• 한국정보통신학회논문지
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• 제19권5호
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• pp.1077-1083
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• 2015

본 논문에서는 다중 셀 상향링크 네트워크에서 합 용량을 크게 개선할 수 있는 에너지 효율적 기회적 간섭 정렬 기술을 소개한다. 각 사용자는 요구되는 신호 질을 만족하면서 타 셀 기지국에게 발생하는 간섭량을 최소화하는 측면에서 최적의 송신 벡터 설계 및 전력 제어를 수행한다. 주요 결과로써, 줄어든 간섭 레벨로 인하여 제안하는 기회적 간섭 정렬 기술은 전력 제어가 없는 기존 기회적 간섭 정렬보다 대부분의 신호 대 잡음 비 영역에서 더욱 높은 합 용량을 취득할 수 있음을 보인다. 뿐만 아니라, 제안하는 기회적 간섭 정렬 기술과 함께 수신 단에서 zero-forcing 검파 및 minimum mean square error (MMSE) 검파기를 사용할 경우 둘 사이 성능 비교를 수행하고, MMSE 검파 기반 기회적 간섭 정렬 기술이 우수한 성능을 취득할 수 있음을 보인다.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2002년도 11월 학술대회지
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• pp.187-190
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• 2002

In this paper, a speech enhancement system using microphone array with MMSE-STSA (Minimum Mean Square Error-Short Time Spectral Amplitude) estimator based post-processing is proposed. Speech enhancement is first carried out by conventional delay-and-sum beamforming (DSB). A new MMSE-STSA estimator is then obtained by refining MMSE-STSA estimators from each microphone, which is applied to the output of conventional DSB to obtain additional speech enhancement. Computer simulation for white and pink noises show that the proposed system is superior to other approaches.

• International Journal of Internet, Broadcasting and Communication
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• 제4권1호
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• pp.18-22
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• 2012

In this paper, we propose an equivalent Wiener-Hopf equation. The proposed algorithm can obtain the weight vector of a TDL(tapped-delay-line) filter and the error simultaneously if the inputs are orthogonal to each other. The equivalent Wiener-Hopf equation was analyzed theoretically based on the MMSE(minimum mean square error) method. The results present that the proposed algorithm is equivalent to original Wiener-Hopf equation. In conclusion, our method can find the coefficient of the TDL (tapped-delay-line) filter where a lattice filter is used, and also when the process of Gram-Schmidt orthogonalization is used. Furthermore, a new cost function is suggested which may facilitate research in the adaptive signal processing area.

• 말소리와 음성과학
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• 제6권2호
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• pp.29-34
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• 2014

In the Vector Taylor Series (VTS)-based noisy speech recognition methods, Hidden Markov Models (HMM) are usually trained with clean speech. However, better performance is expected by training the HMM with noisy speech. In a previous study, we could find that Minimum Mean Square Error (MMSE) estimation of the training noisy speech in the log-spectrum domain produce improved recognition results, but since the proposed algorithm was done in the log-spectrum domain, it could not be used for the HMM adaptation. In this paper, we modify the previous algorithm to derive a novel mathematical relation between test and training noisy speech in the cepstrum domain and the mean and covariance of the Multi-condition TRaining (MTR) trained noisy speech HMM are adapted. In the noisy speech recognition experiments on the Aurora 2 database, the proposed method produced 10.6% of relative improvement in Word Error Rates (WERs) over the MTR method while the previous MMSE estimation of the training noisy speech produced 4.3% of relative improvement, which shows the superiority of the proposed method.

• 한국인터넷방송통신학회논문지
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• 제9권4호
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• pp.47-52
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• 2009

Source로부터 송신된 신호는 무선 채널을 통하여 Destination에 전달된다. 하지만 이동하는 Destination이 Source의 Coverage를 벗어난 경우나 비록 Source의 Coverage 내의 음영 지역에 Destination이 존재하는 경우, Destination은 Source로부터 송신된 신호를 수신할 수 없고 통신을 할 수 없게 된다. 따라서 이러한 문제점을 극복하기 위해 중계기가 사용된다. 이와 같이 중계기를 사용하는 시스템을 다중 홉 중계 (Multi-hop Relay) 시스템이라 한다. 그런데 다중 홉 중계 시스템에서는 서로 다른 시스템용 중계기의 Coverage가 겹치는 경우가 발생할 수 있고, 이 부분에 Destination이 존재하는 경우 Destination에는 간섭이 발생한다. 본 논문에서는 다중 홉 중계 시스템에서 발생 할 수 있는 동일 채널 간섭 (CCI : Co-Channel Interference) 제거 방법에 관해 연구하였다. 간섭 제거 방법은 우선 Zero Forcing (ZF) 또는 Minimum Mean Square Error (MMSE) 개념을 적용한 선형 수신기를 이용하여 간섭을 제거한 후, 정렬된 연속간섭제거 (OSIC : Ordered Successive Interference Cancellation) 알고리즘을 이용하여 추가적인 다이버시티 이득을 얻고 간섭 제거 성능을 향상시킨다. 무선 채널은 레일레이 (Rayleigh) 페이딩 채널을 고려하여 모의 실험을 하였으며, 시스템 성능은 비트 오류 확률 (Bit Error Probability) 측면에서 분석되었다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 통신소사이어티 추계학술대회논문집
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• pp.20-23
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• 2003

A reduced-state sequence estimation(RSSE) for trellis-coded (TC) 8PSK/cyclic prefixed single carrier(CPSC) with minimum mean-square error-liner equalization(MMSE-LE) on frequency-selective Rayleigh fading channels is proposed. The Viterbi algorithm (VA) is used to search for the best path through the reduced-state trellis combined equalization and TCM decoding. The symbol error probability of the proposed scheme is confirmed by computer simulation.

• The Journal of the Acoustical Society of Korea
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• 제15권1E호
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• pp.34-38
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• 1996

In this paper, a new fast voronoi divider for vector quantization (VQ) is introduced, which results from Theorem that the nearest vectors in the sense of minimum mean square error(MMSE) have almost the same mean values of their elements. An improved splitting method for a VQ codebook design using the fast voronoi divider is also presented. Experimental results show that the new method reduces the complexity of training a VQ codebook several times with a high signal to noise ratio(SNR) using an appropriate extensive parameter of codebook.

• ETRI Journal
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• 제38권3호
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• pp.510-517
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• 2016

Recent developments in the field of separation of mixed signals into music/voice components have attracted the attention of many researchers. Recently, iterative kernel back-fitting, also known as kernel additive modeling, was proposed to achieve good results for music/voice separation. To obtain minimum mean square error (MMSE) estimates of short-time Fourier transforms of sources, generalized spatial Wiener filtering (GW) is typically used. In this paper, we propose an advanced music/voice separation method that utilizes a generalized weighted ${\beta}$-order MMSE estimation (WbE) based on iterative kernel back-fitting (KBF). In the proposed method, WbE is used for the step of mixed music signal separation, while KBF permits kernel spectrogram model fitting at each iteration. Experimental results show that the proposed method achieves better separation performance than GW and existing Bayesian estimators.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2001년도 추계학술발표대회 논문집 제20권 2호
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• pp.125-128
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• 2001

본 연구에서는 음성신호의 왜곡에 대해 음성 부재 확률을 고려한 MMSE(Minimum Mean Square Error) STSA(Short-Time Spectral Amplitude Estimator)를 전처리기로 도입하여 HMM(Hidden Markov Model)에 기반 한 음성인식시스템의 인식성능을 평가하였다. 음성인식 시스템의 실시간 구현을 고려하여, MMSE STSA 기법을 음성개선을 위한 전처리기로 사용할 때 MMSE STSA의 이득계산 과정에서 많은 계산량이 요구되는 modified Bessel 함수를 근사 화하여 사용하였다.

• International Journal of Contents
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• 제15권4호
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• pp.27-35
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• 2019

In this paper, we propose an advanced Minimum Mean Square Error (MMSE) channel estimation method for IEEE 802.11p/Wireless Access in Vehicular Environments (WAVE) systems. To improve the performance of MMSE method, we apply the Weighted Sum using Update Matrix (WSUM) scheme to the step of calculating the instantaneously estimated channel and then, a time domain selectively averaging method is applied after the WSUM scheme. Based on that, the accuracy of instantaneously estimated channel increases and then, the accuracy of auto covariance matrix also increases. Consequently, we can achieve the performance gain over the conventional MMSE method. Through simulations based on the IEEE 802.11p standard, it is confirmed that the proposed scheme can outperform the existing channel estimation schemes.