• Title/Summary/Keyword: 신호 지도 갱신

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A Performance Comparison of CR-MMA and CM-MMA Equalization Algorithm in 2-dimensional QAM System (2차원 QAM 시스템에서 CR-MMA 와 CM-MMA 등화 알고리즘의 성능 비교)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.18 no.4
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    • pp.85-90
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    • 2018
  • This paper compares the adaptive eualization performance of CR-MMA (Constellation Reduction-MMA) and CR-MMA (Constellation Matching-MMA) in order to minimization of intersymbol interference that is occur in channel for 2-dimensional QAM signal. For obtain the error signal in order to updating the tap coefficient of conventional adaptive equalization algorithm MMA, the CR-MMA converts the constellation reduction concept of high order 2-dimensional QAM nonconstant modulus signal to constant modulus signal and the CM-MMA use the constellation matchine error concept in order to force the matching the $2^{nd}$ and $4^{th}$ cumulant of equalizer output and transmitted signal constellation. By applying the different method for getting the error signal, these algorithm are possible to obtain the improved equalization performance compared to the conventional MMA algorithm, the improved performance of CR-MMA and CM-MMA were compared by computer simulation in this paper. As a result of simulation, the CM-MMA has more better performance in the equalizer internal performance than CR-MMA, but not in equalizer external performance as in SER.

A Performance Comparison of CM-MMA and RMMA Blind Equalization Algorithm in QAM Signal Transmission (QAM 신호 전송에서 CM-MMA와 RMMA 블라인드 등화 알고리즘의 성능 비교)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.19 no.2
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    • pp.79-84
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    • 2019
  • This paper compare the performance of CM-MMA (Constellation Matching-MMA) and RMMA (Region-based MMA) blind equalization algorithm for improve the QoS by minimizing the intersymbol interference that is occurred in nonlinear communication channel when transmitting the QAM signal. In the tap coefficient update for adaptive, CM-MMA use the error of nonconstant modulus signal adding the current MMA cost fuction and constellation matching error terms of sinusoidal power function, and the RMMA use the error by transfoms the nonconstant modulus signal of equalizer output constellation to 4-QAM constant modulus signal. They has different equalization performance by these error signal, it were compared in this paper by simulation, and performance index such as output signal constellation of equalizer, residual isi, maximum distortion, SER curves are applied for this. As a result of computer simulation, the RMMA has more better performance in the every performance index, convergence speed, residual value, noise robustness compared to CM-MMA.

A Performance Comparison of DSE-MMA and QE-MMA Adaptive Equalization Algorithm in Nonconstant Modulus Signal (Nonconstant Modulus 신호에 대한 DSE-MMA와 QE-MMA 적응 등화 알고리즘의 성능 비교)

  • Lim, Seung-Gag;Ryoo, Si-Yeong
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.21 no.2
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    • pp.67-72
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    • 2021
  • This paper compare the adative equalization performance of the DSE-MMA (Dithered Signed-Error MMA) and QE-MMA (Quantized-Error MMA) which has a simplifies the computational operation of currently used MMA algorithm. The DSE-MMA possible to improve the rubustness to noise by using the dithered signal consider the polarity of error signal in the multiplication part. In QE-MMA, it use the polarity of error signal after performing the nonlinear power-of-two quantizing operation for easiness of H/W implementation. The same channel environment was applied, and it's performance of the output signal constellation, the residual isi and maximum distortion and MSE that means the convergence characteristics, the SER that means the robustness of external noise of algorithm were compared and evaluated. As a result of computer simulation, the QE-MMA has more good in constellation, residual isi, maximum distortion, MSE performanc than DSE-MMA. In SER, the DSE-MMA has more robust due to dither signal than QE-MMA.

A Performance Analysis of Hybrid-DSE-MMA Adaptive Equalization Algorithm based on Adaptive Modulus and Adaptive Stepsize (Adaptive Modulus와 Adaptive Stepsize를 이용한 Hybrid-DSE-MMA 적응 등화 알고리즘의 성능 분석)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.21 no.4
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    • pp.75-80
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    • 2021
  • This paper relates with the Hybrid-DSE-MMA (Hybrid-Dithered Signed Error-MMA) that is possible to improving the equalization performance by using the adaptive modulus and adaptive stepsize in DSE-MMA adaptive equalizer. The DSE-MMA possible to improve the robustness performance to external noise of SE-MMA by using the sign after adding the dither signal for get the error signal in order to update the tap coefficient. But it has a drawback of performance degradation in convergence speed and residual isi by using the fixed modulus and fixed stepsize. In this paper, it was confirmed that this equalization performance degradation was improved by applying the adaptive modulus and stepsize in DSE-MMA propotional to the output power of equalizer by computer simulation. In order to compare the improved equalization performance to currently DSE-MMA, the recovered signal constellation that is the output of the equalizer, residual isi, Maximum Distortion, MSE and the SER were used as a performance index. As a result of computer simulation, the Hybrid-DSE-MMA improve the equalization performance in every index, but gives slower convergence speed compared to DSE-MMA.

Performance Improvement of SE-MMA Adaptive Equalization algorithm by Selective Updating (Selective Updating에 의한 SE-MMA 적응 등화 알고리즘의 성능 개선)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.16 no.2
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    • pp.101-106
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    • 2016
  • This paper proposes the SU-SE-MMA algorithm which applying the concept of selective updaing to the SE-MMA that is possible to reduce the intersymbol interference due to distortion occurred at the channel when transmit the nonconstant modulus 16-QAM signal. The SE-MMA emerged for the simplifying the computational operation from the current MMA adaptation algorithm, then it's has the fast convergence speed and has a problem of increase the residual component in the steady state. The SU-SE-MMA performs the selectively tap updating when the distance of equalizer output and specified transmit signal point is greater than the given threshold value and tap updaing does not occurred in the small distance. By this selective updating process, it is possible to more reduction in the computational operation in the propose algorithm. The improved adaptive equalization performance of SU-SE-MMA like as the equalizer output signal constellation, residual isi, MD, SER were confirmed by computer simulation compared to SE-MMA. As a result of simulation, the AV-SE-MMA has better performance in output signal constellation, residual isi and MD compared to the SE-MMA, but it was confirmed that the AV-SE-MMA has similar in the SER performance that means the robustness to the noise.

Convergence Speed Improvement in MMA Algorithm by Serial Connection of Two Stage Adaptive Equalizer (2단 적응 등화기의 직렬 연결에 의한 MMA 알고리즘의 수렴 속도 개선)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.15 no.3
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    • pp.99-105
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    • 2015
  • This paper deals with the mMMA (modified MMA) which possible to improving the convergence speed that employing the serial connecting form of two stage digital filter instead of signal filter of MMA adaptive equalizer without applying the variable step size for compensates the intersymbol interference by channel distortion in the nonconstant modulus signal. The adaptive equalizer can be implemented by signal digital filter using the finite order tap delay line. In this paper, the equalizer is implemented by the two stage serial form and the filter coefficient are updated by the error signal using the same algorithm of MMA in each stage. The fast convergence speed is determined in the first stage, and the residual isi left at the output of first stage output is minimized in the second stage filter. The same digital filter length was considered in single stage and two stage system and the performance of these systems were compared. The performance index includes the output signal constellation, the residual isi and maximum distortion, MSE that is measure of the convergence characteristics, the SER. As a result of computer simulation, mMMA that has a FIR structure of two stage, has more good performance in every performance index except the constellation diagram due to equalization noise and improves the convergence speed about 1.5~1.8 time than the present MMA that has a FIR structure of single stage.

Subband Sparse Adaptive Filter for Echo Cancellation in Digital Hearing Aid Vent (디지털 보청기 벤트 반향제거를 위한 부밴드 성긴 적응필터)

  • Bae, Hyeonl-Deok
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.11 no.5
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    • pp.538-542
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    • 2018
  • Echo generated in digital hearing aid vent give rise to user's discomfort. For cancelling feedback echo in vent, it is required to estimate vent impulse response exactly. The vent impulse response has time varying and sparse characteristics. The IPNLMS has been known a useful adaptive algorithm to estimate vent impulse response with these characteristics. In this paper, subband sparse adaptive filter which applying IPNLMS to subband hearing aid structure is proposed to cancel echo of vent by estimating sparse vent impulse response. In the propose method, the decomposition of input signal to subband can pre-whiten each subband signal, so adaptive filter convergence speed can be improved. And the poly phase component decomposition of adaptive filter increases sparsity of each components, and the better echo cancellation can be possible without additional computation. To derive coefficients update equation of the adaptive filter, by defining the cost function based weight NLMS is defined, and the coefficient update equation of each subband is derived. For verifying performances of the adaptive filter, convergence speed, and steady state error by white signal input, and echo cancelling results by real speech input are evaluated by comparing conventional adaptive filters.

Performance Improvement of Tree Structured Subband Filtering (트리구조 필터뱅크를 이용한 서브밴드 필터링에서의 수렴 성능 향상)

  • 최창권;조병모
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.4 no.2
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    • pp.407-416
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    • 2000
  • Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

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An Implementation of Image Security System Based on Web (웹기반 화상 감시 시스템의 구현)

  • Park Young-Seak;Cho Pyung-Kee
    • Journal of the Institute of Convergence Signal Processing
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    • v.1 no.1
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    • pp.74-81
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    • 2000
  • In this paper we have developed an image security system based on Internet Web in order to overcome the problem of existing systems that use the dedicated network. The developed system resolves the safety problem of the centralized control model by adapting the distributed control model based on Web, and has the functions of remote control and automatic monitoring for grouped multiple sites on remote Web browser. And the system can operate various computers or operating system because it' s operating software was designed by the concepts of Java Virtual Machine and Virtual Instrument. Also, our system has not need of additional cost for network construction by using Internet and can greatly improve the managemental efficiency of system because the maintenance and publishing of software updates can be performed through Web Server.

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Study on Improvement of Convergence Rate of Acoustic Echo Canceller (음향 반향 제거기의 수렴속도 개선에 대한 연구)

  • Kang, Hee Hoon
    • The Journal of the Korea institute of electronic communication sciences
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    • v.4 no.1
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    • pp.66-69
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    • 2009
  • An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

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