• Journal of the Korean Data and Information Science Society
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• 제20권3호
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• pp.577-585
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• 2009

So far in a nonparametric regression model one of the interesting problems is estimating the error variance. In this paper we propose an estimator of the mean of the squared functions which is the numerator of SNR (Signal to Noise Ratio). To estimate SNR, the mean of the squared function should be firstly estimated. Our focus is on estimating the amplitude, that is the mean of the squared functions, in a nonparametric regression using a simple linear regression model with the quadratic form of observations as the dependent variable and the function of a lag as the regressor. Our method can be extended to nonparametric regression models with multivariate functions on unequally spaced design points or clustered designed points.

• ETRI Journal
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• 제28권6호
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• pp.761-769
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• 2006

In this paper, we address the problem of closely spaced source localization using sensor array processing. In particular, the performance efficiency (measured in terms of the root mean square error) of the unconditional maximum likelihood (UML) algorithm for estimating the direction of arrival (DOA) of near-field sources is evaluated. Four parameters are considered in this evaluation: angular separation among sources, signal-to-noise ratio (SNR), number of snapshots, and number of sources (multiple sources). Simulations are conducted to illustrate the UML performance to compute the DOA of sources in the near-field. Finally, results are also presented that compare the performance of the UML DOA estimator with the existing multiple signal classification approach. The results show the capability of the UML estimator for estimating the DOA when the angular separation is taken into account as a critical parameter. These results are consistent in both low SNR and multiple-source scenarios.

• The Journal of the Acoustical Society of Korea
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• 제29권2E호
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• pp.86-99
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• 2010

This paper analyzes the performance of various single channel speech enhancement algorithms when they are applied to automatic speech recognition (ASR) systems as a preprocessor. The functional modules of speech enhancement systems are first divided into four major modules such as a gain estimator, a noise power spectrum estimator, a priori signal to noise ratio (SNR) estimator, and a speech absence probability (SAP) estimator. We investigate the relationship between speech recognition accuracy and the roles of each module. Simulation results show that the Wiener filter outperforms other gain functions such as minimum mean square error-short time spectral amplitude (MMSE-STSA) and minimum mean square error-log spectral amplitude (MMSE-LSA) estimators when a perfect noise estimator is applied. When the performance of the noise estimator degrades, however, MMSE methods including the decision directed module to estimate a priori SNR and the SAP estimation module helps to improve the performance of the enhancement algorithm for speech recognition systems.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제10권11호
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• pp.5365-5380
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• 2016

Estimation of signal-to-noise ratio (SNR) is an important problem in wireless communication systems. It has been studied for various constellation types and channels using different estimation techniques. Maximum likelihood estimation is a technique which provides efficient and in most cases unbiased estimators. In this paper, we have applied maximum likelihood estimation for systems employing square or cross QAM signals which are undergoing slow flat Rayleigh fading. The problem has been considered under various scenarios like data-aided (DA), non-data-aided (NDA) and partially data-aided (PDA) and the performance of each type of estimator has been evaluated and compared. It has been observed that the performance of DA estimator is best due to usage of pilot symbols, with the drawback of greater bandwidth consumption. However, this can be catered for by using partially data-aided estimators whose performance is better than NDA systems with some extra bandwidth requirement.

• 디지털융복합연구
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• 제10권10호
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• pp.277-282
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• 2012

음성 인식 시스템은 다양하게 변화하는 환경 잡음에 빠르게 적응할 수 없어서 인식 성능을 저하시키는 요인이 된다. 본 논문에서는 평균 예측 LMS 알고리즘을 이용하여 반향 잡음에 강인하게 하는 방법으로 HMM 학습 모델을 구성하는 방법을 제안하였으며, 변화하는 반향 잡음에 적응하도록 HMM 학습 모델을 구성하여 인식 성능을 평가하였다. 실험 결과 변화하는 환경 잡음을 제거하여 얻은 음성의 SNR은 평균 3.1dB이 향상되었고 인식률은 3.9% 향상되었다.

• 디지털융복합연구
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• 제10권11호
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• pp.377-382
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• 2012

본 논문은 반향 제거 평균 예측 LMS 알고리즘을 이용하여 반향 잡음에 강인한 연속 음성 인식 모델인 CHMM 모델을 구성하는 방법을 제안하였다. 변화하는 반향 잡음에 적응하고 연속 음성 인식 성능 향상을 위한 반향 잡음 제거 평균 예측 LMS 알고리즘을 이용하여 CHMM 모델을 구성하였다. 제안한 알고리즘에 의해 구성된 CHMM 모델에 대하여 연속 인식 성능을 평가하였다. 실험 결과 변화하는 환경 잡음을 제거하여 얻은 음성의 SNR은 평균 1.93dB이 향상되었고 연속 음성의 인식률은 2.1% 향상되었다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2005년도 추계종합학술대회
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• pp.175-178
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• 2005

In this paper, a carrier frequency offset (CFO) estimator is proposed for the interleaved OFDMA uplink systems. It is based on the estimation of signal parameters via rotational invariance technique (ESPRIT). Compared with the Cao's estimator, the proposed estimator has low computational complexity. Simulation results demonstrate that the proposed estimator performs better than Cao's estimator at the relatively low SNR region. Hence, the proposed estimator is more applicable to the practical environments than the Cao’s estimator.

• 한국군사과학기술학회지
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• 제9권4호
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• pp.32-38
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• 2006

In this paper a sequence estimator of CDMA communication system is suggested. A sequence estimator uses Galois Field operation. A sequence estimator can provide another CDMA pilot signal which is un-modulated spreaded signal. A estimated sequence signal and received signal have no correlation. Tow signals can be summed using MRC(maximal ratio combine) method. The stronger signal can be added as a larger ratio, but the weaker signal can be added as a smaller ratio. We can distinguish strong signal using SNR estimator. Therefore it is possible to receive an additional pilot signal, and to support more reliable communications by using sequence estimator.

• Investigative Magnetic Resonance Imaging
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• 제20권4호
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• pp.215-223
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• 2016

Purpose: The management of metal-induced field inhomogeneities is one of the major concerns of distortion-free magnetic resonance images near metallic implants. The recently proposed method called "Slice Encoding for Metal Artifact Correction (SEMAC)" is an effective spin echo pulse sequence of magnetic resonance imaging (MRI) near metallic implants. However, as SEMAC uses the noisy resolved data elements, SEMAC images can have a major problem for improving the signal-to-noise ratio (SNR) without compromising the correction of metal artifacts. To address that issue, this paper presents a novel reconstruction technique for providing an improvement of the SNR in SEMAC images without sacrificing the correction of metal artifacts. Materials and Methods: Low-rank approximation in each coil image is first performed to suppress the noise in the slice direction, because the signal is highly correlated between SEMAC-encoded slices. Secondly, SEMAC images are reconstructed by the best linear unbiased estimator (BLUE), also known as Gauss-Markov or weighted least squares. Noise levels and correlation in the receiver channels are considered for the sake of SNR optimization. To this end, since distorted excitation profiles are sparse, $l_1$ minimization performs well in recovering the sparse distorted excitation profiles and the sparse modeling of our approach offers excellent correction of metal-induced distortions. Results: Three images reconstructed using SEMAC, SEMAC with the conventional two-step noise reduction, and the proposed image denoising for metal MRI exploiting sparsity and low rank approximation algorithm were compared. The proposed algorithm outperformed two methods and produced 119% SNR better than SEMAC and 89% SNR better than SEMAC with the conventional two-step noise reduction. Conclusion: We successfully demonstrated that the proposed, novel algorithm for SEMAC, if compared with conventional de-noising methods, substantially improves SNR and reduces artifacts.

• 한국통신학회논문지
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• 제34권3A호
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• pp.265-271
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• 2009

본 논문은 위성방송 표준인 DVB-S2 (Digital Video Broadcasting - Second Generation)의 복조기 설계에서 중요한 부분 중의 하나인 초기 주파수 추정 회로를 효율적으로 설계하는 방법을 제안한다. DVB-D2에서 초기 주파수 오차는 심볼 전송률의 20%에 해당하며 심볼 전송률이 25Msps일 경우 ${\pm}5MHz$에 달한다. 이와 같이 큰 초기주파수 오차를 추정하기 위해서는 추정 범위가 넓은 알고리즘이 요구된다. 본 논문에서는 데이터 도움 방식의 알고리즘들을 분석하고 성능 비교한 결과 M&M (Mengali & Moreli) 알고리즘이 낮은 SNR에서 우수한 추정 성능을 보여줌을 확인하였다. M&M 알고리즘을 적용한 기존의 주파수 추정 회로는 하드웨어 복잡도가 높기 때문에 자기 상관기와 역 탄젠트기의 수를 줄임으로서 전체 초기 주파수 추정기의 하드웨어 복잡도를 낮추는 방법을 제안한다. 제안된 구조는 기존의 구조에 비해 하드웨어 복잡도가 약 64.5%정도 감소하였으며 Xilinx Virtex II FPGA 검증 보드를 이용하여 제안된 구조를 검증하였다.