• ETRI Journal
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• 제32권2호
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• pp.333-335
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• 2010

This letter proposes a non-coherent blind time-of-arrival (TOA) estimation scheme for impulse radio ultra-wideband systems. The TOA estimation is performed in two consecutive phases: the Rayleigh-quotient theorem-based coarse-signal acquisition (CSA) and the iterative-threshold-test-based fine time estimation (FTE). The proposed scheme serves in a blind manner without demanding any a priori knowledge of the channel and the noise. Analysis and simulations demonstrate that the proposed scheme significantly increases the signal detection probability in CSA and ameliorates the TOA estimation accuracy in FTE.

• 한국멀티미디어학회논문지
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• 제14권8호
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• pp.965-980
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• 2011

In this paper, the use of conditional Fuzzy C-Means (CFCM) aimed at estimation of desired states of an unknown digital communication channel is investigated for blind channel equalization. In the proposed CFCM, a collection of clustered centers is treated as a set of pre-defined desired channel states, and used to extract channel output states. By considering the combinations of the extracted channel output states, all possible sets of desired channel states are constructed. The set of desired states characterized by the maximal value of the Bayesian fitness function is subsequently selected for the next fuzzy clustering epoch. This modification of CFCM makes it possible to search for the optimal desired channel states of an unknown channel. Finally, given the desired channel states, the Bayesian equalizer is implemented to reconstruct transmitted symbols. In a series of simulations, binary signals are generated at random with Gaussian noise, and both linear and nonlinear channels are evaluated. The experimental studies demonstrate that the performance (being expressed in terms of accuracy and speed) of the proposed CFCM is superior to the performance of the existing method exploiting the "conventional" Fuzzy C-Means (FCM).

• 전자공학회논문지S
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• 제35S권11호
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• pp.1-11
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• 1998

본 논문에서는 ML(maximum likelihood) 원리와 디지탈 통신 시스템의 고유 특성인 유한 알파벳 특성(FAP: finite alphabet properties)에 근거한 블라인드 최대비 결합(MRC: maximal ratio combining)을 위한 간단한 방법들을 제안한다. 이 방법들은 아주 작은 길이의 데이터를 가지고도 채널 파라미터들을 정확하게 추정할 수 있기 때문에, 이 방법들을 사용하면 거의 완벽한 최대비 결합을 수행할 수 있다. 이 방법들은 교번투영 기법(alternating projection technique)을 이용하여 다이버시티 가지들에 대한 채널 파라미터와 데이터 시퀀스를 동시에 추정한다. 두 가지 다른 JC-DSE (joint combining and data sequence estimation)방법과 PC-BPE (pre-combining and blind phase estimation) 방법이 제안되며, 전영역 최적화를 보장할 수 있도록 하는 효율적인 초기화 방법도 제시된다. 모의실험 결과들을 통하여, 제안된 두가지 방법의 심볼 오류율과 채널 파라미터의 추정 정확도에 관한 성능을 보여준다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.685-688
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• 2003

In this paper, a performance comparison between two blind signal estimation algorithms in a LTI channel is considered. The two algorithms, Iterative Least-Squares with Projection (ILSP) and a modified ILSP, are based on the finite-alphabet property of input symbols. This case typically arises in a multiple access system with a sensor array antenna at the receiving end. We start with the formulation of a maximum-likelihood (ML) estimation problem under an additive white Gaussian noise assumption. A blind ML estimator is derived with its iterative algorithm for calculation. Then we narrow down the consideration of this problem to QPSK case so that a modified algorithm is proposed for $\pi$/4-QPSK case. The modified version is compared with the original ILSP algorithm in terms of the rate of the convergence to global minima. A computer simulation shows that the modified algorithm gives a better performance. This result implies that the performance of the blind separation algorithms may be greatly improved by adopting a smart coding scheme with rich structure.

• ETRI Journal
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• 제31권2호
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• pp.162-172
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• 2009

A linear-prediction-based blind equalization algorithm for single-input single-output (SISO) finite impulse response/infinite impulse response (FIR/IIR) channels is proposed. The new algorithm is based on second-order statistics, and it does not require channel order estimation. By oversampling the channel output, the SISO channel model is converted to a special single-input multiple-output (SIMO) model. Two forward linear predictors with consecutive prediction delays are applied to the subchannel outputs of the SIMO model. It is demonstrated that the partial parameters of the SIMO model can be estimated from the difference between the prediction errors when the length of the predictors is sufficiently large. The sufficient filter length for achieving the optimal prediction is also derived. Based on the estimated parameters, both batch and adaptive minimum-mean-square-error equalizers are developed. The performance of the proposed equalizers is evaluated by computer simulations and compared with existing algorithms.

• 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.

• Journal of Communications and Networks
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• 제9권3호
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• pp.219-228
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• 2007

In this paper, we present a novel blind adaptive multiuser detector structure and three robust recursive filters to improve the performance in CDMA environments: Sigma point kalman filter (SPKF), particle filter (PF), and Gaussian mixture sigma point particle filter (GMSPPF). Our proposed robust recursive filters have superior performance over a conventional extended Kalman filter (EKF). The proposed multiuser detector algorithms initially use Kalman prediction form to estimated channel parameters, and unknown data symbol be predicted. Second, based on this predicted data symbol, the robust recursive filters (e.g., GMSPPF) is a refined estimation of joint multipaths and time delays. With these estimated multipaths and time delays, data symbol detection is carried out (Kalman correction form). Computer simulations show that the proposed algorithms outperform the conventional blind multiuser detector with the EKF. Also we can see it provides a more viable means for tracking time-varying amplitudes and time delays in CDMA communication systems, compared to that of the EKF for near-far ratio of 20 dB. For this reason, it is believed that the proposed channel estimators can replace well-known filter such as the EKF.

• 한국통신학회논문지
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• 제37권6A호
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• pp.450-457
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• 2012

직교 주파수 분할 다중화 시스템에서 비중복 프리코딩을 이용한 미상 채널 추정 방식을 제안한다. 제안한 방식에서는 수신 신호에 대한 공분산 행렬을 구하고, 그 행렬의 각 원소들을 프리코딩 행렬의 각 원소로 나눔으로써 변형된 공분산 행렬을 구한다. 이 행렬의 최대 고유값에 해당하는 고유벡터를 구함으로써 채널 계수들을 추정하게 된다. 이 때, 고유 벡터를 구하기 위하여 많은 계산량을 필요로 하는 고유치 분해 기법 대신에 간단한 파워 기법을 적용함으로써 계산량을 크게 줄이게 된다. 제안하는 채널 추정 방식의 평균 제곱 오차에 대한 이론적인 값을 유도하고, 모의실험 결과와 비교함으로써 유도한 값이 실험 결과와 일치한다는 것을 확인한다. 또한, 모의실험을 통해서, 제안한 방법이 기존 방법들보다 더 우수한 채널 추정 성능과 비트 오율 성능을 나타낸다는 것을 보인다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2004년도 추계학술발표대회논문집 제23권 2호
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• pp.401-404
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• 2004

수중음향통신 관점에서 바다와 같은 수중채널은 수십심벌에 이르는 다중경로와 빠르게 변화하는 도플러 효과가 존재한다. 따라서 신뢰성 있는 수중음향통신을 위해서는 채널 등화가 반드시 필요하다. 본 논문에서는 이차 통계량 기반의 Blind Identification 알고리듬을 이용한 수중음향 채널추정 기법을 이용하여 무향수조의 채널응답을 분석하였다. 실험을 통하여 알고리듬의 추정 정확도 및 트랜스듀서 특성을 포함한 무향수조의 채널 응답 특성에 대하여 논한다.

• 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.