• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.210-217
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• 2022

Passive source depth estimation has been studied for decades since the source depth can be used for target classification, target tracking, etc. The purpose of this paper is to solve the problem of ambiguity in the previous paper [S.-il. Cho et al. (in Korean), J. Acoust. Soc. Kr. 38, 120-127 (2019)] that source depth is estimated in two points. The patterns of phase shift of Channel Impulse Response(CIR) reflected in ocean surface and bottom is used for removing ambiguity of the source depth estimation, and after removing ambiguity, source depth is estimated at one point through the intersection of CIR. In order to extract CIR in case of unknown source signal and continuous signal or noise, Ray-based blind deconvolution is used. The proposed algorithm is demonstrated through numerical simulation in ocean waveguide.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.229-232
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• 2000

Orthogonal frequency division multiplexing(OFDM) has meanwhile become part of several telecommunicati ons standards, such as satellite and terrestrial digital audio broadcasting(DAB), digital terrestrial TV broad casting(DVB), asymmetric digital subscriber line(ADSL) for high-bit-rate digital subscriber services on twisted-pair channels, and broadband indoor wireless systems. In his paper, we show that OFDM signals contain sufficient structure to accomplish blind channel estimation using second order statistics only. This method doesn't require redundancy as cp in transmitter. And the result is compared with PSAM channel estimation as least square, linear minimum mean square, singular value decomposition.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.331-339
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• 2016

This paper suggests an estimation method of the source signal and the channel impulse response (CIR) using ray-based blind deconvolution (RBD) in the underwater acoustic channel environment where Doppler effect exists by the relative motion between source and receiver. It is difficult to estimate the CIR on Doppler effect by the matched filter with a highly Doppler-sensitive waveform such as the m-sequence signal because Doppler shift can severely degrade the correlation between the received signal corrupted by Doppler effect and the original source signal. In this study, the Doppler-shifted source-signal's phase is estimated using the RBD, and the received signal is compensated by it to obtain the Doppler-corrected CIR. It is verified that using the matched filter with the received signal from the experimental data fails to estimate the CIR while the obtained CIR by the suggested method has the similarity to the propagation path of the ray model. Also, the results show that the reconstructed source signal using the RBD has the better Doppler shift compensation than the Doppler-shifted source signal derived from scattering function.

• The Journal of the Acoustical Society of Korea
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• v.36 no.6
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• pp.378-386
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• 2017

Accurate distance measurement between maneuver target in underwater and measuring devices is required to perform quantitative test evaluation in marine weapons system R&D process. In general, the target distance is measured using a one-way ToA (Time of Arrival) method that calculates the time difference between transmitted and received signals from the two accurately synchronized devices. However, the distance estimation performance is degraded because of the multi-path environments. In this paper, the time-variant transfer function of complex underwater environment is estimated from each received data frame using RBD (Ray-based Blind Deconvolution), and the estimated time-variant transfer function is then used to get rid of the effect about complex underwater environment and to recover the data signal using PTRM (Passive Time Reversal Mirror). The result from the simulation and experimental data show that the suggested method improve the distance estimation performance when comparing with the conventional ToA 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.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.169-172
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• 2000

본 논문에서는 다중 경로상의 DS-CDMA 시스템을 위해 우리가 이미 제안한 Rake 수신기와 다이버시티 결합 구조를 이용하는 CMA 기반의 적응 간섭 제거 기법에서 실제 채널 추정 기법을 사용할 때 부정확한 채널추정이 성능에 미치는 영향을 검증하였으며, sliding widow averaging 형태의 단순한 채널 추정 기법을 이용할 경우 slow 페이딩 및 fast 페이딩 환경에서의 모의실험 결과 이상적인 채널 추정을 가정한 경우와 유사한 우수한 성능을 보임을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.4
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• pp.318-324
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• 2013

For blind channel equalization, the reliability of signal estimate determines the convergence speed and steady-state performance of the equalizer. Therefore the nonlinear estimator and reference signal being used in signal estimate should be chosen appropriately. In this paper, to increase the reliability of the signal estimate, two errors were obtained by applying coarse signal points and dense signal points respectively to signal estimate, the relative reliabilities of two errors were calculated, then the equalizer was adapted deferentially utilizing the reliabilities. At this point, by applying two errors alternately, two modes of operation were smoothly combined. Through computer simulations the proposed method was confirmed to achieve fast transient state convergence and low steady-state error compared to traditional methods.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.559-562
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• 2011

In this research, We derived Recursive Least Squares(RLS) algorithm with adaptive maximum-likelihood channel estimate for digital pulse amplitude modulated sequence in the presence of intersymbol interference and additive white Gaussian noise. RLS algorithms have better convergence characteristics than conventional algorithms, LMS (Least Mean Squares) algorithms.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.146-146
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• 2002

We present a new technique for achieving source separation when given only a single channel recording. The main idea is based on exploiting the inherent time structure of sound sources by learning a priori sets of time-domain basis functions that encode the sources in a statistically efficient manner. We derive a learning algorithm using a maximum likelihood approach given the observed single channel data and sets of basis functions. For each time point we infer the source parameters and their contribution factors. This inference is possible due to the prior knowledge of the basis functions and the associated coefficient densities. A flexible model for density estimation allows accurate modeling of the observation, and our experimental results exhibit a high level of separation performance for simulated mixtures as well as real environment recordings employing mixtures of two different sources. We show separation results of two music signals as well as the separation of two voice signals.

• Electronics and Telecommunications Trends
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• v.21 no.6 s.102
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• pp.124-132
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• 2006

무선 통신 시스템에서 다중 경로 감쇠로 인한 심볼의 크기와 위상의 왜곡이 일어나는데 이를 추정하여 보상하기 위해 채널 추정 기법이 사용된다. OFDM 시스템에서의 채널 추정 기법에는 훈련 심볼이나 파일럿을 사용하여 채널을 추정하는 기법과 파일럿을 사용하지 않는 Blind 채널 추정 기법 등이 있다. 본 고에서는 I장 개요에 이어 II장에서는 훈련 심볼을 이용한 채널 추정 기법을 WPAN과 WLAN을 예로 들어 설명하고 LS,1-D LMMSE, Low rank LMMSE 알고리듬에 대해 설명한다. III장에서는 PSAM 채널추정에 대해 1차원 PSAM 기법과 2차원 파일럿 패턴과 Wiener filtering을 이용한 2-D LMMSE, Low rank LMMSE, Separable Wiener filter에 대하여 설명한다. IV장에서는 ESAE와 Blind 채널 추정 기법을 간략히 소개하고, V장에서 채널 추정의 최신 연구 동향을 소개한다.