• Journal of Navigation and Port Research
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• v.26 no.3
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• pp.295-302
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• 2002

Currently digital communication system adapt various digital modulation schemes. All these communication systems are required to cause the minimum interference to adjacent channels, they must therefore employ the linear power amplifiers. In respect to linear power amplifiers, there are many linearization techniques. Feedforward power amplifier represent very wide bandwidth and high linearization capability. In the feedforward systems overall efficiency is reduced due to the loss of delay line. In this paper, delay filter instead of transmission delay line adapted to get more high efficiency. Experimental results showed that ACLR has improved 17.04dB which is added 2.54dB by using the delay filter.

• Journal of Biomedical Engineering Research
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• v.25 no.4
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• pp.315-321
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• 2004

In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.11
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• pp.955-966
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• 2013

We propose two feedback interference cancellers(FICs) to improve the performance of the equalization digital on-channel repeater(EDOCR) with the FIC for the ATSC DTV broadcasting system. The FIC estimates the feedback channel between Tx. and Rx. antennas of the repeater by cross-correlating the reference signal and the feedback signal. Since there is a DC pilot which ruins the white property of the ATSC DTV signals, the FIC cannot estimate the feedback channel accurately. To solve the problem, the structural method which uses an additional DC pilot free reference for feedback channel estimation and the algorithmic method based on the digital signal processing which whitens the ATSC DTV signals and performs the feedback cancellation in the whitened signal domain. Simulation results show that the proposed two FICs show better feedback cancellation performance than the conventional FIC.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.1
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• pp.102-110
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• 2017

In this paper, we have implemented an real-time adaptive feedback cancellation(AFC) algorithm that can be applied to multi-channel digital hearing aid. Multichannel digital hearing aid typically use the FFT filterbank based Wide Dynamic Range Compression(WDRC) algorithm to compensate for hearing loss. The implemented real-time acoustic feedback cancellation algorithm has one integrated structure using the same FFT filter bank with WDRC, which can be beneficial in terms of computation affecting the hearing aid battery life. In addition, when the AFC fails to operate due to nonlinear input and output, the reduction gain is applied to improve robustness in practical environment. The implemented algorithm can be further improved by adding various signal processing algorithm such as speech enhancement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.12
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• pp.103-108
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• 1998

The increase in the capacity of the digital magnetic recording systems inevitably causes severe intersymbol interference (ISI) and nonlinear distortions in the digital magnetic recording channel. In this paper, to cope with severe ISI and nonlinear distortions a neural decision feedback equalizer (NDFE) is applied to the digital magnetic recording channel - partial erasure channel model. In the performance comparison of bit error probability (or bit error ratio : BER) between the NDFE and the conventional decision feedback equalizer (DFE) via computer simulations. It has been found that as nonlinear distortions increase the NDFE has more SNR (SIgnal-to-Noise Ratio) advantage over the conventional DFE. In addition, in spite of the same recording density, as nonlinear distortions are increased, NDFE has the better performance of BER and the greater stability over conventional DFE.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.5
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• pp.887-892
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• 2008

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. To analyze the convergence characteristics of the proposed algorithm, the simulations were carried out about various input signals. And we had compared the performances of convergence for this algorithm with the ones for the NLMS algorithm. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows about 5-10 dB more high SNR than the NLMS algorithm for the colored inputs.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.193-196
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• 2008

지상파 DMB 동일 채널 중계기에서는 송신 안테나로부터 수신 안테나로 재입력되는 궤환 신호가 존재한다. 이 궤환 신호는 적절히 제거되지 않을 경우 수신 신호에 간섭으로 작용하게 되어 중계기 시스템의 안정성이 크게 떨어지게 된다. 과거에 진행된 연구에서는 궤환 채널을 추정하여 간섭을 제거시키는 궤환 신호 제거기(feedback canceller) 를 사용하여 이와 같은 문제를 해결하였다. 하지만 기존의 추정 알고리즘은 간단한 구현에 비해 수렴 성능과 속도 면에서 만족할만한 성능을 얻을 수 없었으며, 신호 간 상관도 제거를 위해 충분한 시간 지연을 해야 하는 단점이 있다. 본 논문은 궤환 채널 추정에 적합한 PNLMS (Proportionate Normalized Least Mean Squares) 적응형 알고리즘을 적용해 새로운 궤환 신호 제거 기법을 제안하였으며, 전산 모의 실험을 통해 이와 같은 문제를 효과적으로 해결할 수 있음을 보여준다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1507-1516
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• 1999

본 논문에서는 전력증폭기 선형화를 위한 Cartesian feedback 방식의 궤환 경로에서 발생하는 DC offset과 이득 및 위상 불일치를 자동적으로 보상하는 개선된 방식을 제안한다. Cartesian feedback에 의한 비선형 전력증폭기 왜곡성분의 감쇠 정도는 시스템 루프의 이득, 대역폭, 시간지연에 의해 결정된다고 알려져 있다. 그러나 궤환 경로 각 소자에서 발생하는 DC offset과 이득 및 위상의 불일치로 인하여 송신기의 출력신호에 원하지 않는 반송파 성분과 이미지 신호가 발생하여 궤환보상의 효과가 반감되는 결과를 초래한다. 본 논문에서는 디지털 신호처리 시스템 구조에서 이진 검색 (binary search) 알고리즘을 이용하여 궤환 경로에서 발생하는 DC offset과 이득 및 위상 불일치를 자동적으로 보상하는 방식을 제안하고 컴퓨터 모의실험을 통하여 제안된 방식의 성능을 분석한다. 모의실험에서 고려된 방식에 비하여 동일한 정도의 DC offset과 이득 및 위상 불일치의 보상에 걸리는 시간을 평균적으로 40% 단축할 수 있었다.