• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1571-1576
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• 2013

This paper proposes an improvement algorithm of signal-to-noise ratios (SNRs) for speech signals under noisy environments. The proposed algorithm first estimates the SNRs in a low SNR, mid SNR and high SNR areas, in order to improve the SNRs in the speech signal from background noise, such as white noise and car noise. Thereafter, this algorithm subtracts the noise signal from the noisy speech signal at each bands using a spectrum sharpening method. In the experiment, good signal-to-noise ratios (SNR) are obtained for white noise and car noise compared with a conventional spectral subtraction method. From the experiment results, the maximal improvement in the output SNR results was approximately 4.2 dB and 3.7 dB better for white noise and car noise compared with the results of the spectral subtraction method, in the background noisy environment, respectively.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.291-294
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• 2004

본 논문에서는 잡음의 변화(variance)가 없는 정적인 통계적 특성(Stationary Static Characteristic)을 갖는 환경에서 잡음 추정을 통해 지각 필터의 성능을 개선하는 알고리즘을 제안한다. 제안된 잡음 추정 알고리즘은 입력되는 잡음에 열화 된 신호의 묵음 구간에서 추정된 잡음을 이용하여 입력되는 잡음의 SNR을 추정 후, 대역 별로 smoothing 상수 값으로 잡음 에너지를 제어하여 첨가된 잡음을 추정함으로써 초기 추정 잡음 보다 가까운 추정 잡음을 얻을 수 있게 된다. 이는 신호를 열화 시킨 잡음을 보다 정확한 추정을 제공함으로써, 지각 필터의 응답을 개선할 수 있고 더불어 잡음에 의해 열화 된 신호의 음질을 개선할 수 있다. 또한 저 대역에 영향을 미치는 잡음인 경우 다른 방법들과는 달리 음질의 개선이 뚜렷하다. 기존의 방식과 비교를 위해 다양한 신호 대 잡음 비(signal-to-noise ratio, SNR)에서 열화 된 오디오 신호를 입력으로 사용하였다. 입력 SNR이 5dB, 10dB, 15dB와 20dB의 각각의 경우에 대하여 SSNR(Segmental SNR)과 잡음 대 마스킹 비(Noise-to-mask ratio, NMR), 음질 테스트를 수행한 결과, 청감 테스트(Mean Opinion Score, MOS Test) 결과의 향상과 음질개선의 개선을 확인할 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.3B
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• pp.259-265
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• 2001

도래각 추정을 위한 대표적인 알고리즘으로 고유치 분해방식을 이용한 ESPRIT 알고리즘이 있다. 이 알고리즘은 다중 신호의 도래각을 추정할 수 있을 뿐만 아니라, 낮은 SNR에서 뛰어난 분해능을 가지고 있다. 이러한 뛰어난 추정 능력에도 불구하고, ESPRIT 알고리즘은 서로 주파수와 위상이 같은 coherent 한 신호를 분해하지 못하는 커다란 결점을 가지고 있다. 본 논문에서는 이러한 보완한 개선된 ESPRIT (Modified ESPRIT )알고리즘을 제안한다. 개선된 ESPRIT 알고리즘 기존의 ESPRIT 에 spatial smoothing 방법을 도입하여 구성되어진다. 모의실험결과 개선된 ESPRIT 알고리즘은 coherent 하게 입사하는 신호의 도래각을 분해하는 능력에 있어서 매우 우수한 성능을 갖고 있음을 확인할 수 있다.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2005.04a
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• pp.437-440
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• 2005

본 논문에서는 고속의 적웅 지각 필터에서 잡음 과추정으로 인해서 발생하는 불필요한 반복 계산 및 결과 신호의 SNR 성능 저하를 개선시키는 방법을 제안한다. 적응 지각 필터를 고속연산이 가능하도록 개선하는 과정에서 시간적인 측면에서는 많은 성능의 개선이 있었지만 음질 개선 과정에서 과추정된 잡음의 적용에 의한 성능 저하가 발생하였다. 제안하는 시스템에서는 적웅 지각 필터의 임계값을 조정하고, 임계값이외에 발생하는 잡음 과추정에 대해서 실험적으로 필터 반복 연산량 제한을 통해 향상된 결과를 얻었다. 이 시스템에서 필터 반복 연산량은 입력 구간의 신호에 따라 적응적으로 제한된다. 제안된 알고리즘의 개선 확인을 위해서 감소된 반복 연산량과 SNR 개선량을 측정하여 기존의 방법과 비교하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1C
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• pp.81-89
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• 2003

For improving the performance of noise suppression in tactical communication environments, an enhanced MELP vocoder is suggested, in which an acoustic noise suppressor is integrated into the front end of the MELP algorithm, and an FEC code into the channel side of the MELP algorithm. The acoustic noise suppressor is the modified IS-127 EVRC noise suppressor which is adapted for the MELP vocoder. As for FEC, the turbo code, which consists of rate-113 encoding and BCJR-MAP decoding algorithm, is utilized. In acoustic noise environments, the lower the SNR becomes, the more the effects of noise suppression is increased. Moreover, The suggested system has greater noise suppression effects in stationary noise than in non-stationary noise, and shows its superiority by 0.24 in MOS test to the original MELP vocoder. When the interleave size is one MELP frame, BER 10-6 is accomplished at channel bit SNR 4.2 ㏈. The iteration of decoding at 3 times is suboptimal in its complexity vs. performance. Synthetic quality is realized as more than MOS 2.5 at channel bit SNR 2 ㏈ in subjective voice quality test, when the interleave size is one MELP frame and the iteration of decoding is more than 3 times.

• Journal of the Korean Society of Radiology
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• v.14 no.6
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• pp.763-772
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• 2020

During X-ray examinations in the DR system, the scattered X-rays physically generated by the patient cause image blurring in poor quality. Although X-rays to increase the contrast of images, this increases the patient's exposure dose and is likely to result in grid induced artifacts. Therefore, the purpose of this study is obtain images similar to those of real-grid with non-grid level conditions using a VGR (Virtual Grid Role) algorithm that serves as a virtual grid. Comparing MTF, SNR and CNR of non-grid and VGR algorithm images obtained with 70% exposure conditions of real-grid images showed that the MTF0.5 differed from 0.265 to 0.350 and the MTF0.1 from 0.412 to 0.467 and the SNR, CNR were also different. In addition, comparing MTF, SNR and CNR of VGR algorithm and real-grid images showed that the MTF0.5 differed from 0.350 to 0.367 and the MTF0.1 from 0.467 to 0.483 and the SNR, CNR by little.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.465-468
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• 2005

MMA(Multi-Modulus Algorithm) is inferior at a initial equalization in high ISI(intersymbol interference), because it is the inaccurate decision. To improve this probel SMMA(Sliced Multi-Modulus Algorithm) is based on using the MCMA(Modified Constant Modulus Algorithm). SMMA is a improved capability than MMA in high SNR but is inaccurate decision in low SNR. In this paper, We propose some multi-modulus blind equalization algorithm scheme. It is a method of operation in some multi-modulus algorithm which does no obstruct a convergence property at the initial equalization in the low SNR. Proposed algorithm improves the steady-state performance. And it uses residual ISI of the equalizer output in order to decide the optimum switching time between the single modulus and the multi-modulus algorithm.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.43-51
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.

• Journal of the Korea Computer Industry Society
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• v.7 no.1
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• pp.39-46
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• 2006

Multipath diversity reception is applied to the compensation of the distortion of channel, which occurs in transmission of data at rapid speed. DS/BPSK systems are composed of the equipment with reception structure combined with a compensation algorithm to diversity branch. The efficiency of the system is evaluated from the point of view of average bit error rate according to SNR. In order to remove intersymbol interference, this algorithm is applied to time-invariant channel at every diversity branch with the number of L. As a result, the average bit error rate is saturated by $10^{-2}$ at the conventional compensation algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.699-702
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• 2009

This paper is on a new adaptive algorithm which can cancel the acoustic feedback signals in the digital hearing aids. The proposed algorithm uses the normalized LMS algorithm with decorrelators. By doing so, it can be reduced the autocorrelation for the voice signals. As the results of simulations, it is proved that the feedback canceller adopting this algorithm shows the improved SNR of about more than 20 dB.