• Journal of the Society of Naval Architects of Korea
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• v.48 no.2
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• pp.121-127
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• 2011

As considerable interests in noise emission from a ship have been increased, the need for localization of noise sources of the marine propeller generating cavitation and singing noise is looming large. In many practical cases, cavitation and singing noise occur on a particular position of the certain blade of the propeller. It is so important to know the position of noise source correctly in order to eliminate or suppress unwanted noise. In this study, we develop "noise source localization technology" using TDOA method. Experimental measurements carried out at the circulating water channel and towing tank show that noise source can be clearly identified and localized using TDOA method.

• ETRI Journal
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• v.45 no.6
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• pp.952-962
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• 2023

In this study, a space-time transmission scheme is proposed to tackle the limitations of channel estimation with orthogonal pilot information in colocated multiple-input multiple-output systems with several transmitting and receiving antennas. Channel information is obtained using orthogonal pilots. Channel estimation introduces pilot heads required to estimate a channel. This leads to bandwidth insufficiency. As a result, trade-offs exist between the number of pilots required to estimate a channel versus spectral efficiency. The detection of data symbols is performed using the maximum likelihood decoding method as it provides a consistent approach to parameter estimation problems. The moment-generating function of the instantaneous signal-to-noise ratio is used to drive an approximate expression of the symbol error rate for the proposed scheme. Furthermore, the order of diversity is less by one than the number of receiver antennas used in the proposed scheme. The effect of the length of a pilot sequence on the proposed scheme's performance is also investigated.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1
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• pp.38-44
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• 1999

In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. A simulation was conducted to verify the effectiveness of the method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3A
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• pp.256-268
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• 2008

This paper proposes an efficient scheme to track the time variant channel induced by multi-path Rayleigh fading in mobile wireless Multiple Input Multiple Output-Orthogonal Frequency Division Multiplexing (MIMO-OFDM) systems with null sub-carriers. In the proposed method, a blind channel response predictor is designed to cope with the time variant channel. The proposed channel tracking scheme consists of a frequency domain estimation approach that is coupled with a Minimum Mean Square Error (MMSE) time domain estimation method, and does not require any matrix inverse calculation during each OFDM symbol. The main attributes of the proposed scheme are its reduced computational complexity and good tracking performance of channel variations. The simulation results show that the proposed method exhibits superior performance than the conventional channel tracking method [4] in time varying channel environments. At a Doppler frequency of 100Hz and bit error rates (BER) of 10-4, signal-to-noise power ratio (Eb/N0) gains of about 2.5dB are achieved relative to the conventional channel tracking method [4]. At a Doppler frequency of 200Hz, the performance difference between the proposed method and conventional one becomes much larger.

• Journal of Digital Contents Society
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• v.6 no.1
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• pp.41-48
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• 2005

In this paper, we propose and evaluate symbol timing recovery algorithm for S-DMT cable modern, which supports more channels and better quality symmetric mutimedia service over HFC network. We adopt timing recovery algorithm of PN sequence insertion in time domain and evaluate the performance of it in various noise channel such as AWGN, ISI, impulse. We verified that performance of this algorithm is depends on the channel noise environment and sampling clock offset and that over 10 dB degradation of Eb/No is occurred at the timing failure probability of $10^3$ in the composite noise channel of AWGN, ISI, and impulse in comparison with impulse noise-alone channel. finally, we verified that this algorithm showed good timing failure probability in case of sampling clock optimization was performed in advance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12C
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• pp.1201-1207
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• 2005

When the isolation between transmitter and receiver in Electronic Warfare equipment is not sufficient, the radiated noise jamming signal from the transmitter feeds back into the receiver and interferes with receiving radar pulse signal. Therefore pulse jamming and noise jamming can't be performed together in the same frequency bands. In this paper, we present a time-sharing TX/RX control technique of the switch matrix which inhibits the transmission of noise jamming signal by using the predicted gate of pulse train and also makes the corresponding channel filter operate to receive the radar pulse signal during the predicted gate pulse. This technique was implemented by EPLD and confirmed by experiment. The proposed technique enables the pulse jamming and the noise jamming to be simultaneously executed in multiple jamming environments.

• ETRI Journal
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• v.20 no.1
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• pp.16-27
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• 1998

In this paper, we consider the acquisition performance of an IS-95 reverse link access channel slot as a function of system design parameters such as postdetection integration length and the number of access channel message block repetitons. The uncertainty region of the reverse link spreading codes compared to that of forward link is very small, since the uncertainty region of the reverse link is determined by a cell radius. Thus, the parallel acquisiton technique in the reverse link is more efficient than a serial acquisition technique in terms of implementation and of acquisition time. The parallel acquisition is achieved by a bank of N parallel I/Q noncoherent correlator are analyzed for band-limited noise and the Rayleigh fast fading channel. The detection probability is derived for multiple correct code-phase offsets and multipath fading. The probability of no message error is derived when rake combining, access channel message block combining, and Viterbi decoding are applied. Numerical results provide the acquisition performance for system design parameters such as postdetection integration length and number of access channel message block repetitions in case of a random access on a mobile station.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.7
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• pp.33-41
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• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.84-89
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• 2002

In this paper, we present the design method of noise reduction circuit in telephonic system. The circuit consists of compressor, expander and a filter. The basic idea of a proposed method compresses the audible signal in order to mask the channel noise during transmission and then expand at the reverse rate the transmitted signal to naturally recover the original signal. Of course, there should be no distortion or other degradation of the audio itself in passing through companding(compress/expand) cycle. In the compressing process, the gain of compressor is automatically controlled by the envelope level of input signal in order to increase the effective dynamic range of input signal and to improve the signal to noise ratio. The compressed rate is the root time of a audible signal. The compressed signal should be expanded at the square time of the signal to recover a original signal. Simulation shows the proposed method improves the performance of the noise reduction of a channel noise as well as stability. 

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.167-171
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• 2002

As a channel of a DS-CDMA system is shared among several users, the receivers face the problem of MAI. Also the multipath scenario leads to ISI. Both components are undesired, but unlike the additive noise process, which is usually completely unpredictable, their space-time structure helps to estimate and remove them. This paper investigates a DS-CDMA system with a fading multipath channel. The investigations have been separated into a channel estimation part and a reception part. In the first part, the estimation of multipath parameters(DOA, TOA) are considered, space-time beamformers(Decorrelating, Match-Filter, Wiener-Hopf) are performed in the second part. To assess many different estimation techniques and beamformers, the simulation compared with theoretical values is performed.