• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4C
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• pp.414-423
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• 2009

This paper considers a linear precoding scheme that achieves near optimal sum rate. While the minimum mean square error (MMSE) precoding provides the better MSE performance at all signal-to-noise ratio (SNR) than the zero forcing (ZF) precoding, its sum rate shows superior performance to ZF precoding at low SNR but inferior performance to ZF precoding at high SNR, From this observation, we first propose a near optimal linear precoding scheme in terms of sum rate. The resulting precoding scheme regularizes ZF precoding to maximize the sum rate, resulting in better sum rate performance than both ZF precoding and MMSE precoding at all SNR ranges. To find regularization parameters, we propose a simple algorithm such that locally maximal sum rate is achieved. As a low complexity alternative, we also propose a simple power re-allocation scheme in the conventional regularized channel inversion scheme. Finally, the proposed scheme is tested under the presence of channel estimation error. By simulation, we show that the proposed scheme can maintain the performance gain in the presence of channel estimation error and is robust to the channel estimation error.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.64 no.12
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• pp.1756-1760
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• 2015

This paper focuses on a priori signal to noise ratio (SNR) estimation method for the speech enhancement. There are many researches for speech enhancement with several ambient noise cancellation methods. The method based on spectral subtraction (SS) which is widely used in noise reduction has a trade-off between the performance and the distortion of the signals. So the need of adaptive method like an estimated a priori SNR being able to making a high performance and low distortion is increasing. The decision directed (DD) approach is used to determine a priori SNR in noisy speech signals. A priori SNR is estimated by using only the magnitude components and consequently follows a posteriori SNR with one frame delay. We propose a modified a priori SNR estimator and the weighted rational transfer function for speech enhancement with wind noises. The experimental result shows the performance of our proposed estimator is better Perceptual Evaluation of Speech Quality scores (PESQ, ITU-T P.862) compare to the conventional DD approach-based systems and different noise reduction methods.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.25 no.7
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• pp.782-790
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• 2014

In this paper, we apply the 2-D Matrix Pencil Method(MPM) to the estimation of the direction of arrival(DOA) of multiple signals of interest(SOIs) in bistatic MIMO radar. The 2-D MPM shows remarkable performance under a low SNR environment and low computational complexity to estimate the DOA of multiple SOIs. Also, it is possible to estimate the direction of departure(DOD) which is an angle from transmitter to target. To verify the proposed algorithm, we applied the proposed algorithm to a uniformly spaced linear array(ULA) and compared the RMSE(Root Mean Square Error) of DOA and DOD under the various SNR with those of the 2-D Capon algorithm.

• Journal of Broadcast Engineering
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• v.17 no.6
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• pp.1061-1068
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• 2012

This paper deals with a noise reduction algorithm which uses the binary masking approach in the time-frequency domain to improve speech intelligibility. In the binary masking approach, the noise-corrupted speech is decomposed into time-frequency units. Noise-dominant time-frequency units are removed by setting the corresponding binary masks as "0"s and target-dominant units are retained untouched by assigning mask "1"s. We propose a binary mask estimation by comparing the local signal-to-noise ratio (SNR) to a threshold. The local SNR is estimated by a training-based approach. An optimal threshold is proposed, which is obtained from observing the distribution of the training database. The proposed method is evaluated by normal-hearing subjects and the intelligibility scores are computed by counting the number of words correctly recognized.

• The Journal of the Acoustical Society of Korea
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• v.36 no.3
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• pp.218-227
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• 2017

In this paper, an algorithm to calculate both bearing and distance error for target detection and localization is proposed using the Cramer Rao lower bound to estimate the minium variance of their error in DOA (Direction Of Arrival) estimation. The performance of arrays in detection and localization depends on the accuracy of DOA, which is affected by a variation of SNR (Signal to Noise Ratio). The SNR is determined by sonar parameters such as a SL (Source Level), TL (Transmission Loss), NL (Noise Level), array shape and beam steering angle. For verification of the suggested method, a Monte Carlo simulation was performed to probabilistically calculate the bearing and distance error according to the SNR which varies with the relative position of the target in space and noise level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10B
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• pp.1360-1366
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• 2001

이 논문에서는, 잡음환경에서의 음성 향상을 위해서 일반적인 주파수 차감법에 기반한 새로운 형태의 방법을 제안한다. 기존의 방법들이 각각의 주파수 성분에 대해 잡음 및 음성스펙트럼을 추정하는데 비해, 본 논문에서는 주파수 대역을 여러 개의 서브밴드로 대역을 나누어 각각의 서브밴드에 대해서 잡음 및 음성의 스펙트럼을 추정한다. 본 논문에서는 잡음 스펙트럼을 추정하기 위하여 최소추적(Minima Tracking) 방법을 선택하였고, 필터링 방법으로는 스펙트럼 차감법에 기반한 Mel-Scaled 필터뱅크를 이용한 새로운 방법을 제안하였다. 모의실험결과, 기존의 방법들에 비해 음성구간에서의 SNR의 향상정도는 입력 SNR이 -10∼4dB의 범위에서 향상된 결과를 얻었다. 또한 전 구간에 대해서도 다른 알고리즘들 보다 향상된 결과를 얻었다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2005.04a
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• pp.437-440
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• 2005

본 논문에서는 고속의 적웅 지각 필터에서 잡음 과추정으로 인해서 발생하는 불필요한 반복 계산 및 결과 신호의 SNR 성능 저하를 개선시키는 방법을 제안한다. 적응 지각 필터를 고속연산이 가능하도록 개선하는 과정에서 시간적인 측면에서는 많은 성능의 개선이 있었지만 음질 개선 과정에서 과추정된 잡음의 적용에 의한 성능 저하가 발생하였다. 제안하는 시스템에서는 적웅 지각 필터의 임계값을 조정하고, 임계값이외에 발생하는 잡음 과추정에 대해서 실험적으로 필터 반복 연산량 제한을 통해 향상된 결과를 얻었다. 이 시스템에서 필터 반복 연산량은 입력 구간의 신호에 따라 적응적으로 제한된다. 제안된 알고리즘의 개선 확인을 위해서 감소된 반복 연산량과 SNR 개선량을 측정하여 기존의 방법과 비교하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.378-381
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• 2006

This paper proposes the method of a channel estimation for MIMO-OFDM with ISI. The proposed method uses a new special training sequence to obtain a constant PAR in OFDM and to remove the effect of ISI on channel estimation. Using this training sequence, we are able to avoid a singular problem in matrix. As a result of simulation, we are able to assure that the proposed system inclosed the performance in MSE of estimated channel by more than 30dB than a conventional method if SNR is high.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.58-66
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• 2001

In this paper, a computationally efficient time delay and doppler estimation algorithm is proposed for active sonar with Linear Frequency Modulated (LFM) signal. To reduce the computational burden of the conventional estimation algorithm, an algebraic equation is used which represents the relationship between the time delay and doppler in cross-ambiguity function of the LFM signal. The algebraic equation is derived based on the Fast maximum Likelihood (FML) method. Using this algebraic relation, the time delay and doppler are estimated with two 1-D search instead of the conventional 2-D search. The estimation errors of the proposed algorithm are analyzed for various SNR's. The simulation result demonstrates the good performance of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.44-52
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• 1989

The UCERSS (Unit Circle Eigendecomposition Rational Signal Subspace) algorithm has extended MUSIC (MUltiple Signal Classification ) by using eigendecomposition on the unit circle in order to estimate incident angles of multiple wide band signals. The purpose of this thesis is to further extend the UCERSS to be able to estimate the direction of arrivals of multiple wide band signals in very low SNR . The wide band ESPRIT (Estimation of Signal Parameter via Rotational Invariance Technique) uses covariance difference matrices to reduce noise components. In this paper the wide band ESPRIT which combines the ideas of UCERSS and ESPRIT Is proposed. Computer simulation results Indicate that the performances of the proposed approaches are superior to those of the UCERSS in very low SNR.