• Journal of Biomedical Engineering Research
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• v.22 no.5
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• pp.431-438
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• 2001

Oxygen saturation of blood is defined as ratio of total hemoglobins density to oxyhemoglobins density And the accuracy of pulse oxymeter that measures the oxygen saturation of blood by a noninvasive method is influenced by a measuring environment, breathing and motion of patient. Especially when patient moved his arms and fingers, it is difficult to eliminate motion artifact because the motion artifact signal has features that are overlap or closed at normal signal in frequency domain. We propose the filtering method that construct the filter banks and a matched falter to improve the Problem. When experimented by the proposed method, the ratio regulation of the proposed methods has 4.1% below than an adaptive filter (39.7%) and a moving average filter (11.2%). So. the Proposed method will be able to get a stable ratio of SpO2.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.446-453
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• 2004

In this paper, a new adaptive perceptual filter using variable threshold to enhance audio signals degraded by additively nonstationary noise is proposed. The adaptive perceptual filter updates variable threshold each time according to the power of signal and the effect of noise variation. So the noisy audio signal is enhanced by the method which controls a residual noise effectively. The proposed algorithm uses the perceptual filter which transforms a time domain signal into frequency domain and calculates an intensity energy and an excitation energy in bark domain. In this method. the stage updated the response of filter is decided by threshold. The proposed algorithm using vairable threshold effectively controls a residual noise using the energy difference of audio signals degraded by the additive nonstationary noise. The proposed method is tested with the noisy audio signals degraded by nonstationary noise at various signal -to-noise ratios (SNR). We carry out NMR and MOS test when the input SNR is 15dB. 20dB. 25dB and 30dB. An approximate improvement of 17.4dB. 15.3dB, 12.8dB. 9.8dB in NMR and enhancement of 2.9, 2.5, 2.3, 1.7 in MOS test is achieved with the input signals. respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.6
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• pp.1224-1228
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• 2003

The use of digital imaging technique and digital contents based on internet has grown rapidly for last several years, and the needs of digital image protection become more important. For the purpose of copyright protection on digital image, the verification of authentication techniques like content authentication, ownership authentication, illegal copy and etc are needs. Digital watermarking, the invisible encryption technique to insert digital watermarking into image, the sophisticated perceptual information should be used for providing transparency and robustness of images on watermarking process. In this paper, we implement the algorithm for preventing forged attack. ownership protection and authentication by transforming the wavelet algorithms in frequency domain in terms of human visual system.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.11a
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• pp.23-26
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• 2003

디지털로 신호를 표현하는 방법은 기존 아날로그 표현하는 방법에 비해 많은 장점을 가지고 있다. 하지만 디지털로 된 데이터는 언제 어디서든 대단위 복제가 가능하다. 즉, 저작권 침해, 불법 복제 및 배포, 손쉽게 위조할 수 있다는 점이 그것이다. 디지털 영상 정보의 보호를 위해 디지털 영상의 불법적인 내용 조작을 막고, 영상의 소유권을 보장할 수 있는 방법으로 디지털 워터마크(Digital Watermark)가 있다. 디지털 워터마크는 공개키 알고리즘이나 방화벽 등으로 해독된 영상에 대하여 부가적인 보호를 제공한다. 본 논문에서는 디지털 영상 데이터의 정보 보호를 위해 주파수 영역에서의 웨이브릿 변환(Wavelet Transform)을 이용한 이미지 적용 디지털 워터마킹(Image-Adaptive Digital Watermarking) 방법을 제안한다. 이미지 적용 웨이브릿(Image-Adaptive Wavelet)은 영상을 주파수적으로 분해하면서 각 대역들의 공간 영역에서의 정보를 함께 지니고 JND(Just noticeable difference)을 포함한다. 이미지 적응 웨이브릿의 이러한 특성을 이용하여 다해상도 분해하고, 손실 압축(Loss Compression)이나 필터링(Filtering), 잡음(Noise)등에 크게 영향받는 저주파 성분과 인간의 시각적으로 큰 의미를 갖는 고주파 성분의 특성을 이용하여 워터마크를 삽입한다.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.911-916
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• 2010

This paper proposes a new algorithm to cancel the acoustic feedback and noise signals in digital hearing aids. The proposed algorithm combines the feedback canceller to remove acoustic feedback signals and the noise canceller to reduce background noises. The feedback canceller is implemented by normal adaptive FIR filter, and the noise canceller is implemented by using the Wiener solution in frequency domain. This noise canceller has the transfer function presented by the power spectral density of signals. To verify the performances of the proposed algorithm, the simulations were carried out for the system. As the results of simulations, it was proved that we can advance 10.85dB output SNR on the average for the forward path gain of 0dB, and 11.04dB output SNR on the average for the forward path gain of 6dB, in the case of using the proposed algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.18 no.6
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• pp.1103-1108
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• 2023

In this paper, we propose a new algorithm for attenuating the background noises in acoustic signal. This algorithm improves the noise attenuation performance by using the FNN(: Full-connected Neural Network) deep learning algorithm instead of the existing adaptive filter after wavelet transform. After wavelet transforming the input signal for each short-time period, noise is removed from a single input audio signal containing noise by using a 1024-1024-512-neuron FNN deep learning model. This transforms the time-domain voice signal into the time-frequency domain so that the noise characteristics are well expressed, and effectively predicts voice in a noisy environment through supervised learning using the conversion parameter of the pure voice signal for the conversion parameter. In order to verify the performance of the noise reduction system proposed in this study, a simulation program using Tensorflow and Keras libraries was written and a simulation was performed. As a result of the experiment, the proposed deep learning algorithm improved Mean Square Error (MSE) by 30% compared to the case of using the existing adaptive filter and by 20% compared to the case of using the STFT(: Short-Time Fourier Transform) transform effect was obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.8
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• pp.1774-1781
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• 1997

In this paper, we propose a full digital frequency and phase locked loop for CATV and HDTV receivers adopting VSB modulation. The CATV and HDTV receivers proposed by the Grand-Alliance in USA are ultilizing analog signal processing technology for carrier recovery. By the way, it is not a good architecture for the development of single chip ASIC operating in digital domain. To solve this problem while improving the performance, we first down convert the received r.f. signal to a near baseband signal for a low-rate AD converter and then we use digital signal processing techniques. The proposed system has the frequency pull-in range of -200 KHz +2.50 KHz. Moreover, it has the ability of adaptive loop bandwidth control according to the amount of frequency offset to improve the acquisition time while reducing the phase noise.

• Journal of the Korean Institute of Intelligent Systems
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• v.17 no.7
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• pp.945-951
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• 2007

Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.6
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• pp.78-87
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• 2008

Digital image find own level core media in multimedia as image restoration technology fields, which remove degradation factor for image enhancement, have been growing. Linear space-invariant image restoration algorithm often introduce ringing artifacts near sharp intensity transition areas. This paper presents a new adaptive post-filtering algorithm for reducing ringing artifact. The proposed method extracts an edge map of the image using wavelet transform Based on the edge information, ringing artifacts are detected, and removed by an adaptive bilateral filter. Experimental results show that the proposed algorithm can efficiently remove ringing artifacts with edge preservation.