• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.3066-3068
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• 2005

Noise can make surrounding environments inferior and deteriorates operation efficiency, and it can bring aural damage as well as give a person psychological stress. Therefore, necessity of study about noise control is increased for better labor conditions and agreeable habitat. In this paper, implementation of active mufflers using a stable IIR adaptive filters is presented. The IIR filter structure is more effective when acoustic feedback exists, but the adaptive IIR filters could be unstable when the filter algorithm is not yet converged. A stabilizing process for adaptive IIR filter is introduced in this paper. Experiments using a TMS320C32 digital signal processor have performed to show the effectiveness of a proposed algorithm.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.838-841
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• 2009

This paper proposes a novel lossless coding scheme for Bayer color filter array (CFA) images which are generally used as internal data of color digital cameras having a single image sensor. The scheme employs a block-adaptive prediction method to exploit spatial and spectral correlations in local areas containing different color signals. In order to allow adaptive prediction suitable for the respective color signals, four kinds of linear predictors which correspond to 2 ${\times}$ 2 samples of Bayer CFA are simultaneously switched block-by-block. Experimental results show that the proposed scheme outperforms other state-of-the-art lossless coding schemes in terms of coding efficiency for Bayer CFA images.

• Proceedings of the Korean Society of Marine Engineers Conference
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• 2006.06a
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• pp.109-110
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• 2006

The performance of a digital baseband signal processing and data transmission rate depends on the modulation technique. In this paper, We implemented DSP communication system for Underwater acoustic communication using by adaptive BPSK modem technique. In order to implement adaptive modem, we suggested SNR detection block. SNR detection block has the reference SNR value that selects between window filter path and matched filter path. In this paper, suggested system is based on software interface and all Hardware(PLL, modem filter, equalizer etc) is implemented by software, exclusive of DSP, A/D, D/A converter, SDRAM and Flash memory.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.939-946
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• 2008

본 논문은 시청자의 위치에 최적화된 입체음향을 제공하는 TV 개발에 관한 것으로 2 개의 TV 스피커만으로 5.1 채널 스피커가 주는 입체음향 효과를 제공해준다. 기존의 Speaker Virtualizer 기술은 시청자가 특정 위치(Sweet Spot)를 벗어나면 입체음향 성능이 현저히 저하된다. 반면, 본 논문에서 제안하는 Adaptive Virtualizer 기술은 초음파가 장착된 리모콘을 사용하여 시청자의 위치를 인식하고 인식된 시청자의 위치 정보를 활용하여 청취위치에 해당하는 HRTF로부터 설계된 Filter를 Update 하고 두 스피커의 출력레벨 및 시간지연 값을 보정함으로써 최적의 입체음향을 재현한다. 본 논문에서는 실시간 구현을 위해 Speaker Virtualizer의 계산량을 최소화하는 기술을 제안하고 다양한 청취 위치에 해당하는 Filter를 설계하고 설계된 Filter를 효율적으로 Update 하는 Adaptive Virtualizer 기술을 제안한다. 또한, 초음파를 이용한 시청자 위치 인식 기술 및 전체 시스템 통합 기술을 제시한다.

• Journal of Biomedical Engineering Research
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• v.10 no.2
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• pp.151-156
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• 1989

Lung sound analyer which can provide an objective diagnosis of patients with pulmonary and bronchial disorders is designed. For the purpose of power spectrum analysis, adaptive digital filtering technique and TM - S320C25 DSP chip is used. As a results, adaptive lattice Wiener filter could eliminate heart sounds with a few of 10th order and on the distribution of power spectrum each patterns has shown in normal vescicular breathy from 100 Hz to 200 Hz, in crackle sound from 100 Hz to 400 Hz, in wheeze sound from 150 Hz to 600 Hz.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.9B
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• pp.1226-1235
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• 2001

The convergence analysis of the improved multiplication free adaptive digital filter (IMADF) with a fractionally-spaced equalizer (FSE) to remove the intersymbol interference (ISI) in fast data transmission is presented. The IMADF structure use the one-step predicted filter in the multiplication-free adaptive digital filter (MADF) structure using the DPCM and Sign algorithm. In the experimental results, the IMADF algorithm has reduced the computational complexity by use of only the addition operation without a multiplier. Also, under the condition of identical stationary-state error, it could obtain the stabled convergence characteristic that the IMADF algorithm is almost same as the sign algorithm, but is better than the MADF algorithm. Here, this algorithm has effective characteristics when the correlation of the input signal is highly.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.53 no.9
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• pp.657-662
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• 2004

The low-power design of the digital hearing aid is indispensable to achieve the compact portable device with long battery duration. In this paper, new low-power adaptive filter structure is proposed based on distributed arithmetic(DA). By modifying the DA technique, the proposed decimation filter structure can significantly reduce the power consumption and implementation area. Through Verilog-HDL coding, cell occupation of the proposed structure is reduced to 33.49% in comparison with that of the conventional multiplier structure. Since Verilog-HDL simulation processing time of the two structures are same, it is assumed that the power consumption or implementation area is proportional to the cell occupation in the simulation.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2002.11a
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• pp.44-48
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• 2002

This paper describes an adaptive signal processing of digital receiver with DDC(Digital Down Convertor), DDC is implemented by using NCO(Numerically Controlled Oscillator), digital low pass filter. for the passband sampling, we present the results of digital receiver simulation with DDC. We confirm that the low IP signal is converted to zero IF by DDC. DOA(Direction Of Arrival) estimation technique using MUSIC(Multiple SIgnal Classification) algorithm with high resolution is presented. We Cow that an accurate resolution of DOA depends on the input sampling number.

• 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 Electromagnetic Engineering and Science
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• v.15 no.4
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• pp.360-367
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• 2004

In this paper, we have implemented the digital signal processing system for intruder detection using speckle pattern variation in multi-me optical fiber with hypersensitive and high fidelity. The performance of the implemented system was evaluated by experiments. In order to improve the system performances we applied the adaptive digital filter. In experimental results we could see 96 % intruder detection and 90 % man/car discrimination probability.