• The Journal of the Acoustical Society of Korea
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• 제23권7호
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• pp.540-547
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• 2004

Since speech enhancement algorithms based on Kalman/Wiener filter require a priori knowledge of the noise and have focused on the minimization of the variance of the estimation error between clean and estimated speech signal, small estimation error on the noise statistics may lead to large estimation error. However, H/sub ∞/ filter does not require any assumptions and a priori knowledge of the noise statistics, but searches the best estimated signal among the entire estimated signal by applying least upper bound, consequently it is more robust to the variation of noise statistics than Kalman/Wiener filter. In this paper, we Propose a speech enhancement method using HMM and multi H/sub ∞/ filters. First, HMM parameters are estimated with the training data. Secondly, speech is filtered with multiple number of H/sub ∞/ filters. Finally, the estimation of clean speech is obtained from the sum of the weighted filtered outputs. Experimental results shows about 1dB∼2dB SNR improvement with a slight increment of computation compared with the Kalman filter method.

• Proceedings of the Acoustical Society of Korea Conference
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• 한국음향학회 1985년도 학술발표회 논문집
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• pp.87-94
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• 1985

In this paper, enhancement of speech corrupted by additive white or colored noise is stuided. The nuconstrained frequency-domain block least-mean-square (UFBLMS) adaptation algorithm and its frequency-weighted version are newly applied to speech enhancement. For enhancement of speech degraded by white noise, the performance of the UFBLMS algorithm is superior to the spectral subtraction method or Wiener filtering technique by more than 3 dB in segmented frequency-weighted signal-to-noise ratio(FWSNERSEG) when SNR of speech is in the range of 0 to 10 dB. As for enhancement of noisy speech corrupted by colored noise, the UFBLMS algorithm is superior to that of the spectral subtraction method by about 3 to 5 dB in FWSNRSEG. Also, it yields better performance by about 2 dB in FWSNR and FWSNRSEG than that of time-domain least-mean-square (TLMS) adaptive prediction filter(APF). In view of the computational complexity and performance improvement in speech quality and intelligibility, the frequency-weighted UFBLMS algorithm appears to yield the best performance among various algorithms in enhancing noisy speech corrupted by white or colored noise.

• Journal of the Korean Institute of Telematics and Electronics
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• 제17권3호
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• pp.20-25
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• 1980

A singular value decomposition (SVD) pseudo-inversion method has been applied to the image reconstruction from projections. This approach is relatively unknown and differs from conventionally used reconstructioll methods such as the Foxier convolution and iterative techniques. In this paper, two SVD pseudo-inversion methods have been discussed for the search of optimum reconstruction and restoration, one using truncated inverse filtering, the other scalar Wiener filtering. These methods partly overcome the ill-conditioned nature of restoration problems by trading off between noise and signal quality. To test the SVD pseudo-inversion method, simulations were performed from projection data obtained from a phantom using truncated inversefiltering. The results are presented together with some limitations particular to the applications of the method to the general class of 3-D image reconstruction and restoration.

• Journal of Institute of Control, Robotics and Systems
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• 제15권12호
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• pp.1203-1208
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• 2009

Green IT based u-City means that u-City having Green IT concept. If we adopt the situation awareness or not, the processing of Green IT may be reduced. For example, if we recognize a lot of speech sound on CCTV in u-City environment, it takes a lot of processing time and cost. However, if we want recognize emergency sound on CCTV, it takes a few reduced processing cost. So, for detecting emergency state dynamically through CCTV, we propose our advanced speech recognition system. For the purpose of that, we adopt HMM (Hidden Markov Model) for feature extraction. Also, we adopt Wiener filter technique for noise elimination in many information coming from on CCTV in u-City environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 한국방송공학회 2012년도 추계학술대회
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• pp.188-190
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• 2012

Compressed Sensing (CS) has been developed for several years. Among many CS algorithms for image, the Block-based Compressed Sensing with Smoothed Projected Landweber (BCS-SPL) demonstrates its excellent performance in low-complexity and near-optimal reconstruction. Several noise filtering algorithms of image reconstruction have been introduced such as the Wiener or the median filters, etc. In general, each filter has its own advantages and disadvantages depending on specific coding scheme. In this paper, we show that reconstruction performance can be varied according to the choice of filter. When a sub-rate value is changed, different filter causes different effect as well. Concerning the sub-rate, an inner filter can be chosen to improve the reconstructed image quality.

• Proceedings of the Korea Information Processing Society Conference
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• 한국정보처리학회 2018년도 춘계학술발표대회
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• pp.457-459
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• 2018

화자인식 기술은 주어진 임의의 두 발화로부터 발화자의 일치 여부를 판단하여 등록된 화자의 목록으로부터 임의로 입력된 발화의 발화자를 식별하는 기술이다. 그러나, 배경잡음이나 반향이 존재하는 경우에는 음성신호가 왜곡되어 화자인식 성능이 저하될 수 있기 때문에 별도의 음성신호 전처리 알고리즘을 함께 사용할 수 있다. 본 논문에서는 배경잡음이 존재하는 환경에서 다수의 마이크로폰을 통해 수집한 음성신호에 대해 화자인식을 수행하는 방법으로써 parametric multi-channel Wiener filter (PMWF)를 이용한 화자일치 점수 앙상블 기법을 제안한다. 입력신호의 신호대잡음비를 기준으로 점수 결합 시 사용되는 결합계수를 정하고, Wiener filter 로 잡음을 제거하여 얻은 점수와 minimum variance distortionless response (MVDR) 빔포머를 통해 잡음을 제거하여 얻은 정수를 가중결합하는 방식으로 동일오류율을 측정한 결과, 각 전처리 알고리즘을 독립적으로 사용하여 점수를 계산한 경우보다 우수한 성능을 보임을 확인할 수 있었다.

• Proceedings of the Korean Society of Computer Information Conference
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• 한국컴퓨터정보학회 2023년도 제68차 하계학술대회논문집 31권2호
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• pp.1-2
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• 2023

본 연구는 환경 요인이 통제되어 있는 실험실 데이터에 산업 현장에서 발생하는 유사 잡음을 노이즈로 추가하였을 때, SNR비에 따른 노이즈별 STFT Log Spectrogram, Mel-Spectrogram, CWT Spectrogram 총 3가지의 이미지를 생성하고, 각 이미지를 입력으로 한 CNN 결함 분류 모델의 성능 결과를 확인하였다. 원본 데이터의 영향력이 큰 0db 이상의 SNR비로 합성할 경우 원본 데이터와 분류 결과상 큰 차이가 존재하지 않았으며, 노이즈 데이터의 영향이 큰 0db 이하의 SNR비로 합성할 경우, -20db의 STFT 이미지 기준 약 26%의 성능 저하가 발생하였다. 또한, Wiener Filtering을 통한 디노이징 처리 이후, 노이즈를 효과적으로 제거하여 분류 성능의 결과가 높아지는 점을 확인하였다.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• 제8권2호
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• pp.126-136
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• 1997

When we apply the spectrally efficient quadrature amplitude modulation(QAM) to mobile communications, it is necessary to estimate and compensate the channel charac- teristics. In this paper, a new type fading estimation method for PSAM using sinc function is presented. Gaussian interpolation method has a drawback that the performance degrades rapidly if pilot symbol period increases even though pilot sysbol period is less than Nyquist sampling rate. The Wiener filter method does not degrade until pilot symbol period is equal to the Nyquist sampling rate. It is difficult for Wiener filter method to be applied to real system because autocorrelation function of channel gain, Doppler frequency and SNR(signal to noise ratio) must be known to optimize the filter coefficients. But proposed method has a similar performance to the Wiener filter method, and does not need to know the autocorrelation function of channel gain, the doppler frequency and SNR. Therefore the proposed method cna be applied to real system easily.

• The Journal of the Acoustical Society of Korea
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• 제40권3호
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• pp.213-218
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• 2021

It is known that total least-squares method shows better estimation performance than least-squares method when noise is present at the input and output at the same time. When total least squares method is applied to data with time series characteristics, Recursive Total Least Squares (RTS) algorithm has been proposed to improve the real-time performance. However, RTLS has numerical instability in calculating the inverse matrix. In this paper, we propose an algorithm for reducing numerical instability as well as having similar convergence to RTLS. For this algorithm, we propose a new RTLS using Iterative Wiener Filter (IWF). Through the simulation, it is shown that the convergence of the proposed algorithm is similar to that of the RTLS, and the numerical robustness is superior to the RTLS.

• The Journal of the Korea institute of electronic communication sciences
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• 제14권6호
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• pp.1249-1256
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• 2019

In this paper, we propose a new algorithm to remove acoustic feedback and noise in hearing aids. Instead of using the conventional FIR structure, this algorithm is a deep learning algorithm using neural network adaptive prediction filter to improve the feedback and noise reduction performance. The feedback canceller first removes the feedback signal from the microphone signal and then removes the noise using the Wiener filter technique. Noise elimination is to estimate the speech from the speech signal containing noise using the linear prediction model according to the periodicity of the speech signal. In order to ensure stable convergence of two adaptive systems in a loop, coefficient updates of the feedback canceller and noise canceller are separated and converged using the residual error signal generated after the cancellation. In order to verify the performance of the feedback and noise canceller proposed in this study, a simulation program was written and simulated. Experimental results show that the proposed deep learning algorithm improves the signal to feedback ratio(: SFR) of about 10 dB in the feedback canceller and the signal to noise ratio enhancement(: SNRE) of about 3 dB in the noise canceller than the conventional FIR structure.