• Journal of Sensor Science and Technology
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• v.27 no.3
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• pp.192-198
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• 2018

Pulse oximetry is a non-invasive method for monitoring how much oxygenated hemoglobin is present in the blood. The principle of pulse oximetry is based on the red infrared light adsorption characteristics of oxygenated and deoxygenated hemoglobin. Even through the convenience of a pulse oximeter, its weak signal-to-noise ratio against motion artifacts and low perfusion makes it difficult to be accepted by execs devices. Several researchers have suggested the use of an adaptive noise cancellation (ANC) algorithm. They have demonstrated that ANC is feasible for reducing the effects of motion artifacts. Masimo Corporation developed a discrete saturation transformation (DST) algorithm that uses a reference signal and ANC. In commercial devices, it is very hard to escape it because Masimo's patents are very powerful and a better method is yet to be developed. This study proposes a new method that can measure noise saturation as well as accurate oxygen saturation from signals with high motion artifacts without using ANC and DST. The proposed algorithm can extract a normal signal without noise from a signal with motion artifacts. The reference signal from a pulse oximeter simulator was used for the evaluation of our proposed algorithm and achieved good results.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.414-417
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• 2008

This paper presents the design and analysis of 2.45GHz Low-IF Mixer using CMOS 0.18um. The Mixer is implemented by using the Gilbert-type configuration, current bleeding technique, and the resonating technique for the tail capacitance. And the design of this Double Balance Mixer is based on its lineaity since it is important in the interference cancellation system. The low flicker noise mixer is implemented by incorporating a double balanced Gilber-type configuration, the RF leakage-less current bleeding technique, and Cp resonating technique. The proposed mixer has a simulated conversion gain of 16dB a simulated IIP3 of -3.3dBm and P1dB is -19dBm. A simulated noise figure of 6.9dB at l0MHz and a flicker corner frequency of 510kHz while consuming only 10.65mW od DC power. The layout of Mixer for one-chip design in a 0.18-um TSMC process has 0.474mm$\times$0.39 mm size.

• International Journal of Highway Engineering
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• v.15 no.4
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• pp.95-105
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• 2013

PURPOSES: The purpose of this study is to research the influence of road traffic noise by road slope through the analysis of the field road traffic noise and determine consideration of road slope in the case of appling active noise cancellation. METHODS: This study measures vehicle's noise by the NCPX method at the three field sections such as uphill, downhill, and flatland. Total sound pressure and sound pressure level by the 1/3 octave band frequency are calculated through the raw field data. Total sound pressure level is compared by ANOVA test and T test statistically. The results obtained are compared in accordance with the road slope and the progress of the uphill section. RESULTS : The noise characteristic of early, medium, and last parts of uphill was found to be consistent when the vehicle was travelling uphill section. The result of statistical test, it was shown that total sound pressures are not different each other. According to the comparison by the geometry, sound pressure of the uphill section was higher than those of the flatland and downhill section in high frequency band. By the result of statistical test, total sound pressure are different according to geometry in the case of high vehicle speed. In the comparison result by road slope, each sound pressure level was found to be consistent in total frequency. However, total sound pressure proportionally increased according to road slope. CONCLUSIONS: It is found that the effect of road slope on noise generation was little in this experimental sites.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.12
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• pp.1551-1557
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• 2019

In modern society, a variety of video media have been widely spread in line with the fourth industrial revolution and the development of IoT technology; in accordance with this trend, numerous researches have been performed to remove noise generated in image and data communications. However, the conventional Additive White Gaussian Noise (AWGN) cancellation techniques are likely to induce a blurring phenomenon in the noise removal process, thus impairing the information of the image. In this study, we propose an algorithm for minimizing the loss of image information in the removal process of AWGN. The proposed algorithm can apply weights according to the characteristics of noise by predicting AWGN in the image, where the output is calculated based on adding and subtracting the outputs of the high pass filter and the low pass filter. Compared to the existing method, the noise reduction using the proposed algorithm exhibited less blurring issues and better noise reduction properties in the AWGN removal process.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.12 no.6
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• pp.636-642
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• 2019

In this paper, we analyzed BER (Bit Error Rate) communication performance according to the detection order of MMSE (Minimum Mean Square Error) based soft decision interference cancellation. As the detection order method, antenna index order method, absolute value magnitude order method of channel elements, absolute value sum order method of channel elements, and SNR (Signal Noise Ratio) order method are proposed. BER performance for the scheme was measured and analyzed. As a simulation environment, 16-QAM (Quadrature Amplitude Modulation) modulation is used in an uncoded environment of an M×M multiple-input multiple-output system, and an independent Rayleigh attenuation channel is considered. The simulation results show that the performance gain is about 1.5dB when the SNR-based detection order method is M=4, and the performance gain is about 3.5dB when M=8 and about 3.5dB when M=16. The more BER performance was confirmed, the more the detection order method of the received signal prevented the interference and error spreading occurring in the detection process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7C
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• pp.413-420
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• 2011

This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.433-438
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• 1999

This paper describes Active Noise Cancellation/Control(ANC) method that removes the information of the unnecessary noise and doesn't remove the informations of the necessary noise(warning sound, operating sound etc.) for the induced noise of the mechanical system. In this paper, the noise source Is axial fan, and the Feedback Active Noise control method that can effectively control BPF generated from the axial fan is used, and the Filtered-X LMS algorithm for adaptive algorithms is used. The experiments are executed for two case(propagating noise in the duct, emission noise for exterior free field). The part to be removed is BPF noise, and the band-pass filter not to effect to the other frequencies is used. Also, to investigate the effect of the noise reduction for human, we are compared with the results that are controlled for using Loudness before and after. As a results, we are certified that the BPF is decreased only and frequencies outside of BPF are not affected, and we acquire the reduction effects of 6.7 dB Loudness Level, then the frequency to be removed is controlled. Therefore, we can be certified that sound pressure as well as loudness can be effectively decreased for human sound quality

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.17-25
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• 2001

Traditional adaptive noise cancelling methods rely their performance on various interfering parameters, such as convergence speed, tracking ability, numerical stability, relative frequency characteristics between target and reverberation signals, and activity of the target. In this paper, an adaptive noise cancelling method is suggested, which Provides a successful tradeoff mon these factors. It is designed to work on the transform domain, adopts the Gram-Schmidt orthogonalization process, and is implemented by the escalator algorithm. The transform domain approach supports a tradeoff between the convergence speed and numerical cost. The proposed method is verified by applying a real-data collected in the shallow waters off the east coasts of korea. It is shown that it has a good reverberation-rejection capability even for the target signal with adjacent frequency components to those of the reverberation, and its performance is invariant for the activity of the target.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.1
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• pp.129-139
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• 1989

During a signal is being transmitted, an interference signal can be introduced through an unknown channel. In these cases, an adaptive system, so called adaptive noise canceller, can restore the original signal from the corrupted signal by first identifying the unknown interference channel on the minimum mean square error criteron, and then by cancelling the interference signal using the identified interference channel. Whereas this method is quite effective when the a priori knowledges about the characteristics of the interference signal and of the intrference channel are unknown or time-varyng, but has a drawback that the presence of the original signal has a severe effect on the optimum value of the interference channel to be identified on the miniumum mean square eror criterion In this paper an adaptive noise canceller and its algorithms are introduced that can restore the original signal more accurately especially when the correlatedness between the original signal and the interference signal is small.

• ETRI Journal
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• v.42 no.1
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• pp.17-25
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• 2020

Multiple-input multiple-output systems can achieve a full sum rate (SR) via full duplex (FD). However, its performance is degraded by self-interference (SI) that occurs between the transmitter and receiver at the same node and thus is constrained by error floors. Conversely, half duplex (HD) can avoid the SI albeit at lower spectral efficiency, and the slope of its error curve is determined by the diversity order. In this study, a link selection scheme based on switching between FD and HD is examined as a simple method to improve the bit error rate (BER) performance of FD systems. In the proposed link selection algorithm, either FD or HD is selected based on the received minimum distance and signal-to-interference plus noise ratio. Simulation results indicate that the proposed hybrid FD/HD switching system offers significant BER performance improvement when compared with that of the conventional FD and FD based on only the received minimum distance under the same fixed SR. Under relatively sufficient SI cancellation, it is demonstrated to outperform the HD with a diversity advantage in low and medium signal-to-noise ratio region.