• Title/Summary/Keyword: channel equalizer

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Design and implementation of a viterbi decoder for a soft output equalizer in the DSC 1800 radio system (DCS 1800 시스템에서 연판정 출력 등화기에 대한 비터비 복호기 설계 및 구현)

  • 김주응;윤석현;이재혁;강창언
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.35S no.3
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    • pp.19-28
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    • 1998
  • This paper is concerned with the implementation of the equalization technique in a DCS 1800 system employing the soft-decision output Viterbi algorithm (SOVA), which makes the hardware complexity comparable to the hard decision MLSE and gives reliable performance. Also, the channel estimation technique with enhances the perfdormance of the soft-decision output equalizer is proposed, and the Viterbi decoder which operates effectively with the soft-decision output of the qualizer is implemented using the Very High Speed ICs Hardware Description Language (VHDL). From the simulation results, it is shown that the implemented Viterbi decoder operates effectively and the SOVA outperforms the hard-decision MLSE in terms of the frame erasure rate (FER) and bit error rate (BER).

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A Study on Individual Tap-Power Estimation for Improvement of Adaptive Equalizer Performance

  • Kim, Nam-Yong
    • Journal of electromagnetic engineering and science
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    • v.4 no.1
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    • pp.23-29
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    • 2004
  • In this paper we analyze convergence constraints and time constant of IT-LMS algorithm and derive a method of making it's time constant independent of signal power by using input variance estimation. The method for estimating the input variance is to use a single-pole low-pass filter(LPF) with common smoothing parameter value, θ. The estimator is with narrow bandwidth for large θ but with wide bandwidth for small θ. This small θ gives long term average estimation(low frequency) of the fluctuating input variance well as short term variations (high frequency) of the input power. In our simulations of multipath communication channel equalization environments, the method with large θ has shown not as much improved convergence speed as the speed of the original IT-LMS algorithm. The proposed method with small θ=0.01 reach its minimum MSE in 100 samples whereas the IT-LMS converges in 200 samples. This shows the proposed, tap-power normalized IT-LMS algorithm can be applied more effectively to digital wireless communication systems.

Improved Multiplication Free Adaptive Digital Filter with the Fractionally-Spaced Equalizer (분할등화기를 이용한 개선된 비적적응필터)

  • Yoon, Dal-Hwan
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.39 no.2
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    • pp.137-146
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    • 2002
  • In order to remove the intersymbol interference(ISI) phenomenon in data transmission channel, the structure and convergence analysis of the improved multiplication free adaptive digital filter(IMADF) is presented. Under conditions of zero-mean, wide-sense stationary and white Gaussian noise, it is shown that this paper analyze the convergence characteristics of the IMADF with a fractionally-spaced equalizer(FSE). In the experimental results, the convergence characteristics of the IMADF algorithm is almost same as the sign algorithm, but is better than the MADF algorithm. Here, this algorithm has useful characteristics when the correlation of the input signal is highly.

Performance Comparison of Single-Carrier and Multi-Carrier Systems in a Terahertz Wireless Communication Environment

  • Asiedu, Derek Kwaku Pobi;Ahiadormey, Roger Kwao;Shin, Suho;Lee, Kyoung-Jae
    • Journal of Advanced Information Technology and Convergence
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    • v.9 no.1
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    • pp.11-24
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    • 2019
  • This paper investigates the performance comparison of a Terahertz (THz) communications for a single-carrier and a multi-carrier single antenna point-to-point communication system. The multi-carrier system and single carrier system consider the orthogonal frequency division multiplexing (OFDM) and the minimum mean square error linear equalizer (MMSE-LE), respectively. We compare the frame-error-rate (FER) and throughput performance of both the systems for a THz communication environment with the carrier frequency of 300GHz and the tapped delay line (TDL) channel models described in 3GPP. It is observed from the simulation results that the OFDM systems outperform the MMSE-LE for various configurations.

Performance of Interference Cancellation for Cooperative Communication Systems with Maximum Likelihood Equalizer (최대 우도 등화기를 적용한 협력통신 시스템의 간섭 제거 성능)

  • Kim, Joo-Chan;Kim, Jin-Young
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.10 no.2
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    • pp.7-12
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    • 2010
  • In this paper, we analyze and simulate the performance of a cooperative communication system adopting a maximum likelihood (ML) equalizer. In wireless communication systems, cooperative communication schemes employing several relays can be applied for extending the communication coverage. It is assumed that both relays and user terminals can move. Therefore, coverages of two or more relays can overlap each other. If wanted and interfering signals are transmitted through the same channel and there are one terminal in the overlapped region, its performance is degraded due to interference. Hence, we use a ML equalizer for rejecting the effect of interfering signal and enhancing the communication system performance. The cooperative system performance is evaluated in terms of bit error probability. From the simulation results, it is demonstrated that the ML receiver shows good interference cancellation performance although its complexity is high.

Design of Digital Signal Processor for Ethernet Receiver Using TP Cable (TP 케이블을 이용하는 이더넷 수신기를 위한 디지털 신호 처리부 설계)

  • Hong, Ju-Hyung;SunWoo, Myung-Hoon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.8A
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    • pp.785-793
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    • 2007
  • This paper presents the digital signal processing submodule of a 100Base-TX Ethernet receiver to support 100Mbps at TP cable channel. The proposed submodule consists of programmable gain controller, timing recovery, adaptive equalizer and baseline wander compensator. The measured Bit Error Rate is less than $10^{-12}BER$ when continuously receiving data up to 150m. The proposed signal processing submodule is implemented in digital circuits except for PLL and amplifier. The performance improvement of the proposed equalizer and BLW compensator is measured about 1dB compared with the existing architecture that removes BLW using errors of an adaptive equalizer. The architecture has been modeled using Verilog-HDL and synthesized using samsung $0.18{\mu}m$ cell library. The implemented digital signal processing submodule operates at 142.7 MHz and the total number of gates are about 128,528.

An Efficient FTN Decoding Method using Separation of LDPC Decoding Symbol in Next Generation Satellite Broadcasting System (차세대 위성 방송 시스템에서 LDPC 복호 신호 분리를 통한 효율적인 FTN 복호 방법)

  • Sung, Hahyun;Jung, Jiwon
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.16 no.2
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    • pp.63-70
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    • 2016
  • To increase throughput efficiency and improve performance, FTN(Faster Than Nyquist) method and LDPC(Low Density Parity Code) codes are employed in DVB-S3 system. In this paper, we proposed efficient turbo equalization model to minimize inter symbol interference induced by FTN transmission. This paper introduces two conventional scheme employing SIC(Successive Interference Cancellation) and BCJR equalizer. Then, we proposed new scheme to resolve problems in this two conventional scheme. To make performance improved in turbo equalization model, the outputs of LDPC and BCJR equalizer are iteratively exchange probabilistic information. In fed LDPC outputs as extrinsic informa tion of BCJR equalizer. we split LDPC output to separate bit probabilities. We compare performance of proposed scheme to that of conventional methods through using simulation in AWGN(Additive White Gaussian Noise) channel. We confirmed that performance was improved compared to conventional methods as increasing throughput parameters of FTN.

A Study of Efficient Viterbi Equalizer in FTN Channel (FTN 채널에서의 효율적인 비터비 등화기 연구)

  • Kim, Tae-Hun;Lee, In-Ki;Jung, Ji-Won
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.6
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    • pp.1323-1329
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    • 2014
  • In this paper, we analyzed efficient decoding scheme with FTN (Faster than Nyquist) method that is transmission method faster than Nyquist theory and increase the throughput. we proposed viterbi equalizer model to minimize ISI (Inter-Symbol Interference) when FTN signal is transmitted. the proposed model utilized interference as branch information. In this paper, to decode FTN singal, we used turbo equalization algorithms that iteratively exchange probabilistic information between soft Viterbi equalizer (BCJR method) and LDPC decoder. By changing the trellis diagram in order to maximize Euclidean distance, we confirmed that performance was improved compared to conventional methods as increasing throughput of FTN signal.

FC-MMA Adaptive Equalization Algorithm to improve the Convergence Speed of MMA in 16-QAM System (16-QAM 시스템에서 MMA의 수렴 속도를 개선시킨 FC-MMA 적응 등화 알고리즘)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.14 no.1
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    • pp.93-99
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    • 2014
  • This paper deals with the FC-MMA (Fast Convergence-Multi Modulus Algorithm) which is improving the convergence characteristics of the MMA (Multiple Modulus Algorithm) adaptive equalization algorithm that is used for the minimization of the intersymbol interference which occurs in the time dispersive communication channel. In the time varying charateristics and the abnormal situation like as outage of the communication channel, the adaptive equalizer needs to adapting the new environment more rapidly. For this problem, the residual isi and the maximum distortion performance index which are meaning the convergence characteristics are widely adapted in the adaptive equalizer. The 16-QAM signal is transmitted and it was confirmed that the proposed algorithm, the FC-MMA has the fast convergence performance such as in the 1.75 times fast in residual isi and 2.5 times fast in the maximum distortion in order to reaching the steady state compare to the MMA algorithm in the same channel environment by the computer simulation.

A Performance Analysis of AM-SCS-MMA Adaptive Equalization Algorithm based on the Minimum Disturbance Technique (Minimum Disturbance 기법을 적용한 AM-SCS-MMA 적응 등화 알고리즘의 성능 해석)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.16 no.3
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    • pp.81-87
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    • 2016
  • This paper analysis the AM-SCS-MMA (Adaptive Modulus-Soft Constraint Satisfaction-MMA) based on the adaptive modulus and minimus-disturbance technique in order to improve the stability and robustness in low signal to noise power of current MMA adaptive equalization algorithm. In AM-SCS-MMA, it updates the filter coefficient applying the adaptive modulus and minimum-disturbance technique of deterministic optimization problem instead of LMS or gradient descend algorithm for obtain the minimize the cost function of adaptive equalization. It is possible to improve the equalizer filter stability, robustness to the various noise characteristic and simultaneous reducing the intersymbol interference due to the amplitude and phase distortion occurred at channel. The computer simulation were performed for confirming the improved performance of SCS-MMA. For these, the output signal constellation of equalizer, residual isi, MSE, EMSE (Excess MSE) which means the channel traking capability and SER which means the robustness were applied. As a result of computer simulation, the AM-SCS-MMA have slow convergence time and less residual quantities after steady state, more good robustness in the poor signal to noise ratio, but poor in channel tracking capabilities was confirmed than MMA.