• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.101-104
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• 2001

In this paper, we propose the Digital Automatic Gain Controller for IEEE 802.11a High-speed Physical Layer in the 5 GHz Band. The input gain is estimated by calculating the energy of the training symbol that is a synchronizing signal. The renewal gain is calculated by comparing the estimated gain with the ideal gain. The renewal gain is converted into the controlled voltage for GCA to reduce or amplify the input signals. We used a piecewise-linear approximation to reduce the hardware size. The gain control is performed seven times to provide more accurate gain control. The proposed automatic gain controller is designed with VHDL and verified by using the Xilinx FPGA.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.241-244
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• 2000

잔향음 세한 환경에서 능동소오나를 운용시 잔향음은 표적탐지를 저해하는 중요한 요인으로 작용한다. 그러한 환경에서 잔향음을 효과적으로 제거하고 표적신호를 보존하기 위한 기법으로 priori estimation error를 사용하는 deterministic LSL(least squares lattice) 알고리즘을 적용하였으며, LSL 필터의 입력신호는 천해에서 고주파 센서를 이용하여 실측한 CW 및 FM 잔향음신호와 모의된 표적신호를 합성한 신호를 이용하였다. 모의 실험 결과 잔향음신호는 상당히 감소되었을 뿐 만 아니라 도플러 변화가 없는 표적신호에 대해서도 LSL필터는 양호한 응답을 보였다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.783-786
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• 2000

본 논문에서는 웨이브렛 변환을 적용한 광대역 음성부호화 알고리즘을 제안하였다. 제안한 음성부호화 알고리즘은 split-band 구조를 가지며, 16 kHz로 sampling된 입력신호를 QMF를 이용해서 동일한 대역폭을 갖는 두 개의 subband 신호로 나누고 이를 8kHz의 sampling율을 갖도록 downsampling 한다. 그리고 저대역 신호는 GSM-EFR 음성부호화 알고리즘을 이용하여 부호화하고, 고대역 신호는 DWT(Discrete Wavelet Transform)을 적용하여 subband로 나누어 부호화하였다. 각 subband에서 양자화 된 파라미터는 IDWT(Inverse DWT)과정을 거쳐서 upsampling되고 합성 QMF를 통과시켜 최종 합성음을 구하였다. 제안한 음성부호화기는 저대역 신호의 GSM-EFR 부호화에 12.2 kbps, 웨이브렛 변환을 이용한 고대역 신호의 부호화에 7.8 kbps로 전체 20 kbps의 전송율을 가지면서 G.722 표준안의 56 kbps에서의 합성음과 비슷한 음질을 나타내었다.

• 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.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1004-1016
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• 2007

In this paper we proposed a Weight-Update-Control Adaptive Noise Canceller which improves speech when environmental noise is stationary and it is hard to acquire a reference signal. Adaptive Noise Canceller(ANC) needs a reference signal, but it is not easy to measure pure noise without voice for reference in factory. Because there are mixed various mechanical noise and workers' voice. Therefore ANC is not suitable to reduce background noise. So we proposed the method that uses an arbitrary constant as an input signal and inputs microphone signal to the reference signal. The noise is eliminated using updated weights in non-speech range. In speech range the weight is fixed and the modified voice is acquired then voice is restored through transversal filter. The proposed method is based on facts that the factory noise is stationary and the noise is not changed in short conversation range. As a result of simulation using MATLAB, we confirmed that the proposed method is effective for reducing factory noise and has high signal to noise ratio(SNR).

• Journal of the Korean Institute of Telematics and Electronics
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• v.9 no.4
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• pp.17-24
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• 1972

A simply constructed and inexpensive amplifier that exhibits unusually low noise is studied. The high-performance differential amplifier combines high input impedence, adjustable gain, low in put noise and low output impedance. The amplifier is particularly useful in applications which call for large amplificaions of very low level signals.

• Proceedings of the Acoustical Society of Korea Conference
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• 1997.06a
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• pp.13-16
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• 1997

본 논문에서는 Bayesian 결정 이론을 이용한 기존의 Radial Basis Function Networks 이되는 출력층에서 선형 조합되는 것과는 다른 형태의 방법을 제안하고자 한다. 제안하고자 하는 방법은 은닉층의 출력값과 가중치와의 곱해진 값이 출력층의 입력으로 들어오는데 이들 입력신호를 경쟁을 통하여 가장 큰 값만을 출력신호 인정하는 방법이다. 이런 경우에 파라미터 갱신을 할 때도 모든 가중치를 다 갱신하는 것이 아니라 출력되는 은닉층에 연결된 가중치만을 갱신하게된다. 이렇게 할 경우 계산량 감소뿐만 아니라 학습시간을 단축할 수 있다는 장점이 있다. 그리고 제안한 방법을 이용할 경우 비선형 분류문제에서도 우수한 성능결과를 확인 할 수 있었으며 기존의 RBFN rhk Wiener Filter와 성능을 비교하였다.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.1755-1758
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• 2003

색 재현 범위(Color Gamut)가 기존의 표준신호(예 sRGB, Rec. 709)대비 상이한 Laser 디스플레이 장치에서 Gamut Matching Algorithm을 이용하여 표준 컬러의 재현을 가능하게 하고, CRT 대비 광범위한 Laser Color Gamut 의 모든 색 영역을 표현 할 수 있다. 이 방법은 일정한 휘도 및 hue 에서 표준입력과 출력장치의 색 재현 범위에 따라 주어진 입력신호의 Chroma를 늘이거나, 줄임으로써 출력장치의 전 색 재현 범위를 사용할 수 있게 된다.

• The Journal of the Acoustical Society of Korea
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• v.30 no.6
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• pp.353-359
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• 2011

This paper proposes a vocal enhancement technique for improving the performance of vocal pitch detection in polyphonic music signal. The proposed vocal enhancement technique predicts an accompaniment signal from the input signal and generates an accompaniment replica signal according to the vocal power. Then, it removes the accompaniment replica signal from the input signal, resulting in a vocal-enhanced signal. The performance of the proposed method was measured by applying the same vocal pitch extraction method to the original and the vocal-enhanced signal, and the vocal pitch detection accuracy was increased by 7.1 % point in average.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.3
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• pp.53-58
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• 2020

This paper is concerned with the performance analysis of MSAGF-MMA with variable step size whose step size varies according to input power signal and decision-directed error signal. The proposed algorithm is made to change according to the input power signal which can reliably increase the convergence speed to the steady state by making the step size less affected by the fluctuation of the input signal in the MMA having the binary flag obtained from the modified Stop-and-Go algorithm. At the same time, the step size can be varied according to the decision-directed error signal so that the residual error can be reduced in the steady state. As a result of computer simulations, it is confirmed that the proposed algorithm has a very good performance in the evaluation of residual ISI and averaged-MSE in steady state as well as in terms of convergence speed to steady state compared to MMA and MSAGF-MMA.