• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.90-94
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• 2008

It is well-blown that the convergence rate gets worse when an input signal to an adaptive filter is correlated. In this paper we propose a new adaptive filtering algorithm that makes the convergence rate much improved even for highly correlated input signals. By introducing an orthogonal constraint between successive input signal vectors we overcome the slow convergence problem of the LMS algorithm with the correlated input signal. Simulation results show that the proposed algerian yields fast convergence speed and excellent tracking capability under both time-invariant and time-varying environments, while keeping both computation and implementation simple.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.922-923
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• 2013

최근 고령화가 증가하면서 헬스케어기술도 함께 발달해 가고 있다. 또한 스마트폰이 발전해 가면서 헬스케어기술과 함께 융합한 기술이 많이 연구되어 가고 있다. 기존의 대형화 되었던 기술들이 스마트폰을 통해 제작되면서 생체센서만 부착되면 U-헬스케어 기술이 구현될 수 있는 세상이 실현된 것이다. 본 논문에서는 스마트폰에서 공용으로 부착되어 있는 오디오(AUX) 단자를 사용해 생체신호를 입력받았다. 일반적으로 스마트폰의 오디오단자는 음성의 입출력을 할 수 있도록 설계되었으나 오디오의 마이크 단자를 활용할 경우 생체신호를 입력 받을 수 있다. 본 연구에서는 PPG회로를 구현하고 오디오 단자를 통해 입력받은 생체신호를 애플리케이션을 통해 모니터링 하는 프로그램을 제작하였다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.5
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• pp.435-441
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• 2017

Detecting an inter-harmonic accurately is not easy work, because it has small magnitude, and its frequency which can be observed is not an integer multiple of fundamental frequency. In this paper, a new method using filter bank system and adaptive predictor is proposed. Filter bank system decomposes input signal to sub bands. In adaptive predictor, inter-harmonic is detected with decomposed sub band signal as input, and error signal as output. In this scheme, input-output characteristic of adaptive predictor is notch filter, as predicted harmonic is canceled in error signal, so detecting an inter-harmonic can be possible. Magnitude and frequency of detected inter-harmonic is estimated by recursive algorithm. The performances of proposed method are evaluated to sinusoidal signal model synthesized with harmonics and inter-harmonics. And validity of the method is proved as comparing the inter-harmonic detection results to MUSIC and ESPRIT.

• Journal of the Institute of Convergence Signal Processing
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• v.24 no.1
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• pp.51-56
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• 2023

This study investigates the effect of sEMG sampling frequency on CNN learning performance at Korean finger number recognition application. Since the bigger sampling frequency of sEMG signals generates bigger size of input data and takes longer CNN's learning time. It makes making real-time system implementation more difficult and more costly. Thus, there might be appropriate sampling frequency when collecting sEMG signals. To this end, this work choose five different sampling frequencies which are 1,024Hz, 512Hz, 256Hz, 128Hz and 64Hz and investigates CNN learning performance with sEMG data taken at each sampling frequency. The comparative study shows that all CNN recognized Korean finger number one to five at the accuracy of 100% and CNN with sEMG signals collected at 256Hz sampling frequency takes the shortest learning time to reach the epoch at which korean finger number gestures are recognized at the accuracy of 100%.

• Journal of Broadcast Engineering
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• v.22 no.5
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• pp.665-670
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• 2017

In this paper, we propose an implementation technology for an active attenuator with the maximum attenuation of -110dBm in the frequency band of 50MHz to 2,15GHz. It provides USB interface to connect to PC providing GUI that sets signal frequency and attenuation step of minimum 1dB. As it attenuates the input signal level down to -110dBm, circuit and equipment design technologies are applied to control both internal and external electro-magnetic noises.. Through the performance test, it is assured that it attenuates input signal level down to -110dBm for the input signal levels of -10 to -30dBm.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.149-156
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• 2012

A digital input class-D audio amplifier is presented for digital hearing aid. The class-D audio amplifier is composed of digital and analog circuits. The analog circuit converts a digital input to a analog audio signal (DAC) with noise suppression in the audio band. An interpolated digital delta-sigma modulator is used to convert data types between digital signal processor (DSP) and digital-to-analog converter (DAC). An 16-bit, 25-kbps pulse code modulated (PCM) input is interpolated to 16-bit, 50-kbps by a digital filter. The output signal of interpolation filter is noise-shaped by a third-order digital sigma-delta modulator (SDM). As a result, 1.5-bit, 3.2-Mbps signal is applied to simple digital to analog converter.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.65-73
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• 2008

The characterization process for the accurate color reproduction in mobile phone with camera and LCD is popular. The camera and LCD characterization, gamut mapping process is necessary to map the camera's input color stimulus, CIEXYZ value, into the LCD's output color stimulus. Each characterization is the process estimating the relation between input and output signals. In case of LCD, because of output device, the output color stimulus for the arbitrary input signal can be measured by spectro-radiometer However, in the camera, as the input device, the characterization is an inaccurate and needs the manual works in the process obtaining the output signal because the input signal can not be generated. Moreover, after gamut mapping process, the noise is increased because the optimized gamma tone curve of camera for the noise is distorted by the characterization. Thus, this paper proposed the system of obtaining the output signal of camera and the method of gamma correction for the noise. The camera's output signal is obtained by RGB values of patches from captured the color chart image. However, besides the illumination, the error for the location of the chart in the viewfinder is generated when many camera modules are captured the chart. The method of correcting the position to correct the error from manual works. The position of camera is estimated by captured image. This process and moving of camera is accomplished repeatedly, and the optimized position can be obtained. Moreover, the lightness curve of camera output is corrected partly to reduce the noise from the characterization process.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.193-196
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• 1999

본 논문은 음향 임펄스를 모델링하는데 코사인 변조된 필터 뱅크와 Decimation을 이용하여 수렴 속도를 개선하는 방법을 제안하고 이를 잡음제거에 응용하였다. 제안된 구조는 입력신호를 필터뱅크를 이용하여 각 서브밴드로 분할한 후 필터 입력신호의 고유벡터의 최대값과 최소값의 비를 줄이고 필터의 탭수를 줄이기 위해서 decimation을 행한다. 그리고 서브밴드대역의 샘플링 주파수를 낮추어 신호 스펙트럼을 확장시켜 이를 적응필터에 입력하여 수렴속도를 향상시켰다. 실험 결과, Colored잡음의 경우 LMS 알고리즘보다 제안된 방법이 MSE(Mean Square Error)는 좋지는 않았다. 실제 음향시스템의 모델링에는 거의 같은 MSE을 갖으며, 수렴 속도에는 모두 빠른 성능을 보였으며, 이를 음질향상에 적용하여 향상된 음질을 얻을 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.96-103
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• 2003

This paper describes a method for sinusoidal modeling of audio signals using perceptually weighted matching pursuit. Matching pursuits extracts iteratively the greatest energy signals from the input signals until the residual between the original and the reconstructed signal is zero. In this paper, perceptual matching pursuits using psychoacoustic model to matching pursuit extracts greatest perceived energy iteratively. To evaluate the performance of the perceptual matching pursuits it is compared with the sinusoidal matching pursuits which is not included perceptual weighting. For various audio signals the result of simulation shows that the perceptual matching pursuit is superior to the sinusoidal matching pursuits, especially for a high change rate in time domain it can synthesized original signal.

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.4
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• pp.293-296
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• 2007

In this paper, we propose the systolic array architecture for the vector median filter. In the color image processing, the vector signal (i.e. the color) consists of three elements, red, green and blue. The vector median filter is very effective to utilize the correlation among red, green and blue elements. The computational complexity of the proposed architecture for computing the vector median of N vector signals is (N+2) clock periods compared to the (3N+1) clock periods in the previous method. In addition to, the input vector signals can be loaded in serial in the proposed architecture. In the previous method, N input vector signals should be loaded to the vector median filter in parallel at the first clock. The proposed architecture is implemented with FPGA.