• The Journal of the Acoustical Society of Korea
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• v.29 no.2
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• pp.148-153
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• 2010

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. A Speech signal consists of Voice and Unvoiced Band in The MBE excitation model. And the energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced. We use the IMBE vocoder for the Voice or Unvoice band of segmented speech signal. Continuously we calculate the segmented SNR using that information and the energy of each band. And we estimate the SNR of continuous speech signal.

• Journal of the Korean Society of Radiology
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• v.13 no.3
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• pp.453-458
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• 2019

In this study we analyzed the tendency of the image characteristic by changing filtering factor for the proposed fast non local means (FNLM) noise reduction algorithm with designed Male Adult mesh (MASH) phantom through Geant4 application for tomographic emission (GATE) simulation program. To accomplish this purpose, MASH phantom for human copy was designed through the GATE simulation program. In addition, we acquired degraded image by adding Gaussian noise with a value of 0.005 using the MATALB program in MASH phantom. Moreover, in degraded image, the FNLM noise reduction algorithm was applied by changing the filtering factors, which set to 0.005, 0.01, 0.05, 0.1, 0.5, and 1.0 value, respectively. To quantitatively evaluate, the coefficient of variation (COV), signal to noise ratio (SNR), and contrast to noise ratio (CNR) were calculated in reconstructed images. Results of the COV, SNR and CNR were most improved in image with a filtering factor of 0.05 value. Especially, the COV was decreased with increasing filtering factor, and showed nearly constant values after 0.05 value of the filtering factor. In addition, SNR and CNR were showed that improvement with increasing filtering factor, and deterioration after 0.05 value of the filtering factor. In conclusion, we demonstrated the significance of setting the filtering factor when applying the FNLM noise reduction algorithm in degraded image.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.383-391
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• 2009

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. The waveform which is stationary region of voiced speech is very correlated by pitch period. So we can estimate the SNR by correlation of near waveform after dividing a frame for each pitch. For unvoiced speech signal, vocal track characteristic is reflected by noise, so we can estimate SNR by using spectral distance between spectrum of received signal and estimated vocal track. Lastly, energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.4
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• pp.555-560
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• 2013

This paper first proposes a classification algorithm which detects a voiced, unvoiced, and silence signal using distance measure, logarithm power and root mean square methods at each frame, then a spectrum subtraction algorithm based on a subband filter. The proposed algorithm subtracts spectrums of white noise and street noise from noisy signal based on the subband filter at each frame. In this experiment, experimental results of the proposed spectrum subtraction algorithm demonstrate using the speech and noise data of Aurora-2 database. Based on measuring the speech-to-noise ratio (SNR), experiments confirm that the proposed algorithm is effective for the speech by contaminated the noise. From the experiments, the improvement in the output SNR values was approximately 2.1 dB and 1.91 dB better for white noise and street noise, respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.8A
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• pp.723-729
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• 2011

FCC of U.S. permitted usage of unlicensed system on unused spectrum in TV white space after DTV transition. The unlicensed systems are required to avoid harmful interference to licensed users by employing geo-location database and spectrum sensing. The conventional spectrum sensing algorithms for ATSC signal were focused on detection of pilot signal. However, they can not guarantee detection of ATSC signal when pilot signal is attenuated by channel environment such as fading. To overcome drawbacks of conventional schemes, in this paper, we propose a signal detection and data fusion algorithm using cyclo-stationary feature weighted by signal energy. Simulation results verify that the proposed algorithm can provide 2dB SNR gain for 90% detection probability compare with the conventional scheme. We can reduce quiet period for spectrum sensing and improve signal detection probability by employing the proposed algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.5
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• pp.811-816
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• 2019

This paper first proposes a technique to separate clean speech signals and mixed noise signals by using an independent vector analysis algorithm of frequency band for 4 channel speech source signals with a noise. An improved output speech signal from the proposed independent vector analysis algorithm is obtained by using the cross-correlation between the signal outputs from the frequency domain delay-sum beamforming and the output signals separated from the proposed independent vector analysis algorithm. In the experiments, the proposed algorithm improves the maximum SNRs of 10.90dB and the segmental SNRs of 10.02dB compared with the frequency domain delay-sum beamforming algorithm for the input mixed noise speeches with 0dB and -5dB SNRs including white noise, respectively. Therefore, it can be seen from this experiment and consideration that the speech quality of this proposed algorithm is improved compared to the frequency domain delay-sum beamforming algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.529-536
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• 2003

In this Paper, a new DCT HLMS DFE(Discrete Cosine Transform Hierarchical Least Mean Square Decision Feedback Equalizer) algorithm is proposed to improve the convergence speed and MSE(Mean Square Error) performance of a receive channel equalizer in ATSC(Advanced Television System Committee) 8VSB(Vestigial Side Band) digital terrestrial TV system. The proposed algorithm reduces the eigenvalue range of input data autocorrelation by transforming LMS (Least Mean Square) DFE into the subfilter of hierarchical structure. Moreover, the use of DCT and power estimation algorithm makes it possible to reduce the eigenvalue deviation of input data which results from distortion and delay of the receive signal in the miulti-path environment. Simulation results show that proposed DCT HLMS DFE has SNR improvement of approximately 3.8dB, 5dB and 2dB as compared to LMS DFE when the equalized symbol error rate is 0.2 in ATTC defined digital terrestrial TV broadcasting channels A, B and F, respectively.

• The Journal of the Acoustical Society of Korea
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• v.14 no.1
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• pp.73-80
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• 1995

In this paper, Polarity Thresholding(PT) algorithm has been studied to enhance the received signal in ultrasonic inspection of the stainless-steel(SUS 304) which is the primary piping material of a nuclear power plant. The spectral decomposition components obtained by splitting the spectrum of received signals are composed of dispersive signal of the interference pattern produced by the grain boundaries and nondispersive signal by the flaw. PT algorithm enhance the SNR of received signal by using above properties. In experiment the stainless-steel has been chosen as the sample and heat-treated at 1125, 1150, 1175, and $1200^\circ{C}$, respectively. And the flat-bottom hole type defects have been made artificially in samples. The pulse-echo signals from the sample by using ultrasonic transducer of center frequency 5 MHz have been processed by PT algorithm. It has been shown that PT algorithm enhanced the SNR by average 14.2 dB.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.271-274
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• 2004

본 논문에서는 매 프레임 단위로 새롭게 노이즈를 추정하는 방법을 적용하는 전처리 기법을 이용하여 적응 지각필터의 속도를 향상하는 알고리즘을 제안한다. 제안된 전처리 잡음 추정 알고리즘은 잡음에 열화 된 대역으로부터 잡음을 추정하여 적응 지각 필터에 적용함으로써 오디오 신호의 음질을 개선하는 알고리즘이다. 이는 처리되는 신호 구간에 따라 잡음에 열화 된 대역으로부터 잡음을 추정함으로써 초기 추정 잡음에 보다 가까운 추정 잡음을 얻을 수 있다. 결과적으로 적응 지각 필터의 수행 횟수를 효과적으로 줄일 수 있다. 이는 기존의 묶음 구간에서 추정잡음을 이용한 적응 지각 필터의 SNR 및 MNR 비교와 적응 지각 필터 적용 횟수, 동작 시간 등을 이용하여 개선을 확인할 수 있다.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2562-2570
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• 1997

This paper deals with the design and performance of the adaptive equalizer using high order statistics in order to improve the transmission characteristics of multipath fading channel. The multipath propagational phenomenon occurred in digital radio transmission causes the distortion and ISI of receiving signal. These are main reasons to increase the bit error rate and degrade the performance of receivers. In this paper, the adaptive equalization algorithm using high order statistics of received signal is used instead of CMA algorithm, Bussgang and Godard which are known widely. The performance of this algorithm (residualisi, recovered constellation, calculation) is presented varing SNR. As the result of the computer simulation, equalizer algorithm using high order statistics is better than CMA in the range of low SNR, $10{\sim}20dB$. Therefore, considering the actual communication systems which use the range of $14{\sim}20$ SNR, the adaptive equalizer using high order statistics can be used in the real multipath fading environment.