• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.3
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• pp.723-728
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• 2010

This study proposed a design of a portable device for measuring heart rate using photoplethysmograph signal to minimize load of a nurse increased from insufficiency of an internal hospital nurse, and algorithm to measure reliable heart rate in PPG signals despite the existence of patient's motion artifacts. The proposed method for measuring heart rate is the method to minimize the motion interference by using the adaptive filter based on harmonic characteristic of PPG signal. To evaluate the performances of the a portable device implemented by the proposed method, we used several motion artifacts including finger and wrist movements; we then compared out results with the performance of the moving average filter. In this results, the proposed method showed a better performance than that of the moving average filter. Therefore, when nurses use the a portable device for measuring heart rate proposed in this study, it will enable to improve nurse work and to measure the reliable heart rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1935-1940
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• 2016

This paper proposes a phase-locked loop (PLL) to minimize the loop filter output voltage fluctuation by using a comparator including RC time constant circuits. The voltage variation of loop filter is inputted to RC time constant circuits which have two RC time constants, large and small. While a small RC time constant circuit quickly conveys the output voltage variation of loop filter, a large RC time constant circuit conveys slowly the output voltage variation of loop filter and its output looks like constant voltage. The output signal of the comparator controls the sub charge pump and reduces the input voltage variation of voltage-controlled oscillator (VCO). Therefore, the proposed PLL generates a phase noise reduced signal. It has been designed with a 1.8V supply voltage, 0.18um multi - metal and multi - poly layer CMOS process and proved by Hspice simulation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.6
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• pp.8-12
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• 2008

A ring resonator based band-pass filter enabling enhanced sideband suppression was proposed and demonstrated. It is composed of a single-ring based band-pass section combined with a band-rejection section based on two different coupled rings. The sideband region of the transfer characteristics of the first band-pass filter section was selectively suppressed by means of the band-rejection section without causing any excessive insertion loss. As a result, the overall transfer characteristics of the proposed filter was unproved so that its sideband suppression was elevated to allow for sharper roll-off and narrower bandwidth. In this work, a conventional single-ring band-pass filter and a proposed filter including the band-rejection section were designed and fabricated. As for the achieved device performance, with the center frequency at about 4 GHz, the sideband suppression ratio was increased as much as 8 dB and the 3-dB bandwidth was reduced from 98 MHz to 48 MHz by about 51% with no remarkable excessive insertion loss as expected. And the quality factor of the device was increased by about 105%.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.11
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• pp.2715-2720
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• 2014

Median filtering is very effective to remove impulse type noises, so it has been widely used in many signal processing applications. However, due to the time complexity of its non-linearity, median filtering is often used using a small filter window size. A lot of work has been done on devising fast median filtering algorithms, but most of them can be efficiently applied to input data with finite integer values like images. Little work has been carried out on fast 2-d median filtering algorithms that can deal with real-valued 2-d data. In this paper, a fast and simple median 2-d filter is presented, and its performance is compared with the Matlab's 2-d median filter and a heap-based 2-d median filter. The proposed algorithm is shown to be much faster than the Matlab's 2-d median filter and consistently faster than the heap-based algorithm that is much more complicated than the proposed one. Also, a more efficient median filtering scheme for 2-d real valued data with a finite range of values is presented that uses higher-bit integer 2-d median filtering with negligible quantization errors.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1A
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• pp.48-57
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• 2006

TWTA is to perform the function that amplifies the input RF signal and outputs it to the antenna. This paper proposes a method that is to improve the cathode ripple or the SHF TWTA for satellite communications. Through the embodiment and experiment of 600W SHF TWTA, this method satisfies the design specifications. Also, RF performance is improved by reducing the noise of auxiliary power sources supplied to the RF part and eliminating the unexpected noise. Therefore, this method is very effective and can be used to develop the similar equipments.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.68-75
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• 1994

In many applications of signal processing, we have to deal with densities which are highly non-Gaussian or which may have Gaussian shape in the middle but have potent deviations in the tails. To fight against these deviations, we consider a finite mixture distribution for the speech excitation. We utilize the EM algorithm for the estimation of speech parameters and their enhancement. Robust Kalman filtering is used in the enhancement process, and a detection/estimation technique is used for parameter estimation. Experimental results show that the proposed algorithm performs better in adverse SNR input conditions.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.11a
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• pp.151-154
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• 2012

본 논문에서는 화상통신의 현실감을 증진시킬 수 있는 화자 간 시선 맞춤 시스템을 제안한다. 제안하는 방법은 Kinect 거리 카메라로부터 입력된 영상에서 화자의 얼굴 영역을 획득하여 화자의 시선이 카메라를 응시하도록 획득한 영역을 변환한 후에 원본 영상과 합성한다. Kinect 거리 카메라에서 획득한 얼굴 영역에는 다양한 형태의 잡음이 많아 미디언 필터와 모폴로지 연산을 통해 얼굴 영역의 잡음을 제거한다. 화자의 위치에 상관 없이 화자가 카메라를 응시하는 영상을 생성하기 위해서 Kinect 가 제공하는 거리 정보를 이용하여 시선 보정 각도와 회전 축을 획득한다. 시선이 보정된 얼굴 영역은 원본 영상에서 존재하지 않는 영역을 포함하고 있기 때문에, 원본 영상의 각 화소를 삼각형 메쉬로 구성한 후 해당 영역을 보간하여 최종적으로 시선이 보정된 영상을 생성한다. 제안하는 방법은 시선 맞춤 영상을 생성하는 데 필수적인 눈과 주변 얼굴 영역만 선택해서 변환하므로 영상의 왜곡이 적고 실시간 처리가 가능하다는 장점이 있다. 또한 카메라와 화자 사이의 거리 정보를 이용해 화자의 위치에 적응적인 시선 맞춤 영상을 생성할 수 있다. 실험을 통해 Intel i5 CPU 를 장착한 PC에서 $320{\times}240$ 크기의 영상을 사용할 경우 초당 약 35 프레임의 보정된 영상을 생성하여 제안하는 방법이 실시간 처리가 가능하다는 것을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.12-20
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• 1999

In this paper, we compared speaker verification performance of the speech data collected in clean environment and in channel environment. For the improvement of the performance of speaker verification gathered in channel, we have studied on the efficient feature parameters in channel environment and on the preprocessing. Speech DB for experiment is consisted of Korean doublet of numbers, considering the text-prompted system. Speech features including LPCC(Linear Predictive Cepstral Coefficient), MFCC(Mel Frequency Cepstral Coefficient), PLP(Perceptually Linear Prediction), LSP(Line Spectrum Pair) are analyzed. Also, the preprocessing of filtering to remove channel noise is studied. To remove or compensate for the channel effect from the extracted features, cepstral weighting, CMS(Cepstral Mean Subtraction), RASTA(RelAtive SpecTrAl) are applied. Also by presenting the speech recognition performance on each features and the processing, we compared speech recognition performance and speaker verification performance. For the evaluation of the applied speech features and processing methods, HTK(HMM Tool Kit) 2.0 is used. Giving different threshold according to male or female speaker, we compare EER(Equal Error Rate) on the clean speech data and channel data. Our simulation results show that, removing low band and high band channel noise by applying band pass filter(150～3800Hz) in preprocessing procedure, and extracting MFCC from the filtered speech, the best speaker verification performance was achieved from the view point of EER measurement.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.84-89
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• 2002

In this paper, we present the design method of noise reduction circuit in telephonic system. The circuit consists of compressor, expander and a filter. The basic idea of a proposed method compresses the audible signal in order to mask the channel noise during transmission and then expand at the reverse rate the transmitted signal to naturally recover the original signal. Of course, there should be no distortion or other degradation of the audio itself in passing through companding(compress/expand) cycle. In the compressing process, the gain of compressor is automatically controlled by the envelope level of input signal in order to increase the effective dynamic range of input signal and to improve the signal to noise ratio. The compressed rate is the root time of a audible signal. The compressed signal should be expanded at the square time of the signal to recover a original signal. Simulation shows the proposed method improves the performance of the noise reduction of a channel noise as well as stability. 

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.2 no.2 s.3
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• pp.1-12
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• 2003

DAB(Digital audio broadcasting) is a next generation radio broadcasting system which provides CD quality audio, various data services and superior reception ability when moving. Also, it can show traffic informations and news literally or graphically. In this paper, we design and fabricate the DAB tuner for ITS service that follows Eureka-147 and ETSI 300 401 specifications. This small-sized tuner can be adopted to mny electronic equipments such as a Hi-Fi audio, DVD player, car audio system etc.. The overall performance of the tuner depends on a phase noise of VCO and the sensitivity of the receiving system is influenced by LNA, image rejection filter and channel selection filter. All our measurement results satisfy the specification for a DAB system with the return loss of 9dB, the noise figure of 6dB for both Band 111 and L-band and the sensitivity of -97dBm.