• 대한전기학회:학술대회논문집
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• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 B
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• pp.548-551
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• 2003

The adaptive filter is proposed for removing EOG from measured EEG on the frontal lobe. The proposed adaptive filter has been implemented and the feedback output control algorithm has been employed to control the alpha wave ratio on the basis of TMS320C31 DSP board with the on-line and real time performance. The feedback algorithm controls the input voltage of stimulating devices on the portable bio-feedback system. The EEG data are acquired at the $F_{p1}$ and $F_{p2}$ localization and are processed by the proposed adaptive filter. We demonstrated that the proposed adaptive filter could effectively remove EOG from the measured EEG on the frontal lobe and the feedback algorithm is proper to control the output voltage of DSP board using the ratio of the alpha wave.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2008년도 심포지엄 논문집 정보 및 제어부문
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• pp.238-239
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• 2008

In this paper, active noise control (ANC) of a Volterra system with a nonlinear secondary path is proposed in the presence of a linear acoustic feedback, whereby the conventional ANC of a linear system with online acoustic feedback-path modeling is further extended to ANC of a Volterra system with a linear acoustic feedback path. In particular, the proposed ANC system consists of two adaptive Volterra filters (for nonlinear noise control and nonlinear adaptive noise cancellation) and one feedback-path modeling filter. Simulation results show that the proposed approach yields more effective reduction of disturbances arising from the acoustic feedback, in addition to high nonlinear ANC performance.

• 전자공학회논문지SC
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• 제42권4호
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• pp.45-50
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• 2005

디지털 보청기 시스템에서는 소형화, 고이득 증폭이 요구되는데 이로 인한 문제점 중의 하나가 하울링(howling)이라는 음향의 궤환이다. 본 논문에서는 시간지연과 시변 상관성 제거 필터를 이용하여 보청기에서 사용 가능한 음향 궤환 제거 알고리즘을 제안하였다. 제안한 알고리즘은 적응형 필터 구조를 기본으로 음향 궤환 신호 제거 효과를 증대시키기 위해 시간 지연과 상관성 제거 필터를 삽입하였다. 상관성 제거 필터로 사용된 전대역 통과 필터는 저주파 변조기를 이용하여 시변 특성을 갖도록 구현 하였다. 컴퓨터 시뮬레이션 결과를 통하여 제안한 알고리즘의 궤환 제거 특성이 우수함을 확인하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(1)
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• pp.69-72
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• 2002

In this paper, adaptive minimum mean-squared error (MMSE) multiuser detector combined with decision-feedback detector (DFD) is considered fur near-far resistant DS-CDMA system. To provide a reliable input to the adaptive MMSE detector, multiple-access interference (MAI) is regenerated using bit estimates from DFD and subtracted from the received signal. In the adaptive MMSE detector, the effect of the imperfect cancellation is compensated by a least mean square (LMS) algorithm. Through the numerical results, it is shown that, in a near-far situation, the proposed scheme provides superior performance to the matched filter (MF) receiver, adaptive MMSE detector, and DFD in terms of the bit error rate (BER).

• 한국전자파학회논문지
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• 제19권9호
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• pp.1058-1064
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• 2008

본 논문에서는 RF(Radio Frequency) 중계기에서 그룹화 CMA(Constant Modulus Algorithm)와 LMS(Least Mean Square) 알고리즘을 이용하여 적응 필터를 적용시킨 새로운 혼합 간섭 제거기를 제안한다. 송신 안테나에서 수신안테나로 궤환되는 신호는 수신 시스템의 성능을 저하시킨다. 제안한 간섭 제거기는 그룹화 CMA 알고리즘 간섭 제거 기법을 적용시키기 때문에 기존 구조보다 나은 채널 적응 성능과 낮은 MSE(Mean Square Error)을 가진다. 이 구조는 기존 비선형 간섭 제거기에 비해 같은 MSE(Mean Square Error)에 대한 반복수와 하드웨어 복잡도를 줄여준다. 즉, 제안한 알고리즘은 LMS 알고리즘에 비해 평균 자승 에러가 적응 상수에 따라 2.5 dB 또는 4 dB 정도 낮은 값을 보였다. 또한, VSS(Variable Step Size)-LMS 알고리즘에 비해 수렴 속도가 빠르고, 비슷한 평균 자승 에러를 가진다.

• 한국정보통신설비학회:학술대회논문집
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• 한국정보통신설비학회 2007년도 학술대회
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• pp.440-444
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• 2007

As the mobile communication service is widely used, the demand for wireless repeaters is rapidly increasing because of the easiness of extending service areas. But a wireless repeater has a problem that the output of the transmit antenna is partially fed back to the receive antenna, which results in feedback interference. In this paper, we propose a new varable step-size LMS algorithm, which utilizes correlation between reference and error signals to adjust the step sizes, for cancelling the feedback interference signals in the WCDMA repeater under time-varying multi-path channels. The proposed algorithm was investigated through computer simualation by being applied to the time-varying channels. The simulation results demonstrated that the proposed one is superior to the conventional ones in terms of cancelation perormance.

• 소음진동
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• 제2권2호
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• pp.107-116
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• 1992

An adaptive signal processing technique is implemented for the active noise cancellation of the plane acoustic wave propagating in a duct. To avoid the instability caused by the acoustic feedback from the control speaker to the detect microphone, an off-line modeling of the acoustic feedback plant is done using the FIR filter. Auxiliary path required for the filtered-x LMS algorithm is modeled as well. Before going into the experiments, a simulation is carried out under the same conditions with experiments. The simulation shows that the longer the length of the adaptive filter is, the better the results are achieved. Experiments have been carried out at lower audio frequency range (50 - 400Hz), and the results are in good agreements with those of simulation study. As a results of this adaptive noise control, around 50dB is reduced for a pure tone noise, and for a bandlimited noise with the bandwidth of 316Hz, a maximum of 30dB noise reduction is attained.

• Journal of Communications and Networks
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• 제7권3호
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• 한국정보전자통신기술학회논문지
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• 제9권2호
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• pp.205-212
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• 2016

본 논문에서는 Matlab에서 제공하는 Simulink를 활용하여 다채널 디지털 보청기 알고리즘 개발 플랫폼의 구현을 제안하였다. 디지털 보청기는 난청자의 원활한 의사소통을 돕는 의료기구로 그 중요성이 날로 증가하고 있다. 특히 다채널 디지털 보청기는 난청자의 주파수 별 청력 손실 정도에 따른 보상이 가능해 고막의 손상을 최소화하는 동시에 보청기 사용자에게 적합한 증폭이 가능해진다. 본 논문에서 구현한 개발 플랫폼은 WOLA 필터뱅크를 이용해 입력 신호의 분석 및 합성이 이루어지며 광역동범위압축(Wide Dynamic ragne compression) 기반의 난청 보상 알고리즘, 적응 필터를 이용한 음향 궤환 제거 알고리즘(Adaptive feedback cancellation)을 포함한다. Simulink를 이용한 개발 플랫폼에서 각 블록의 파라미터를 설정 할 수 있고 블록별 결과가 확인이 가능하다. 이를 이용해 기계어 코딩 전 단계에서 알고리즘 테스트가 가능하기 때문에 보청기 알고리즘의 개발 시간이 단축 가능하고 계산량 및 성능 최적화가 가능해졌다.

• ETRI Journal
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• 제37권5호
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• pp.906-916
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• 2015

High-speed I/O channels require adaptive techniques to optimize the settings for filter tap weights at decision feedback equalization (DFE) read channels to compensate for channel inter-symbol interference (ISI) and crosstalk from multiple adjacent channels. Both ISI and crosstalk tend to vary with channel length, process, and temperature variations. Individually optimizing parameters such as those just mentioned leads to suboptimal solutions. We propose a joint optimization technique for crosstalk cancellation (XTC) at DFE to compensate for both ISI and XTC in high-speed I/O channels. The technique is used to compensate for between 15.7 dB and 19.7 dB of channel loss combined with a variety of crosstalk strengths from $60mV_{p-p}$ to $180mV_{p-p}$ adaptively, where the transmit non-return-to-zero signal amplitude is a constant $500mV_{p-p}$.