• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4A
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• pp.370-377
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• 2004

In this paper, we present an adaptive modulation technique for orthogonal frequency division multiplexing (OFDM) for broadband wireless communications. Also, using improved channel prediction, we enhance the performance of adaptive OFDM in high mobility environments. Adaptive modulation technique has been shown to achieve reliable high-rate data transmission over frequency-selective fading channel when OFDM is employed. This scheme requires the accurate channel information between two stations for a better performance. In an outdoor high mobility environment, most of adaptive OFDM systems have to be given the channel information transmitted from the receiver. Even if it is possible, there is some delay. Moreover, the channel impulse response between two stations is very rapidly varied. If the channel information is obsolete at the time of transmission, then poor system performance will result. In order to solve this problem, we propose adaptive OFDM with improved channel predictor. The proposed bit allocation algorithm has a lower complexity and the proposed scheme mitigates the effect of channel delay. Robust approach is less sensitive to outdated channel information. Performance results show that the proposed scheme can achieve considerable performance enhancement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1573-1580
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• 2004

In this paper, an adaptation mode controller for adaptive microphone array system is proposed for high-quality speech acquisition in real environments. To ensure proper adaptation of the adaptive array algorithm, the proposed adaptation mode controller uses not only temporal information, but also spatial information. The proposed adaptation mode controller is constructed with two processing stages: an initialization stage and a running stage. In the initialization stage, a sound source localization technique is adopted, and a signal correlation characteristic is used in the running stage. For the adaptive may algorithm, a generalized sidelobe canceller with an adaptive blocking matrix is used. The proposed adaptation mode controller can be used even when the adaptive blocking matrix is not adapted, and is much stable than the power ratio method. The proposed algorithm is evaluated in real environment, and simulation results show 13dB SINR improvement with the speaker sitting 2m distance from the may.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.5
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• pp.961-968
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• 2007

Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.10
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• pp.108-117
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• 2015

If typical contrast enhancement algorithms for natural images are applied to X-ray images, they may cause artifacts such as overshooting or produce unnatural visual quality because they do not consider inherent characteristics of X-ray images. In order to overcome such problems, we propose a locally adaptive block-based contrast enhancement algorithm for X-ray images. After we derive a weighted cumulative distribution function for each block, we apply it to each block for contrast enhancement. Then, we obtain images that are removed from block effect by adopting block-based overlapping. In post-processing, we obtain the final image by emphasizing high frequency components. Experimental results show that the proposed block-based contrast enhancement algorithm provides at maximum 5-times higher visual quality than the exiting algorithm in terms of quantitative contrast metric.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.1
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• pp.102-110
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• 2017

In this paper, we have implemented an real-time adaptive feedback cancellation(AFC) algorithm that can be applied to multi-channel digital hearing aid. Multichannel digital hearing aid typically use the FFT filterbank based Wide Dynamic Range Compression(WDRC) algorithm to compensate for hearing loss. The implemented real-time acoustic feedback cancellation algorithm has one integrated structure using the same FFT filter bank with WDRC, which can be beneficial in terms of computation affecting the hearing aid battery life. In addition, when the AFC fails to operate due to nonlinear input and output, the reduction gain is applied to improve robustness in practical environment. The implemented algorithm can be further improved by adding various signal processing algorithm such as speech enhancement.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9A
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• pp.1013-1021
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• 2004

In the WCDMA mobile multimedia communication system, the adaptive array antenna IS adopted to improve the performance of the system by reducing inter-user interference using antenna beam control. Usually, the interference resulted from the higher data rate users is much more significant to the lower data rate users than the other way around, so the overall performance can be enhanced by reducing the interference from higher data rate users. In order to maximize the efficiency of adaptive antenna operation, an optimal call admission control, especially during handoff, adaptive to the data rates is a critical problem. In this paper, We propose a call admission control algorithm based on the Soft QoS concept for the efficient processing of the handoff of higher data rate calls, and an adaptive handoff control mechanism according to the data rates. The proposed algorithm has been evaluated by computer simulation that it accommodates high data rate users among many lower data rate users much better, and the average call blocking probability for lower rate users becomes much lower than the conventional call admission control algorithm.

• Korean Journal of Radiology
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• v.21 no.6
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• pp.756-763
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• 2020

Objective: To evaluate the effects of attenuation threshold on CT pulmonary vascular volume ratios in children and young adults with congenital heart disease, and to suggest an optimal attenuation threshold. Materials and Methods: CT percentages of right pulmonary vascular volume were compared and correlated with percentages calculated from nuclear medicine right lung perfusion in 52 patients with congenital heart disease. The selected patients had undergone electrocardiography-synchronized cardiothoracic CT and lung perfusion scintigraphy within a 1-year interval, but not interim surgical or transcatheter intervention. The percentages of CT right pulmonary vascular volumes were calculated with fixed (80-600 Hounsfield units [HU]) and adaptive thresholds (average pulmonary artery enhancement [PA_avg] divided by 2.50, 2.00, 1.75, 1.63, 1.50, and 1.25). The optimal threshold exhibited the smallest mean difference, the lowest p-value in statistically significant paired comparisons, and the highest Pearson correlation coefficient. Results: The PA_avg value was 529.5 ± 164.8 HU (range, 250.1-956.6 HU). Results showed that fixed thresholds in the range of 320-400 HU, and adaptive thresholds of PA_avg/1.75-1.50 were optimal for quantifying CT pulmonary vascular volume ratios. The optimal thresholds demonstrated a small mean difference of ≤ 5%, no significant difference (> 0.2 for fixed thresholds, and > 0.5 for adaptive thresholds), and a high correlation coefficient (0.93 for fixed thresholds, and 0.91 for adaptive thresholds). Conclusion: The optimal fixed and adaptive thresholds for quantifying CT pulmonary vascular volume ratios appeared equally useful. However, when considering a wide range of PAavg, application of optimal adaptive thresholds may be more suitable than fixed thresholds in actual clinical practice.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.122-125
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• 2011

In real environment, background noise exists everywhere and degrades the performance of system. To reduce this distortion, a speech enhancement algorithm can be very useful and variety methods have been proposed. In this paper, we propose a postfilter to improve the performance of optimally modified log-spectral amplitude (OM-LSA) estimator. Proposed algorithm uses the formant postfilter to minimize perceptual distortion caused by background noise. We adjust an emphasizing parameter which is varied by spectral flatness and first reflection coefficient. The performance of the proposed algorithm is evaluated by measuring the log-spectral distance (LSD) and the perceptual evaluation of speech quality (PESQ) score. The test results show the improvement of proposed algorithm compared to conventional OM-LSA.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2017.11a
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• pp.101-103
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• 2017

본 논문은 human visual system(HVS)에 따른 주파수 민감도와 공간에서 다양한 특성들을 구현하기 위한 신호처리 방법을 개발하였다. 인간의 눈은 주파수 성분에 따라 민감도가 다르며 초점에서 멀수록 인지 가능한 해상도가 떨어진다. 주파수 민감도를 구현하기 위해서 본 논문은 영상 신호의 에너지 스펙트럼 모양이 contrast sensitivity function(CSF)의 모양이 되도록 하여 영상 신호의 에너지를 증가시켰으며 신호 방향에 적응적인 multiband energy scaling 방법을 개발하였다. 기존의 시스템에서 능률만을 향상시키는 기존의 분석 모델과 비교하면 개발한 방법은 HVS에 좀 더 적절하고 선호되게 영상 신호를 처리할 수 있다.