• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.5
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• pp.961-968
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• Progress in Superconductivity
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• v.5 no.1
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• pp.26-30
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• 2003

Magneto-cardiogram(MCG) signals may be highly distorted by the environmental noise, such as power-line interference, broadband white noise, surrounding magnetic noise, and baseline wondering. Several kinds of digital filters and noise cancellation methods have been designed and realized by many researchers, but these methods gave some problems that the original signal may be distorted by digital filter due to the wideband characteristics of background noise. To eliminate noise effectively without distortion of MCG signals, we performed multi-level frequency decomposition using wavelet packets and local adaptive noise cancellation in each local frequency range. In addition to the proposed wavelet filter to eliminate these various non-stationary noise elements, the local adaptive filter using the least mean square(LMS) algorithm and the soft threshold do-noising method are introduced in this paper. The signal to noise ratio(SNR) and the reconstruction square error(RSE) are calculated to evaluate the performance of the proposed method and compared with the results of the conventional wavelet filter and adaptive filter. The experimental results show that the proposed local adaptive filtering method is better than the conventional methods.

• Journal of Institute of Control, Robotics and Systems
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• v.9 no.9
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• pp.679-691
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• 2003

The band-width of disk drive servo system is rapidly increasing for the robustness to external disturbance as the track density is increasing. The increase of the band-width may cause mechanical resonance of an actuator. In disk drive servo system, a notch filter is usually used to suppress the mechanical resonance of the actuator. However, the resonance frequency differs from drive to drive because of manufacturing tolerance and varies with temperature even within a single drive. The variation of resonance frequency degrades the suppression performance of the notch filter. In this paper, we present an adaptive digital notch filter that identifies the resonance frequency of the disk drive servo actutaor precisely and adjusts automatically its center frequency. For this, we design an adaptive FIR digital filter for the estimation of the resonance frequency. The estimation filter identifies the resonance frequency adaptively using the output signal generated from the servo system, which is excited with an excitation signal including all the expected resonance frequency components. We prove mathematically the convergence of the resonance frequency estimation filter. Furthermore, in order to demonstrate the practical use of our work, we present some experimental results using a commercially available disk drive.

• 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.10 no.6
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• pp.308-316
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• 1985

A new implementation of Block Adaptive filter in frequency domain is presented in this paper. Block digital filtering involves the calculation of a block or finite set of filter out put from a block of input values. A fast convergence algorithm of block adaptive filter is developed using Gordar theory and compared with the performance results of Satio algorithm and BLMS algorithm. Form the result we can be shown that the convergence state of given algorithm is not only faster than BLMS algorithm but also the resulting convergence error is less than the convergence error of Satio algorithm.

• 한국정보디스플레이학회:학술대회논문집
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• 2007.08b
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• pp.1692-1695
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• 2007

We describe the image enhancement method of applying two spatial filters with different characteristics adaptively. An adaptive method is introduced so that sharpness enhancement is performed only in regions where the image exhibits significant dynamics, while noise reduction is achieved in smooth regions. Simulation results show that the proposed method improved the image quality.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.3
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• pp.231-238
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• 2004

This paper presents the high speed noise reduction processing system using the modified discrete fourier transform(MDFT) on the frequency domain. The proposed filter uses the linear prediction coefficients of the adaptive line enhance(ALE) method based on the Sign algorithm The signals with a random noise tracking performance are examined through computer simulations. It is confirmed that the fast adaptive digital filter is realized by the high speed adaptive noise reduction(HANR) algorithm with rapid convergence on the frequency domain(FD).

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.8
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• pp.375-377
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.30-32
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• 2006

An adaptive IIR filter in ANC(Active Noise Control) systems is more effective than an adaptive FIR filter when acoustic feedback exists, in which cause an order of an adaptive FIR filter must be very large if some of poles of the ideal control filter are near the unit circle. But the IIR filters may have stability problems especially when the adaptive algorithm for adaptive filters is not yet converged. In this paper, a stabilized multi-channel recursive LMS (MCRLMS) algorithm for an adaptive multi-channel IIR filter is presented. RLMS algorithms usually diverge before the algorithm is not yet converged. So, in the beginning of the ANC system, the stability of the RLMS algorithms could be Improved by pulling the poles of the IIR filter to the center of the unit circle, and returning the poles to their original positions after the filter converges. Computer simulations and experiments for dipole ducts using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.