• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1986.04a
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• pp.1-3
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• 1986

We are concerned with transmitting numerical source data of {0, 1, 2, ..., 2k-1} through a channel coding system. The rate 1/v dual-k convolutional code with the orthogonal MFSK modulation and the Viterbl decoding is employed for the implementation of the channel coding system. The mean square error of the dual-k convolutional code is evaluated for the numerical source transmitted over an additive white Gaussian noise channel with Rayleigh fading.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.12
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• pp.1910-1917
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• 1989

This paper shows the performances of asynchronous binary direct-sequence spread-spectrum multiple access communication systems with Rayleigh fading and White Gaussian noise. The performance measures considered are worst-case bit error probability and average SNR depending on code sequences and chip waveforms. The code sequences used are m-sequence and Gold sequence with period 31.The chip waveforms employed are rectangular, sinusoidal and something other chip waveforms.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.25-28
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• 2000

시변 특성을 갖는 무선 채널 환경에서 UEP(Unequal Error Protection) 채널 부호화 방법은 효율적인 정보 전송을 가능케 한다. 본 논문에서는 IP-SCCC (Interleaved Puncturing Serially Concatenated Convolutional Codes)로 CSI(Channel State Information)와 SSI(Source Significance Information)를 이용한 두가지 UEP 채널 부호화 방법을 제안한다. 또한 AWGN(Additive White Gaussian Noise)과 Rayleigh 페이딩이 존재하는 채널을 가정하고 모의 실험을 통해 알고리즘의 우수성을 검증한다.

• 전자공학회논문지 IE
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• v.49 no.2
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• pp.9-17
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• 2012

One of the problems on the application of the speech recognition system in the real world is the degradation of the performance by acoustical distortions. The most important source of acoustical distortion is the additive noise. This paper describes a spectral compensation technique based on a spectral peak enhancement scheme followed by an efficient noise subtraction scheme for noise robust speech recognition. The proposed methods emphasize the formant structure and compensate the spectral tilt of the speech spectrum while maintaining broad-bandwidth spectral components. The recognition experiments was conducted using noisy speech corrupted by white Gaussian noise, car noise, babble noise or subway noise. The new technique reduced the average error rate slightly under high SNR(Signal to Noise Ratio) environment, and significantly reduced the average error rate by 1/2 under low SNR(10 dB) environment when compared with the case of without spectral compensations.

• The Journal of the Acoustical Society of Korea
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• v.16 no.6
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• pp.25-35
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• 1997

Most of speech analysis methods developed up to date are based on second order statistics, and one of the biggest drawback of these methods is that they show dramatical performance degradation in noisy environments. On the contrary, the methods using higher order statistics(HOS), which has the property of suppressing Gaussian noise, enable robust feature extraction in noisy environments. In this paper we propose a text-independent speaker identification system using higher order statistics and compare its performance with that using the conventional second-order-statistics-based method in both white and colored noise environments. The proposed speaker identification system is based on the vector quantization approach, and employs HOS-based voiced/unvoiced detector in order to extract feature parameters for voiced speech only, which has non-Gaussian distribution and is known to contain most of speaker-specific characteristics. Experimental results using 50 speaker's database show that higher-order-statistics-based method gives a better identificaiton performance than the conventional second-order-statistics-based method in noisy environments.

• Journal of information and communication convergence engineering
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• v.9 no.6
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• pp.763-768
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• 2011

Image denoising is a lively research field. Today the researches are focus on the wavelet domain especially using wavelet threshold method. We proposed an adaptive threshold method which considering the characteristic of different sub-band, the method is adaptive to each sub-band. Experiment results show that the proposed method extracts white Gaussian noise from original signals in each step scale and eliminates the noise effectively. In addition, the method also preserves the detail information of the original image, obtaining superior quality image with higher peak signal to noise ratio(PSNR).

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2003.09a
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• pp.563-566
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• 2003

Quality measures play an important role in the field of image processing. Such measures are commonly used to assess the performance of different algorithms that are designed to perform a specific image processing task. In this paper we propose two novel measures for image quality assessment based on the notion of correlation between fuzzy sets. Two different definitions fur the correlation between fuzzy sets have been used. In order to calculate the proposed quality measures two approaches were evaluated, one with direct application of the measures to the image′s pixels and the other using the fuzzy set corresponding to the normalized histogram of the image. A comparative study of the proposed measures is performed by investigating their behavior using images with different types of distortions, such as impulsive "salt at pepper" noise, additive white Gaussian noise, multiplicative speckle noise, blurring, gamma distortion, and JPEG compression.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.969-974
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• 2007

In this paper, a beamforming technique is introduced to measure the acoustic impedance at both normal and oblique incidence in a free field. The acoustic impedance is obtained by separating incident and reflected signals using the adaptive nulling method which is one of the various beamforming algorithms. To obtain better results, pressure vector commonly used in array signal processing is replaced with the transfer function vector between each microphone and the white Gaussian noise is suppressed by a wavelet shrinkage technique. The experiments conducted in a semi-anechoic room show that the proposed method is efficient and accurate in measuring the acoustic impedance of sound absorbing materials under a free field condition.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.76-80
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• 1992

We consider the tapped-delay line (TDL) equalizer with the few calculation quantity and the simplity, the decision feedback equalizer (DFE) with the good property for interference, and lattice equalizer(LE) with high insensitivity to roundoff noise in mobile communication fading channel. The used adaptive algorithm is the LMS algorithm and RLS algorithm. In this paper, we have evaluated the performance of the TDL equalizer, the decision feedback equalizer, and lattice-structured equalizer, for the digital signal corrupted by the impulsive noise and the white gaussian noise under the fading channel environment. From the results of error performance analysis, it is confirmed that lattice-structured equalizer has better performance than DFE equalizer and TDL equalizer.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.11
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• pp.2942-2953
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• 1996

This paper studies the effect of intersymbol interference due to sampling-time jitter on the worst-case bit error probability in a digital modultation over an additive white Gaussian noise channel, with the squared-root raised-cosine filters in the transmitter and the receiver. It derives approximation formulas using the Taylor series approximations. the principal results of this paper is the relationship between the worst-casse bit error probability, the degree of jitter, the roll factor of the raised cosine filter, and other quantities. Numerical results show, as expected, that the intersymbol interference decreases as the roll-off factor increases and the jitter decreases. They also show that the approximation formulas are accurate for smally intersymbol interference, i.e., for large roll-noise ratio $E_{b/}$ $N_{0}$.leq.7 dB and begin to lose accuracy for larger signal-to-noise ratio.o.o.