• The Journal of the Korea Contents Association
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• v.17 no.9
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• pp.99-106
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• 2017

This study focused on the issue that when a diagnostic detector is found to have a defect, a patient would be exposed to radiation and image quality would be degraded. Though dose analysis, an experiment was conducted to evaluate detector performance as Signal to Noise Ratio (SNR) and Contrast to Noise Ratio (CNR). Absorbed dose, SNR and CNR were measured using a dosimeter and a tissue equivalent phantom. The experiment was conducted to compare whether the dose value shown after being attached to the back side of the phantom matches the dose value attached behind the detector, where in the conditions of skull, chest and abdomen were set at 75 kVp, 25 mAs, 110 kVp, 8 mAs, and 80 kVp, 20 mAs, respectively. As a result, there was a difference in that the dose values attached to the back side of the detector were 0.004 mGy, 0.006 mGy, 0.003 mGy, whereas those of the back side of the phantom were 0.006 mGy, 0.016 mGy, 0.017 mGy. In order to match both values, the condition was increased and SNR and CNR also increased from 88.32, 88.10, 4.09, 1.63, 87.94, 79.97 to 93.87, 93.75, 4.91, 4.03, 92.02, 84.92. Though this study, we found that when a detector is found to have a aging, it shortens the life of equipment and increases the dose of a patient, also the improvement effect of image quality is inadequate.

• Proceedings of the KIEE Conference
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• 2004.07d
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• pp.2685-2686
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• 2004

심자도(Magnetocardiography: MCG)는 심장에서 발생하는 자기신호로 크기가 수 pico Tesla에서 수 femto Tesla 정도로 지구 자기에 비하여 $10^{-6}{\sim}10^{-10}$ 정도로 매우 작기 때문에 보통 3층의 차폐 막 구조로 되어 있는 자기차폐실을 사용하여 외부 잡음을 줄인다. 그러나 자기차폐실의 비용이 크기 때문에, 자기차폐실의 비용을 줄이고 다양한 신호처리를 병행하여 신호대 잡음비를 높이고 있다. 본 논문에서는 1Giga FLOPS (FLoating point Operationals Per Second)의 부동 소숫점 연산능력을 가진 TMS320C6701을 사용하여 실시간 신호처리가 가능한 신호처리 시스템을 설계하였다. 개발된 DSP 보드는 PCI-bus 기반으로 설계하여 신호 측정 컴퓨터에 내장이 가능하도록 하였다. 프로그램과 데이터 처리를 위한 외부 메모리를 장착하였고, PCI 콘트롤러를 갖추어 PC 와의 대용량 메모리 공유가 가능하도록 하였다. 제작된 DSP 보드를 사용하여, 심자도 신호에서 실시간으로 적응 잡음 소거 및 필터링을 구현하여 신호대 잡음비의 향상을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.22 no.8
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• pp.687-693
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• 2003

In this paper, a new adaptive audio signal enhancement algorithm is proposed. In order to remove a broadband noise from a noisy signal, a filter is designed and applied adaptively to noisy audio signal. The noisy signal is first transformed to frequency domain and divided into bark domain to calculate excitation energy. A filter will be calculated to eliminate the noise by using the excitation energy and noisy energy which is obtained from a silent area. The filter is adaptively adjusted and continuously applied until the threshold point is met. The algorithm also works well even though the noise's energy change all of a sudden. SNR, NMR comparison and MOS Test are performed to show the effectiveness of the proposed algorithm.

• Journal of the Korean Institute of Telematics and Electronics
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• v.17 no.3
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• pp.52-56
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• 1980

The Signal -to- Noise Ratio of delta modulation for a fi rEt -order Gauss -Markov signal is derived and an approximate expreession of SND is discussed, in the case that only granular noise arises. Cross covariance of input and error signals are negligible when the adjacent correlation of input signal is larger than the difference between the adjacent correlation and the prediction coefficient of local decoder. The approximately derived SNR is available for any value of adjacent correlation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.10A
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• pp.1782-1793
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• 2001

터보부호 심벌비트들을 서로 다른 부호율로 반복부호화한 뒤 여러 부채널로 나란히 보내는 터보 부호화 여러반송파 직접수열 부호분할 다중접속 시스템을 제안한다. 레일리 감쇄채널에서 신호대잡음비가 높을 때에는 이 시스템의 성능 한계를 얻었고, 신호대잡음비가 낮아 한계를 적용할 수 없을 때에는 모의실험에서 성능을 살펴보았다. 신호대잡음비가 낮을 때 성능 손실이 조금 있지만 홀짝 검사(parity check) 심벌에 다양성 차수를 크게 주어 오류 마루를 (error floor) 낮출 수 있음을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.17 no.8
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• pp.93-97
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• 1998

미지신호에 대한 측정에서는 오차분산의 명확한 표현이 어렵다. 본 연구에서는 이와 같은 수동측정에서의 측정정확도를 예측하기 위한 방법을 정립한다. 제안된 예측 방법은 신 뢰도를 기반으로 신호대잡음비 및 주파수대역폭과 시간적분과의 곱에 의해 정확도를 표현한 다. 본 논문에서는 배경소음 환경에서의 측정음향을 그대로 분석하는 경우와 배경소음을 예 측하여 측정음향에서 제거시키는 경우를 구분하여 광대역 음향측정시의 성능예측 기준을 시 뮬레이션하고 정확도를 예측한다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.173-181
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• 2002

Generally the one of serious problems in Voice Activity Detection (VAD) is speech region detection in noise environment. Therefore, this paper propose the new method using energy, lsp varation. As a result of processing time and speech quality of the proposed algorithm, the processing time is reduced due to the accurate detection of inactive period, and there is almot no difference in the subjective quality test. As a result of bit rate, proposed algorithm measures the number of VAD=1 and the result shows predominant reduction of bit rate as SNR of noisy speech is low (about 5∼10 dB).

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.323-329
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• 2002

This paper proposes a frame reliability weighting method to compensate for a time-selective noise that occurs at random positions of speech signal contaminating certain parts of the speech signal. Speech frames have different degrees of reliability and the reliability is proportional to SNR (signal-to noise ratio). While it is feasible to estimate frame Sl? by using the noise information from non-speech interval under a stationary noisy situation, it is difficult to obtain noise spectrum for a time-selective noise. Therefore, we used statistical models of clean speech for the estimation of the frame reliability. The proposed MFR (model-based frame reliability) approximates frame SNR values using filterbank energy vectors that are obtained by the inverse transformation of input MFCC (mal-frequency cepstral coefficient) vectors and mean vectors of a reference model. Experiments on various burnt noises revealed that the proposed method could represent the frame reliability effectively. We could improve the recognition performance by using MFR values as weighting factors at the likelihood calculation step.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.217-220
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• 2002

본 연구는 폐쇄된 임의의 공간상에서 2개의 마이크로폰 어레이를 이용하여 마이크로폰에 수신된 신호들의 도착 시간차를 추정하는 새로운 알고리즘을 제안한다. 제안된 알고리즘은 입력 음성신호를 Discrete wavelet transform을 이용하여 인간의 청각 특성과 가장 유사한 주파수 해상도를 갖도록 대역 분할한 후 각 주파수 대역에서 신호 대 잡음비를 구하여 신호 대 잡음비가 가장 높은 대역만 선택적으로 취하고 해당 대역에서만 최종적인 시간 지연 값을 추정하게 된다. 최종 시간 지연 측정에 사용된 알고리즘은 기존의 CPSP에 해당 대역의 주파수 SNR을 가중치로 주어 구하게된다. 이러한 대역 분할 가중방식은 다양한 형태의 동적인 잡음 환경 하에서 안정적인 성능을 가질 수 있다. 제안된 알고리즘은 저주파와 고주파 각각의 모의 잡음환경 하에서 컴퓨터 실험을 통해 성능을 입증하도록 한다.

• Journal of Korea Multimedia Society
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• v.4 no.1
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• pp.46-55
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• 2001

In this paper, a new voice activity detection method estimating SNR of enhanced speech with extended spectral subtraction (ESS) is proposed. Voice activity detection is performed by putting an second Wiener filter behind an Wiener filter used in the ESS to estimate speech and noise power of output signal of first Wiener filter. The proposed voice activity detection method does not require many computational loads and performs well under severe input SNR. Boll's spectral substraction algorithm with proposed voice activity detection was compared to ESS under several noise environment having different time-frequency distributions. During speech and non-speech activity, performance of Boll's spectral substraction algorithm with proposed voice activity detection is superior to that of ESS.