 93% for the corresponding environments, respectively. The classification of speech and nonspeech for the start point detected region of input signal is performed by the pitch information-base method. The percentages of correct classification for speech and nonspeech input are 99% and 90%, respectively. The eight microphone array-based speech enhancement using the DSBF algorithm shows the maximum SNR gaing of 6dB over a single microphone and the error reductin of more than 15% in the spontaneous speech recognition domain.