• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.6
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• pp.1167-1173
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• 2005

In this paper, the use of an antenna array is considered for code timing acquisition of DS-CDMA signals. The probabilities of acquisition are evaluated by applying multiple narrow fixed-beams to the conventional MUSIC acquisition approach in the multiuser environment on the time-varying Rayleigh fading channel. Each fixed-beam for spatial filtering is dedicated to an individual angular sector that is formed by dividing the entire angular domain by the number of antenna elements. The fixed-beams with a capability of interference suppression provide the additional degrees of freedom,. Hence, the multibeam-based MUSIC estimator can be used to synchronize to more users than the conventional MUSIC algorithm for one antenna. The multibeam-based subspace method is evaluated to significantly improve the performance of a single antenna based MUSIC technique in multiuser scenarios.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.2
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• pp.218-230
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• 1993

A power line modem(PLM) which transfers data safely through power lines in houses or small offices is considered. When a power line is used for communications, transmitted signals could be affected by the channel characteristics such as frequency-selective fading, interference, and time-varying attenuation. In order to overcome these impairments, a direct sequence(DS) technique which is well known as an effective instrument against a variety of interferences and hostile channel properties is employed. Using a DS technique, however, requires more circuits such as PN code generator circuits, code modification circuits, and complicated synchronization circuits, and it also results in substantial acquisition delay. In this paper, some of these circuits are implemented via software programmed in the system controller, and the complicated synchronization circuits are replaced by simple circuits utilizing a 60 Hz power signal for synchronization. The synchronization ciruits used in this paper virtually eliminate the substantial acquisition delay, and is also designed to free influence of 60 Hz zero crossing jitters which reside in a power signal. As a result, a PLM using a DS technique is realized in the form of wall-socket plug, and the PLM hardware would be very much simplified.

• Journal of Digital Contents Society
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• v.8 no.3
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• pp.355-363
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• 2007

One of major factors for learning achievement is the student's learning preference according to his character type. In course of learning, if a student studies e-learning contents opposed to his preference, then he would be under stress and his blood pressure and heart beat be changed. For measuring unwillingness, we used spectral components in frequency domain known as stress measure. For 13 children attending kindergarten we examined S(sensing)/ N(intuition) of MBTI and presented same learning contents during 10 minutes. During learning we gathered ECG signals, changed into HRV(heart rate variability), transformed time-varying HRV signal into spectral density in frequency domain. And then, we divided it into three areas of low(LF), middle(MF), and high-frequency(HF) and calculated stress measures by rates of those frequency area. We compared estimated stress measures of S group with them of N group whether students in different group preferred different contents or not. Experimental shows that students according to MBTI type prefer different contents.

• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.6
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• pp.628-633
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• 2014

This paper proposes a performance improvement method of the conventional double talk detection method which can operate before convergence of the echo canceller. The proposed method estimates the coefficients of the linear predictive coding(LPC) filter by using the primary input signal. The time-varying threshold for double talk detection is determined based on the LPC filter gain of the primary input signal level. The proposed method can reduce not only false detection rate which means wrong detection of single talk as double talk but also double talk detection delay. Computer simulation was performed using a long-term real speech signals. It is shown that the proposed method improves the conventional method in terms of lowering the false detection rate and shortening the detection delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.73-81
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• 2005

We propose an automatic modulation recognition scheme which extracts pre-defined key features from the received signal and then applies equal gain combining method to determine the used modulation. Moreover, we compare and analyze the performance of the proposed algorithm with that of decision-theoretic algorithm. Our scheme extracts five pre-defined key features from each data segment, a data unit for the key feature extraction, which are then averaged over all the segments to recognize the modulation according to the decision procedure. We check the performance of the proposed algorithm through computer simulations for analog modulations such as AM, FM, SSB and for digital modulations such as FSK2, FSK4, PSK2, and PSK4, by measuring recognition success rate varying SNR and data collection time. The result shows that the performance of the proposed scheme is comparable to that of the decision-theoretic algorithm with less complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.1
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• pp.1-8
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• 2015

In this paper, a double-input single-output architecture of a LNA(Low Noise Amplifier) is presented to enable to be devised for light weight and small-sized OTM(On-The-Move) satellite communication system suitable to be mounted on vehicles. In spite of advantages of the double-input single-output architecture of a LNA such as reduction of the number of physical channels, it results in time-varying phase error between a fundamental mode path and a high-order mode path. This paper shows that the error can be corrected by adding pilot signals to the LNA and using signal processing, and also gives the measurement data to use the method mentioned above.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1845-1851
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• 2007

Background noise and room reverberation are two causes of degradation in speech in listening situations. Many algorithms developed to enhance reverberant speech. In this paper we propose a dereverberation method for enhancement of speech using modified the linear prediction(LP) residual in reverberant room condition. The proposed dereberberation method based on the fact that the signification excitation of the vocal tract system takes place at the instant of glottal closure in voiced speech. Our method used delay information form each sensor, and we need reverberant signals from 3 sensors. We obtain a new LP residual signal using modified IP residual combination which derived form weighting of the LP residual and the Hilbert transform of LP residual. The nature of the coherently added Hilbert envelop has several large amplitude spikes because of the effects of noise and reverberation. This residual of the clean speech is used to excite the time-varying all-pole filter to obtain the enhanced speech. We achieved simulation of proposed algorithm for performance analysis in reverberation environment. The proposed algorithm improves substantially the quality of reverberant speech.

• Journal of Magnetics
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• v.21 no.4
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• pp.509-515
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• 2016

We report on development of a ground-based bi-axial Search-Coil Magnetometer (SCM) designed to measure time-varying magnetic fields associated with magnetosphere-ionosphere coupling processes. The instrument provides two-axis magnetic field wave vector data in the Ultra Low Frequency or ULF (1 mHz to 5 Hz) range. ULF waves are well known to play an important role in energy transport and loss in geospace. The SCM will primarily be used to observe generation and propagation of the subclass of ULF waves. The analog signals produced by the search-coil magnetic sensors are amplified and filtered over a specified frequency range via electronics. Data acquisition system digitizes data at 10 samples/s rate with 16-bit resolution. Test results show that the resolution of the magnetometer reaches $0.1pT/{\sqrt{Hz}}$ at 1 Hz, and demonstrate its satisfactory performance, detecting geomagnetic pulsations. This instrument is scheduled to be installed at the Korean Antarctic station, Jang Bogo, in the austral summer 2016-2017.

• Journal of IKEEE
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• v.24 no.4
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• pp.1081-1085
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• 2020

Instantaneous frequencies (IF) of the AM-FM signal is estimated based on the inflection point detection (IPD) method. Local maxima/minima are detected using the IPD, and they are exploited to find the IF of AM and FM components, respectively. The envelope of the maxima/minima is obtained to estimate the IF of the AM part. And the distance between neighboring maxima (or minima) is used to estimate the IF of the FM component. Computer simulation shows that the proposed method properly estimates the IF of the AM and FM when the signal has fixed frequencies for both parts. In the case of the time-varying IF of the FM part, the estimated IF shows some deviation from the true IF due to the rough sampling effect of the maximum/minimum points. Thus, the post-processing such as the lowpass filtering of the estimated IF is required to refine the resulting IF estimation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.18 no.6
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• pp.1659-1674
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• 2024

Conventional maritime communication struggles to provide high data rate services for Internet of Things (IoT) devices due to the variability of maritime environments, making it challenging to ensure consistent connectivity for onboard sensors and devices. To resolve this, we perform mathematical modeling of the maritime channel and compare it with real measurement data. Through the modeled channel, we verify the received beam gain at buoys on the ocean surface. Additionally, leveraging the modeled wave motions, we estimate future angles of the buoy to use the Extended Kalman Filter (EKF) for design beamforming strategies that adapt to the evolving maritime environment over time. We further validate the effectiveness of these strategies by assessing the results from an outage probability perspective. focuses on improving maritime communication by developing a dynamic model of the maritime channel and implementing a Kalman filter-based buoy motion tracking system. This system is designed to enable precise beamforming, a technique used to direct communication signals more accurately. By improving beamforming, the aim is to enhance the quality of communication links, even in challenging maritime conditions like rough seas and varying sea states. In our simulations that consider realistic wave motions, you've observed significant improvements in link quality due to the enhanced beamforming technique. These improvements are particularly notable in environments with high sea states, where communication challenges are typically more pronounced. The progress made in this area is not just a technical achievement; it has broad implications for the future of maritime communication technologies. This paper promises to revolutionize the way we approach communication in maritime environments, paving the way for more reliable and efficient information exchange on the seas.