• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.141-148
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• 2006

This paper deals with the performance improvement by eliminating loop interference signal and inserting adaptive equalizer for phase compensation in OFDMA based Wibro relay station. The Wibro relay station is used for the extension of communication service area and for throughput improvement of base station. The loop interference is important factor of performance determination of relay station when transmitter and receiver is very closely located. In order to design interference canceller, we generated base-band OFDMA signal and then transmitted the signal along with pilot tones alined with two different combinations for training mode. And then, we generated received fading signal due to the loop interference added noise to the received signal. In the receiver, the transmitted signal is recovered by elimination of the interference signal with channel estimate and compensating phase by adaptive equalizer. The performance improvement was verified by computer simulation which show channel estimation, constellation of signal and BER characteristics according to the variation of SNR ratio.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4C
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• pp.305-311
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• 2008

In this paper, we proposed an algorithm for adaptive noise cancellation (ANC) using the variable step size normalized least mean square (VSSNLMS) in real-time automobile environment. As a basic algorithm for ANC, the LMS algorithm has been used for its simplicity. However, the LMS algorithm has problems of both convergence speed and estimation accuracy in real-time environment. In order to solve these problems, the VSSLMS algorithm for ANC is considered in nonstationary environment. By computer simulation using real-time data acquisition system(USB 6009), VSSNLMS algorithm turns out to be more effective than the LMS algorithm in both convergence speed and estimation accuracy.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.117-121
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• 1991

본 논문은 견인선(tow-ship)에서 발생하는 자체 잡음을 제거하여 수중 신호처리 시스템에서 표적 탐지(target detection)와 표적 식별(target identification) 등의 성능 향상을 위하여 표적 방향으로 형성된 빔의 출력을 원시 입력신호(primary input)로 사용하고 견인선 방향으로 형성된 빔의 출력을 참고 입력신호(reference input)로 사용한 적응 잡음 제거기(adaptive noise canceller)에 대해 연구하였다. 잡음 제거를 위해 사용되는 계수들은 LMS(Least Mean Square) 알고리듬을 이용하여 조정하였다. 컴퓨터 시뮬레이션을 통하여 TDL(Tapped-Delay Line) 구조와 LAT(LATtice) 구조를 갖는 적응 잡음 제거기 성능을 여러 가지 환경에서 비교, 관찰하였다. 두 알고리듬을 사용할 경우, 자체 잡음이 어떠한 형태로 나타나더라도 제거시킬 수 있음을 보여 주었으나 고유값 분포율(eigenvalue spread ratio)이 큰 경우에는 LMS-LAT가 LMS-TDL보다 수렴 속도뿐만 아니라 성능면에서도 우수함을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.17-25
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• 2001

Traditional adaptive noise cancelling methods rely their performance on various interfering parameters, such as convergence speed, tracking ability, numerical stability, relative frequency characteristics between target and reverberation signals, and activity of the target. In this paper, an adaptive noise cancelling method is suggested, which Provides a successful tradeoff mon these factors. It is designed to work on the transform domain, adopts the Gram-Schmidt orthogonalization process, and is implemented by the escalator algorithm. The transform domain approach supports a tradeoff between the convergence speed and numerical cost. The proposed method is verified by applying a real-data collected in the shallow waters off the east coasts of korea. It is shown that it has a good reverberation-rejection capability even for the target signal with adjacent frequency components to those of the reverberation, and its performance is invariant for the activity of the target.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1A
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• pp.72-78
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• 2006

It is impossible to provide full diversity and full rate simultaneously using more than two transmit antennas in transmit diversity system. To do this, simple interference cancellation scheme and transmit power allocation scheme have been proposed, recently. But the former has increased noise power and the latter has increased interference which is induced by other channel in fading channel. In this paper, we propose an adaptive transmit power allocation algorithm to minimize the estimation error in the channel environments which have different fading levels each other and to improve the system performance.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2E
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• pp.56-67
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• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2388-2396
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• 2012

This paper is on an simplified orthogonal projection method which cancel the acoustic echo signals in the stereo acoustic echo canceller. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances for signals with the high auto-correlation, and has small computational quantities. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it was proved that the stereo acoustic echo canceller adopting the proposed algorithm shows about 3dB more high ERLE than the NLMS algorithm for the white noise signals, and 5dB for the colored voice signals.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.433-438
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• 2017

Containing an Angle-of-Arrival(: AOA) estimation and interference suppression techniques, an adaptive beamformer is one of core techniques for the Signal Intelligence(: SIGINT) which collect various intelligence utilizing cutting edge devices including the radar and satellite. It generates a beam with the directivity in a corresponding direction, to efficiently receive a signal from the specific direction, using antenna array. In this paper, we present the received signal model including interference signals and noise, which can be applied to an input of the signal intelligence satellite system equipped with the AOA estimation and the interference cancellation techniques, and analysis the characteristics of various signals, which can be included in the proposed received signal model. This proposed signal model can be directly applied to the performance evaluation for a variety of beamforming techniques. Also, we verify the spectrum characteristic of the presented received signal model in the frequency domain through computer simulation examples.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.67 no.5
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• pp.675-679
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• 2018

A novel heart rate estimation algorithm is presented based on normalized least-mean-square (NLMS) algorithm. This paper presented a three-step processing scheme for estimating heart rate from PPG signal with motion artifacts. The proposed active noise cancellation algorithm has low computational complexity compared to the NLMS algorithm. Experimental results show that the proposed algorithms perform similar with the previous algorithm under motion artifact noises.