• Journal of Communications and Networks
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• 제5권2호
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• pp.116-123
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• 2003

In this paper, we address the problem of designing multirate codes for a multiple-input and multiple-output (MIMO) system by restricting the receiver to be a successive decoding and interference cancellation type, when each of the antennas is encoded independently. Furthermore, it is assumed that the receiver knows the instantaneous fading channel states but the transmitter does not have access to them. It is well known that, in theory, minimummean- square error (MMSE) based successive decoding of multiple access (in multi-user communications) and MIMO channels achieves the total channel capacity. However, for this scheme to perform optimally, the optimal rates of each antenna (per-antenna rates) must be known at the transmitter. We show that the optimal per-antenna rates at the transmitter can be estimated using only the statistical characteristics of the MIMO channel in time-varying Rayleigh MIMO channel environments. Based on the results, multirate codes are designed using punctured turbo codes for a horizontal codedMIMOsystem. Simulation results show performances within about one to two dBs of MIMO channel capacity.

• 한국통신학회논문지
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• 제32권8A호
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• pp.813-819
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• 2007

본 논문에서는 트랜스패런트(transparent) 모바일 다중 홉 릴레이(Mobile Multi-hop Relay; MMR)를 갖는 OFDM 기반 셀룰러 시스템에서 발생하는 전파 지연(propagation delay)의 영향을 분석한다. 또한, 수율 상승(throughput enhancement; TE) 또는 상호협력 전송(cooperative transmission)을 위한 MMR을 갖는 OFDM 시스템에서 전파 지연에 대한 영향을 극복할 수 있는 LS 채널 추정 기법과 MMSE 채널 추정 기법을 제안한다. 본 논문에서 제안하는 채널 추정 기법은 TE MMR 시스템 또는 상호협력 MMR 시스템에서 전파 지연으로 인한 채널 추정 성능 열화 현상을 극복할 수 있다. 모의실험을 통해 MSE와 BER 측면에서 본 논문에서 제안하는 기법이 기존 기법보다 우수함을 보인다.

• Journal of Communications and Networks
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• 제5권4호
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• pp.354-364
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• 2003

A general method for the evaluation of the symbol error probability (SER) of ultra wideband (UWB) systems with various kind of modulation schemes (N-PAM, M-PPM, Bi-Orthogonal), in presence of multipath channel, multiuser and strong narrowband interference, is presented. This method is shown to be able to include all the principal multiaccess techniques proposed so far for UWB, time hopping (TH), direct sequence (DS) and optical orthogonal codes (OOC). A comparison between the performance of these multiple access and modulation techniques is given, for both ideal Rake receiver and minimum mean square error (MMSE) equalizer. It is shown that for all the analyzed multiple access schemes, a Rake receiver exhibits a high error floor in presence of narrowband interference (NBI) and that the value of the error floor is in-fluenced by the spectral characteristics of the spreading code. As expected, an MMSE receiver offers better performance, representing a promising candidate for UWB systems. When the multiuser interference is dominant, all multiple access techniques exhibit similar performance under high-load conditions. If the number of users is significantly lower than the spreading factor, then DS outperforms both TH and OOC. Finally 2PPM is shown to offer better performance than the other modulation schemes in presence of multiuser interference; increasing the spreading factor is proposed as a more effective strategy for SER reduction than the use of time diversity.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권7호
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• pp.2998-3017
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• 2018

Massive (large-scale) MIMO (multiple-input multiple-output) is one of the key technologies in next-generation wireless communication systems. This paper proposes a high-performance low-complexity turbo receiver for SC-FDMA (single-carrier frequency-division multiple access) based MMIMO (massive MIMO) systems. Because SC-FDMA technology has the desirable characteristics of OFDMA (orthogonal frequency division multiple access) and the low PAPR (peak-to-average power ratio) of SC transmission schemes, the 3GPP LTE (long-term evolution) has adopted it as the uplink transmission to meet the demand high data rate and low error rate performance. The complexity of computing will be increased greatly in base station with massive MIMO (MMIMO) system. In this paper, a low-complexity adaptive turbo equalization receiver based on normalized minimal symbol-error-rate for MMIMO SC-FDMA system is proposed. The proposed receiver is with low complexity than that of the conventional turbo MMSE (minimum mean square error) equalizer and is also with better bit error rate (BER) performance than that of the conventional adaptive turbo MMSE equalizer. Simulation results confirm the effectiveness of the proposed scheme.

• 대한전자공학회논문지TC
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• 제40권8호
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• pp.301-306
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• 2003

본 논문에서는 RBF Network를 이용한 새로운 수신기법을 MIMO 환경에서 제안한다. 그리고, 제안된 수신기의 성능을 Rich-scattering 페이딩 채널에서의 모의 실험을 통해 검증한다. 모의 실험 결과를 통해 제안된 수신기가 MLD와 유사한 성능을 나타내고, VBLAST-ZF와 VBLAST-MMSE보다 우수한 성능을 나타냄을 확인하였다. 그리고, 다양한 송수신 안테나의 개수 및 변조 기법에 따른 RBF 개수가 성능에 미치는 영향을 조사하였으며, 제안된 수신기의 성능을 RBF 중심값의 초기화 율에 따라 확인하였다.

• 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.

• 한국산학기술학회:학술대회논문집
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• 한국산학기술학회 2009년도 추계학술발표논문집
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• pp.985-988
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• 2009

본 논문에서는 3GPP-LTE 상향링크 기술인 SC-FDMA에서 적응형 등화기 성능에 대해 알아보았다. SC-FDMA의 등화기에서 계산량을 줄여 효율성을 높이는 방법으로 ZF(Zero Forcing)과 MMSE(Minimum Mean Square Error)를 이용한 스위칭 기법을 제안하고, 제안한 시스템에서의 성능을 비교 분석하고자 한다. 제안한 스위칭 기법을 사용함으로써 SNR이 낮을 때는 MMSE를 이용하여 잡음에 대한 영향을 최소화 시켜주고, SNR이 높을 때는 ZF을 써서 상대적으로 복잡도가 적은 등화 기법을 통해 전체적인 시스템 복잡도를 줄여 효율성을 높이고자 한다.

• 한국통신학회논문지
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• 제24권6A호
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• pp.881-887
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• 1999

In this paper, were introduce two types of equalizers, called equalizer-a and equalizer-b, applying to wireless communications having unknown channel characteristics. The equalizer-a, which has the single sample detector with equalizer system, is developed while the equalizer-b has the partition detectors with the same system used in equalizer-a. The methodologiy we adopt for designing the equalizers is that the sample space is partitioned into finite number of regions by using quantiles, which are estimated by robbins-monro stochastic approximation (RMSA) algorithm, and the coefficients of equalizers are calculated based on nonlinear minimum mean, square error (MMSE) algorithm. Through the computer simulation, the equalizers show much better performance in equiprobably partitioned sample subspaces of observations than the single sample detector and the detector, which has the conventional equalizer, in unquantized observation space under various noise environments.

• 대한음성학회지:말소리
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• 제67호
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• pp.195-216
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• 2008

This paper addresses the problem of single-channel speech separation to extract the speech signal uttered by the speaker of interest from a mixture of speech signals. We propose to apply time-frequency smoothing to the existing statistical single-channel speech separation algorithms: The soft mask and the minimum-mean-square-error (MMSE) algorithms. In the proposed method, we use the two smoothing later. One is the uniform mask filter whose filter length is uniform at the time-Sequency domain, and the other is the met-scale filter whose filter length is met-scaled at the time domain. In our speech separation experiments, the uniform mask filter improves speaker-to-interference ratio (SIR) by 2.1dB and 1dB for the soft mask algorithm and the MMSE algorithm, respectively, whereas the mel-scale filter achieves 1.1dB and 0.8dB for the same algorithms.

• 말소리와 음성과학
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• 제3권1호
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• pp.87-94
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• 2011

The decision directed (DD) approach is widely used to determine a priori SNR from noisy speech signals. In conventional speech enhancement systems with a DD approach, a priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a phase-dependent two-step a priori SNR estimator based on the minimum mean square error (MMSE) in the log-mel spectral domain so that we can consider both magnitude and phase information, and it can overcome the performance degradation caused by one frame delay. From the experimental results, the proposed estimator is shown to improve the output SNR of enhanced speech signals by 2.3 dB compared to the conventional DD approach-based system.