• Title/Summary/Keyword: Minimum Mean Square Error(MMSE)

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A Feasibility Study on Opportunistic Interference Alignment: Improved Energy Efficiency via Power Control (기회적 간섭 정렬의 실현 가능성 연구: 전력 제어를 통한 에너지 효율성 개선)

  • Shin, Won-Yong;Yoon, Jangho
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.19 no.5
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    • pp.1077-1083
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    • 2015
  • In this paper, we introduce an energy-efficient opportunistic interference alignment (OIA) scheme that greatly improves the sum-rates in multi-cell uplink networks. Each user employs optimal transmit vector design and power control in the sense of minimizing the amount of generated interference to other-cell base stations while satisfying a required signal quality. As our main result, it is shown that owing to the reduced interference level, the proposed OIA schemes attains larger sum-rates than those of OIA with no power control for almost all signal-to-noise ratio regions. In addition, when both zero-forcing and minimum mean square error (MMSE) detectors are employed at the receiver along with the OIA scheme, it is shown that the OIA scheme with MMSE detection shows superior performance.

Speech Enhancement Using Microphone Array with MMSE-STSA Estimator Based Post-Processing (MMSE-STSA 추정치에 기반한 후처리를 갖는 마이크로폰 배열을 이용한 음성 개선)

  • Kwon Hong Seok;Son Jong Mok;Bae Keun Sung
    • Proceedings of the KSPS conference
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    • 2002.11a
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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.

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MMSE based Wiener-Hopf Equation

  • Cho, Juphil;Lee, Il Kyu;Cha, Jae Sang
    • International Journal of Internet, Broadcasting and Communication
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    • v.4 no.1
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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.

Noise Robust Speech Recognition Based on Noisy Speech Acoustic Model Adaptation (잡음음성 음향모델 적응에 기반한 잡음에 강인한 음성인식)

  • Chung, Yongjoo
    • Phonetics and Speech Sciences
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    • v.6 no.2
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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.

A Study on the Efficient Interference Cancellation for Multi-hop Relay Systems (다중 홉 중계 시스템에서 효과적인 간섭 제거에 관한 연구)

  • Kim, Eun-Cheol;Cha, Jae-Sang;Kim, Seong-Kweon;Lee, Jong-Joo;Kim, Jin-Young;Kang, Jeong-Jin
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.9 no.4
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    • pp.47-52
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    • 2009
  • The transmitted signal from a source is transmitted to a destination through wireless channels. But if the mobile destination is out of the coverage of the source or exists in the shady side of the coverage, the destination can not receiver the signal from the source and they can not maintain communication. In order to overcome these problems, we adopt relays. A system employing relays is a multi-hop relay system. In the multi-hop relay system, coverages of each relay that is used for different systems can overlap each other in some place. When there is a destination in this place, interference occurs at the destination. In this paper, we study on the efficient co-channel interference (CCI) cancellation algorithm. In the proposed strategy, CCI is mitigated by zero forcing (ZF) or minimum mean square error (MMSE) receivers. Moreover, successive interference cancellation (SIC) with optimal ordering algorithm is applied for rejecting CCI efficiently. And we analyzed and simulated the proposed system performance in Rayleigh fading channel. In order to justify the benefit of the proposed strategy, the overall system performance is illustrated in terms of bit error probability.

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Reduced-state sequence estimation for trellis-coded 8PSK/cyclic prefixed single carrier (트렐리스 부호화된 8PSK/CPSC를 위한 RSSE 방식)

  • 고상보;강훈철;좌정우
    • Proceedings of the IEEK Conference
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    • 2003.11c
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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.

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Fast Voronoi Divider for VQ Code book Designs

  • Jang, Gang-Yi;Choi, Tae-Young
    • The Journal of the Acoustical Society of Korea
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    • v.15 no.1E
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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.

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Music/Voice Separation Based on Kernel Back-Fitting Using Weighted β-Order MMSE Estimation

  • Kim, Hyoung-Gook;Kim, Jin Young
    • ETRI Journal
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    • v.38 no.3
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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.

Comparison of Recognition Performance for Preprocessing Method of USE STSA with Approximated Modified Bessel Function (Modified Bessel 함수 근사화를 적용한 MMSE STSA 전처리 기법의 음성인식 성능 비교)

  • Son Jong Mok;Kim Min Sung;Bae Keun Sung
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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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.

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Advanced Channel Estimation Method for IEEE 802.11p/WAVE System

  • Jang, DongSeon;Ko, Kyunbyoung
    • International Journal of Contents
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    • v.15 no.4
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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.