• Journal of electromagnetic engineering and science
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• 제12권3호
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• pp.203-209
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• 2012

In order to improve sensing performance when the noise variance is not known, this paper considers a so-called blind spectrum sensing technique that is based on eigenvalue models. In this paper, we employed the spiked population models in order to identify the miss detection probability. At first, we try to estimate the unknown noise variance based on the blind measurements at a secondary location. We then investigate the performance of detection, in terms of both theoretical and empirical aspects, after applying this estimated noise variance result. In addition, we study the effects of the number of SUs and the number of samples on the spectrum sensing performance.

• 한국통신학회논문지
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• 제21권7호
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• pp.1777-1786
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• 1996

This paper describes theories and implementation techniques of a digital high-definition television(HDTV) modem based on 32-QAM for terrestrial broadcasting. We proposed a digital demodulation scheme and a symbol timing recovery structure based on the band edge component maximization(BECM) method. The adaptive equalizer has 256 complex taps to remove the multipath of delays ranging from -2.mu.s~+24.mu.s with a new T/2-spaced blind equalization algorithm. computer simulation results reveal that the proposed algorithm outperforms other conventional blind equalization algorithm a digital HDTV modem with 4.91MHz symobol rate is implemented by utilizing the proposed algorithms. All processings for modem operations such as demodulation, estimation of symbol timing phase error, adaptive equalization, and carrier recovery except IF signal processing and sampling phase control part of the AD converter are done in digital domain. Especially, the carrier recovery loop can track a carrier offset of upto .+-.350KHz.

• Journal of Communications and Networks
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• 제5권2호
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• pp.157-166
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• 2003

This paper deals with the problem of channel identification for Single Input Multiple Output (SIMO) slow fading channels using clustering algorithms. Due to the intrinsic memory of the discrete-time model of the channel, over short observation periods, the received data vectors of the SIMO model are spread in clusters because of the AWGN noise. Each cluster is practically centered around the ideal channel output labels without noise and the noisy received vectors are distributed according to a multivariate Gaussian distribution. Starting from the Markov SIMO channel model, simultaneous maximum ikelihood estimation of the input vector and the channel coefficients reduce to one of obtaining the values of this pair that minimizes the sum of the Euclidean norms between the received and the estimated output vectors. Viterbi algorithm can be used for this purpose provided the trellis diagram of the Markov model can be labeled with the noiseless channel outputs. The problem of identification of the ideal channel outputs, which is the focus of this paper, is then equivalent to designing a Vector Quantizer (VQ) from a training set corresponding to the observed noisy channel outputs. The Linde-Buzo-Gray (LBG)-type clustering algorithms [1] could be used to obtain the noiseless channel output labels from the noisy received vectors. One problem with the use of such algorithms for blind time-varying channel identification is the codebook initialization. This paper looks at two critical issues with regards to the use of VQ for channel identification. The first has to deal with the applicability of this technique in general; we present theoretical results for the conditions under which the technique may be applicable. The second aims at overcoming the codebook initialization problem by proposing a novel approach which attempts to make the first phase of the channel estimation faster than the classical codebook initialization methods. Sample simulation results are provided confirming the effectiveness of the proposed initialization technique.

• 한국통신학회논문지
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• 제42권5호
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• pp.951-958
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• 2017

채널 재구성 기법이란 통신시스템에서 의도되지 않은 수신자가 수신 신호로부터 어떤 채널 부호가 사용되었는지, 주요 파라미터는 무엇인지를 알아내는 기법이다. 본 논문은 수신한 신호가 길쌈부호로 부호화된 경우, 사용된 길쌈부호의 주요파라미터인 입출력단의 비트수인 k와 n, 그리고 $k{\times}n$ 생성다항식행렬(Polynomial Generator Matrix, PGM)을 찾아내는 기법에 대해 다룬다. 본 논문은 M. Marazin 등이 제안한, 피버팅을 통한 가우스 조단소거법(Gauss Jordan Elimination Through Pivoting, GJETP)을 사용한 길쌈부호의 채널 재구성 기법에서 채널오류율과 무관하게 임계값을 설정해주던 것과 달리, 수신한 시퀀스로부터 2진 대칭 채널(Bynary Symetric Channel, BSC)의 채널오류확률을 추정하고 이로부터 임계값을 설정하는 방식을 제안하고, S. Shaojing 등의 연판정(soft decision) 값을 이용한 기법을 적용시켜서 채널 재구성 기법의 성공률을 향상시켰다.

• Journal of Biosystems Engineering
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• 제34권4호
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• pp.278-285
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• 2009

Navigation system is applied in variety of fields including the simple location positioning, autopilot navigation of unmanned robot tractor, autonomous guidance systems for agricultural vehicles, construction of large field works that require high precision and map making process. Particularly utilization of GPS (Global Positioning System) is very common in the present navigation system. This study introduces a navigation system for autonomous field robot that travels to the pre-input path using GPS information. Performance of the GPS- based navigation is highly depended on its receiving rate because GPS receivers do not acquire any navigation information in the period between the refresh intervals. So this study presents an algorithm that improves an accuracy of the navigation by estimation the positional information during the blind period of a low rate GPS receiver. In fact the algorithm calculated the robot's heading in a 50 Hz rate, so the blind period of an 1 Hz GPS receiver is extensively covered. Consequently implementation of the algorithm to the GPS based navigation showed an improvement in guidance accuracy. The conventional field robot directly carried an expensive control computer and sensors onboard, therefore the miniaturization and weight reduction of the robot was limited. In this paper, the field robot carried only communication equipments such as GPS module, normal RC receiver, and bluetooth modem. This enabled the field robot to be built in an economic cost and miniature size.

• 한국음향학회지
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• 제41권2호
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• pp.210-217
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• 2022

수동 소나 시스템에서 음원 깊이 추정 분야는표적 식별, 추적 등 다양한 전술에 활용할 수 있기 때문에 많은 연구가 지난 수십년간 진행되어 왔다. 본 논문은 기존 논문[조성일 외, 한국음향학회지 제38권 제1호, 120-127(2019)]의 문제점이었던 음원 깊이가두 곳에서 추정되는 모호성을 해결하는데 목적이 있다. 해수면과 해저면에 반사되어 나타나는채널 임펄스 응답의 위상천이 패턴을 이용하여 모호성을 제거하며, 제거 후 하나의 깊이에서 채널 임펄스 응답의 교차점을 통해 음원의깊이를 추정한다. 음원에 대한 정보가 없고, 연속적인 신호 혹은 소음에서 채널 임펄스 응답을 추정하기 위해 음선 기반 블라인드 디컨벌루션 기법이 사용되며, 제안된 알고리즘은 시뮬레이션를 통하여 검증하였다.

• Journal of Communications and Networks
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• 제4권3호
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• pp.189-198
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• 2002

Precoder has been shown to be able to provide source diversity and design flexibility. In this paper we employ precoding techniques for block transmission based on a multirate filterbank structure. To accommodate multiuser communication with variable data rates, different precoders with corresponding coefficients and up/down sampling rates are used. However, due to unknown multipath distortion, different interferences may exist in the received data, such as multiuser interference, intersymbol interference and interblock interference. To estimate channel parameters for a desired user, we employ all structured signature waveforms associated with different symbols of that user and apply subspace techniques. Therefore better performance of channel estimator can be achieved than the conventional subspace method based only on the signature of the current symbol. The delay for that user can also be jointly estimated. Channel identifiability conditions and asymptotic channel estimation error are investigated in detail. Numerical examples are provided to justify the proposed method. gest either multicode (MC) or multiple processing gain (MPG) mechanism [2], while requiring data rates to be integral multiples of some basic low-rate. In order to support variable rate transmission however, a comprehensive scheme needs to be investigated.

• ETRI Journal
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• 제41권5호
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• pp.658-669
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• 2019

For dual-channel time-frequency (TF) overlapped signals with low sparsity in underdetermined blind source separation (UBSS), this paper proposes an effective method based on interval probability to estimate and expand the types of mixing matrices. First, the detection of TF single-source points (TF-SSP) is used to improve the TF sparsity of each source. For more distinguishability, as the ratios of the coefficients from different columns of the mixing matrix are close, a local peak-detection mechanism based on interval probability (LPIP) is proposed. LPIP utilizes uniform subintervals to optimize and classify the TF coefficient ratios of the detected TF-SSP effectively in the case of a high level of TF overlap among sources and reduces the TF interference points and redundant signal features greatly to enhance the estimation accuracy. The simulation results show that under both noiseless and noisy cases, the proposed method performs better than the selected mainstream traditional methods, has good robustness, and has low algorithm complexity.

• 한국통신학회논문지
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• 제36권6C호
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• pp.391-397
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• 2011

본 논문에서는 직교 주파수 분할 다중화 (orthogonal frequency division multiplexing: OFDM) 기반 인지 무선 (cognitive radio: CR) 시스템을 위한 주파수 옵셋 추정 기법에 대해 연구한다. CR 통신 환경에서는 종종 비정규 잡음에 발생하므로 이를 고려하지 않은 주파수 옵셋 추정 기법들의 추정 성능은 크게 하락한다. 본 논문에서는 비정규 잡음 환경에서 OFDM 기반 CR 시스템을 위한 블라인드 방식의 최대 우도 이론 기반 주파수 옵셋 추정 기법을 제안한다. 제안하는 기법은 이제까지의 기법보다 비정규 잡음 환경에서 우수한 추정 성능을 갖는다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 추계종합학술대회 논문집(1)
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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.