• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.3B
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• pp.295-301
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• 2001

전송과정에서 발생하는 왜곡량에 따라 전송률의 하한이 결정되는 율-왜곡 이론은 시간적으로 민감한 부분을 왜곡없이 부호화하여 전송하는 영상 시스템에서 기본이 되는 중요한 요소이다. 율-왜곡 이론은 정보량의 개념으로부터 시작되어 원 신호의 확률분포와 왜곡의 측정기준에 의해 결정되는데, 이 논문에서는 가우시안과 라플라시안 분포함수에 절대치 오차기준과 자승 오차기준을 적용하여 율-왜곡 함수를 각각 구하였다. 나아가서 저전송률 부호기로 개발된 H.263 부호기에 이 함수를 적용하여 분석하였다. 비교를 위해 자승 오차기준에 위한 이론치와 실제 측정치를 제시하였다. H.263 부호기는 엔트로피 부호화, 부호화를 블록 패턴 등 다양한 기법을 사용하여 율-왜곡 함수에 의한 이론치보다 주어진 MSE에서 정규화 비트율이 최대 0.55만큼 더 낮은 전송률을 얻을 수 있었다.

• 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 Institute of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.15-22
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• 2012

This paper deals with the performance comparison and evaluation of blind adaptive equalizer, the PMCMA-MDD and DW-MCMA, that is used for compensation of the amplitude and phase distortion which occurs in the time dispersive channel. Basically, these algorithms are modification of MCMA cost function in order to obtain the fast convergence speed and reduced residual isi by taking the parallel and serial double structured and the combination of the concept of RCA for the updating the tap coefficient. We implements the algorithm of it and compare the recovered constellation, residual isi, MSE characteristics curve and SER in the signal to noise ratio given the time dispersive channel. As a result of simulation, the PMCMA-MDD has a good in recovered constellation than DW-MCMA. But in the SER, the DW-MCMA has a good than PMCMA-MDD.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.19 no.5
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• pp.107-113
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• 2019

This paper proposes the Hybrid-CRMMA adaptive equalization algorithm that is possible to improves the performance of CR-MMA based on adaptive modulus and adaptive stepsize. The 16-QAM nonconstant modulus signal is reduced to 4-QAM constant modulus signal, and the error signal were obtained based on the fixed statistic modulus of transmitted signal. It is possible to improving the currently MMA adaptive equalization performance. The proposed Hybrid-CRMMA composed of adaptive modulus which is propotional to the power of equalizer output and adaptive stepsize which is function of the nonlinearties of error signal, and its improved equalization performance were confirmed by computer simulation. For this purpose, the output signal constellation, the residual isi and maximum distortion and MSE that is for the convergence characteristics, the SER that is meaning the robustness of external noise of algorithm were used. As a result of computer simulation, it was confirmed that the proposed Hybrid-CRMMA has more superior performance in every index compared to currently CR-MMA.

• The Journal of the Acoustical Society of Korea
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• v.41 no.1
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• pp.38-44
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• 2022

Speech enhancement is performed to improve intelligibility and quality of the noise-corrupted speech. In this paper, speech enhancement performance was compared using different loss functions in time and frequency domains. This study proposes a combination of loss functions to utilize advantage of each domain by considering both the details of spectrum and the speech waveform. In our study, Scale Invariant-Source to Noise Ratio (SI-SNR) is used for the time domain loss function, and Mean Squared Error (MSE) is used for the frequency domain, which is calculated over the complex-valued spectrum and magnitude spectrum. The phase loss is obtained using the sin function. Speech enhancement result is evaluated using Source-to-Distortion Ratio (SDR), Perceptual Evaluation of Speech Quality (PESQ), and Short-Time Objective Intelligibility (STOI). In order to confirm the result of speech enhancement, resulting spectrograms are also compared. The experimental results over the TIMIT database show the highest performance when using combination of SI-SNR and magnitude loss functions.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.3
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• pp.63-68
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• 2014

This paper related with the I-CMA (Improved-CMA) algorithm that is possible to compensates of phase in CMA adatpve equalizer which is used for the elemination of intersymbol interference in the multipath fading and band limit characteristics of channel. The new cost function is proposed for the eliminate the amplitude and phase simulataneous by modifying the cost fuction for get the error signal in present CMA algorithm. It has a merit to the algorithm simplicities and eliminats the PLL device for phase compensation after equalization. For proving this, the recovered signal constellation that is the output of equalizer output signal and the residual isi and Maximum Distortion charateristic learning curve that are presents the convergence performance in the equalizer and the overall frequency transfer function of channel and equalizer were used. As a result of computer simulation, the I-CMA has more good compensation capability of amplitude and phas in the recovered constellation. But the convergence time is slow due to the simultaneously phase compensation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.668-675
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• 2002

This paper proposes pre-distorter combined LUT with 2ιpartition interpolation method to reduce nonlinear distortion which was caused by high PAPR and to update LUT quickly. Pre-distorted gain and phase can be found by using LUT which consisted of AM/AM and AM/PM value, and OFDM signal amplitude. The proposed 2ιpartition interpolation can accurately find predistorted gain and phase using bit shift and add component instead of increasing size of LUT which requires increasing the amount of computation. The performance of the proposed method was measured by the difference between HPA input and output characteristics by the LUT size, constellation, SER performance by the HPA, and LUT update error by the HPA characteristic changes. As a result, it is shown that when the size of the LUT is 32 and 64 for 16-QAM and 64-QAM, nonlinear distortion nearly didn't occurred.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.21 no.1
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• pp.55-60
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• 2021

This paper relates with the performance evaluation of QE-MMA (Quantized Error-MMA) adaptive equalization algorithm based on the stepsize and quantizer bit number in order to reduce the intersymbol interference due to nonlinear distortion occurred in the time dispersive channel. The QE-MMA was proposed using the power-of-two arithmetic for the H/W implementation easiness substitutes the multiplication and addition into the shift and addition in the tap coefficient updates process that modifies the SE-MMA which use the high-order statistics of transmitted signal and sign of error signal. But it has different adaptive equalization performance by the step size and quantizer bit number for obtain the sign of error in the generation of error signal in QE-MMA, and it was confirmed by computer simulation. As a simulation, it was confirmed that the convergence speed for reaching steady state depend on stepsize and the residual quantities after steady state depend on the quantizer bit number in the QE-MMA adaptive equalization algorithm performance.

• Journal of IKEEE
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• v.27 no.4
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• pp.464-471
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• 2023

In this paper, we propose a method to reduce age distortion in facial expression image generation using StyleGAN Encoder. The facial expression image generation process first creates a face image using StyleGAN Encoder, and changes the expression by applying the learned boundary to the latent vector using SVM. However, when learning the boundary of a smiling expression, age distortion occurs due to changes in facial expression. The smile boundary created in SVM learning for smiling expressions includes wrinkles caused by changes in facial expressions as learning elements, and it is determined that age characteristics were also learned. To solve this problem, the proposed method calculates the correlation coefficient between the smile boundary and the age boundary and uses this to introduce a method of adjusting the age boundary at the smile boundary in proportion to the correlation coefficient. To confirm the effectiveness of the proposed method, the results of an experiment using the FFHQ dataset, a publicly available standard face dataset, and measuring the FID score are as follows. In the smile image, compared to the existing method, the FID score of the smile image generated by the ground truth and the proposed method was improved by about 0.46. In addition, compared to the existing method in the smile image, the FID score of the image generated by StyleGAN Encoder and the smile image generated by the proposed method improved by about 1.031. In non-smile images, compared to the existing method, the FID score of the non-smile image generated by the ground truth and the method proposed in this paper was improved by about 2.25. In addition, compared to the existing method in non-smile images, it was confirmed that the FID score of the image generated by StyleGAN Encoder and the non-smile image generated by the proposed method improved by about 1.908. Meanwhile, as a result of estimating the age of each generated facial expression image and measuring the estimated age and MSE of the image generated with StyleGAN Encoder, compared to the existing method, the proposed method has an average age of about 1.5 in smile images and about 1.63 in non-smile images. Performance was improved, proving the effectiveness of the proposed method.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.4
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• pp.7-13
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• 2012

The performance of blind equalization algorithm ISCA was compared with MSCA that is used for the minimization of the inter symbol interference which occurs in the time dispersive communication channel for digital transmission. Because of the non-linearities of a magnitude and phase transfer characteristics of a communication channel, the transmitting signal will be received that band limited and time dispersived. Therefore the distortion was compensated by using the self adaptive equalizer at the receiving side, then passing through the detector for the decision of "1" or "0". At this time the Constellation Dependent Constant is played an important role in the adaptive equalizer used on the receiver. In order to calculation of this constant, the ISCA and MSCA was used the second order statistics. The ISCA and MSCA which are possible to compensation of mensioned transfer function simulataneously, are improved the performance of original SCA algorithm and then was compared the performance by computer simulation. For this, the recovered constellation, residual isi and MSE was used, and a result of performance comparison, the ISCA algorithm has better than the MSCA in every performance index. But on the steady state of equalizer, the variation of performance due to the CME terms in the MSCA equalization algorithm was less than the ISCA, so MSCA has better stability.