• Title/Summary/Keyword: Least mean square (LMS)

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Design of a Blind DFE Equalizer for high-speed data communication (고속 데이터 통신을 위한 Blind DFE Equalizer의 설계)

  • 박원흠;선우명훈
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.7C
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    • pp.704-711
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    • 2002
  • This paper proposes a DFE (Decision Feedback Equalizer) equalizer ASIC using the Multi-Modulus Algorithm (MMA) for cable modem applications. We believe that it is the first effort to combine the DFE structure and the MMA algorithm. The proposed equalizer has been designed for 64/256 QAM modems. The existing MMA equalizer uses two transversal filters and updates two tap weights while the proposed equalizer uses two DFE filter banks to improve the channel adaptive performance and to reduce the number of taps and updates only one tap weights. We have used the 0.35 $\mu\textrm{m}$ standard cell library. The implemented equalizer ASIC operates at 8 MHz and provides 64 Mbps which is higher than existing equalizers. The total number of gates are about 160,000.

Voice Activity Detection Algorithm base on Radial Basis Function Networks with Dual Threshold (Radial Basis Function Networks를 이용한 이중 임계값 방식의 음성구간 검출기)

  • Kim Hong lk;Park Sung Kwon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.12C
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    • pp.1660-1668
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    • 2004
  • This paper proposes a Voice Activity Detection (VAD) algorithm based on Radial Basis Function (RBF) network using dual threshold. The k-means clustering and Least Mean Square (LMS) algorithm are used to upade the RBF network to the underlying speech condition. The inputs for RBF are the three parameters in a Code Exited Linear Prediction (CELP) coder, which works stably under various background noise levels. Dual hangover threshold applies in BRF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental result show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

Interference Signal Control using Linear BISP Algorithm (선형 BISP 알고리즘을 이용한 간섭 신호 제어)

  • Na, Ha-Sun;Kim, Moon-Hwan;Suk, Kyung-Hyu;Song, Sun-Hee;Park, Dong-Suk;Joung, U-Sun;Bae, Chul-Soo;Na, Sang-Dong
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • v.9 no.1
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    • pp.630-634
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    • 2005
  • 본 논문은 신경망을 이용한 간섭 신호 제어 복합 다층 퍼셉트론에서 DS/SS 이동 통신에서 수신된 신호를 검출하는 것에 대하여 연구한다. 수신 신호가 일정한 비트율을 갖는 채널에 전송하기 위하여 신경망을 이용한 새로운 탭 가중치 갱신 제어 방법을 제안한다. 적응 횡단선 필터는 상호 심볼간 간섭을 억압하기 위해 LMS 알고리즘 사용하고, 응답과 실제 출력간의 차인 에러를 이용하여 탭 가중치 조절 메카니즘을 통해 탭 가중치를 갱신함으로서 효과적으로 간섭을 제거한다. 본 논문은 상호 심볼간 간섭을 효율적으로 억압해온 다계층 퍼셉트론 조합을 이용하여 제안된 알고리즘을 통해 탭 가중치 갱신이 보다 효율적으로 이루어질 수 있도록 한다. 시뮬레이션을 통해 평균자승 에러의 수렴 특성이 우월하다는 것을 연구한다.

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A Design of Adaptive Equalizer for Terrestrial Digital Television Receivers (지상파 디지털 TV 수신기의 적응등화기 설계)

  • 정진희;김정진;권용식;장용덕;정해주
    • Journal of Broadcast Engineering
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    • v.8 no.2
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    • pp.153-162
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    • 2003
  • This paper describes a structure of adaptive equalizer to improve reception performance of ATSC digital television (DTV) for 8-VSB receivers. There are many strong and dynamic echoes affecting reliable reception of DTV signal. Conventional DFE based least mean square (LMS) algorithm is readily implemented and has good Performance. There are still problems to be solved, however, in handling strong echoes and indoor reception. In this paper, structure of adaptive equalizer to mitigate these Problems in strong multipath interference conditions and indoor reception environment is first presented. Methods to reduce error propagation effects on DFE and initialization scheme of filter coefficients for fast convergence are then introduced. Computer simulation results prove that an adaptive equalizer with proposed design methods can combat with Brazil Ensemble and the Threshold of Visibility(TOV) is improved.

Channel Fading Effect Analysis on Diffusion Cooperation Strategies over Adaptive Networks

  • Yang, Jie;Mostafapour, Ehsan;Aminfar, Amir;Wang, Jie;Huang, Hao;Akhbari, Afsaneh;Ghobadi, Changiz;Gui, Guan
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.13 no.1
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    • pp.172-185
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    • 2019
  • In this paper, we investigate the performance of the diffusion adaptation strategies for parameter estimation in wireless adaptive networks, where the nodes exchange information over noisy and fading wireless channels. This paper shows the differences between the effect of Rayleigh and Rician fading over wireless adaptive networks and proves that the Rician fading is a more practical model in such kinds of networks. Simulation results imply that the effect of Rayleigh fading is more degrading for the estimation process than Rician fading. Also, the simulation results show the performance of adapt then combine (ATC) diffusion algorithm is better than the combine then adapt (CTA) algorithm by merely considering noise in wireless channels. While the performance of CTA prevails ATC over the wireless adaptive network in the presence of noise plus channel fading.

Performance Analysis of Own Ship Noise Cancellation in Hull Mounted Sonar System Using Adaptive Filter (HMS시스템에서 적응필터를 이용한 자함의 소음감소 성능분석)

  • Yoon, Kyung-Sik;Jung, Tae-Jin;Lee, Kyun-Kyung
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.1
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    • pp.10-17
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    • 2010
  • In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.

Subband Affine Projection Algorithm Using Variable Step Size (가변 스텝사이즈를 이용한 부밴드 인접투사 알고리즘)

  • Choi, Hun;Bae, Hyeon-Deok
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.2
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    • pp.69-74
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    • 2007
  • In signal processing applications with highly correlated input signals, subband affine projection algorithm and step size controlling is a good solution for improving the slow convergence rate and large computational complexity of LMS-type algorithms. This paper proposes a subband affine projection algorithm using a variable step size. The proposed method achieves fast convergence rate and small steady-state error with a small computational complexity by combining the SAP and step size controlling in a subband structure. Experimental results on highly correlated input signal show that the proposed method is superior to the conventional methods.

A Design of Pipelined Adaptive Decision-Feedback Equalized using Delayed LMS and Redundant Binary Complex Filter Structure (Delayed LMS와 Redundant Binary 복소수 필터구조를 이용한 파이프라인 적응 결정귀환 등화기 설계)

  • An, Byung-Gyu;Lee, Jong-Nam;Shin, Kyung-Wook
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.37 no.12
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    • pp.60-69
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    • 2000
  • This paper describes a single-chip full-custom implementation of pipelined adaptive decision-feedback equalizer(PADFE) using a 0.25-${\mu}m$ CMOS technology for wide-band wireless digital communication systems. To enhance the throughput rate of ADFE, two pipeline stages are inserted into the critical path of the ADFE by using delayed least-mean-square(DLMS) algorithm. Redundant binary (RB) arithmetic is applied to all the data processing of the PADFE including filter taps and coefficient update blocks. When compared with conventional methods based on two's complement arithmetic, the proposed approach reduces arithmetic complexity, as well as results in a very simple complex-valued filter structure, thus suitable for VLSI implementation. The design parameters including pipeline stage, filter tap, coefficient and internal bit-width, and equalization performance such as bit error rate (BER) and convergence speed are analyzed by algorithm-level simulation using COSSAP. The single-chip PADFE contains about 205,000 transistors on an area of about $1.96\times1.35-mm^2$. Simulation results show that it can safely operate with 200-MHz clock frequency at 2.5-V supply, and its estimated power dissipation is about 890-mW. Test results show that the fabricated chip works functionally well.

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Localization Using Extended Kalman Filter based on Chirp Spread Spectrum Ranging (확장 Kalman 필터를 적용한 첩 신호 대역확산 거리 측정 기반의 위치추정시스템)

  • Bae, Byoung-Chul;Nam, Yoon-Seok
    • Journal of the Institute of Electronics Engineers of Korea SC
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    • v.49 no.4
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    • pp.45-54
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    • 2012
  • Location-based services with GPS positioning technology as a key technology, but recognizing the current location through satellite communication is not possible in an indoor location-aware technology, low-power short-range communication is primarily made of the study. Especially, as Chirp Spread Spectrum(CSS) based location-aware approach for low-power physical layer IEEE802.15.4a is selected as a standard, Ranging distance estimation techniques and data transfer speed enhancements have been more developed. It is known that the distance measured by CSS ranging has quite a lot of noise as well as its bias. However, the noise problem can be adjusted by modeling the non-zero mean noise value by a scaling factor which corresponds to the change of magnitude of a measured distance vector. In this paper, we propose a localization system using the CSS signal to measure distance for a mobile node taken a measurement of the exact coordinates. By applying the extended kalman filter and least mean squares method, the localization system is faster, more stable. Finally, we evaluate the reliability and accuracy of the proposed algorithm's performance by the experiment for the realization of localization system.

Adaptive Lattice Step-Size Algorithm for Narrowband Interference Suppression in DS/CDMA Systems

  • Benjangkaprasert, Chawalit;Teerasakworakun, Sirirat;Jorphochaudom, Sarinporn;Janchitrapongvej, Kanok
    • 제어로봇시스템학회:학술대회논문집
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    • 2003.10a
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    • pp.2087-2089
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    • 2003
  • The presence of narrowband interference (NBI) in Direct-sequence code division multiple access (DS/CDMA) systems is an inevitable problem when the interference is strong enough. The improvement in the system performance employs by adaptive narrowband interference suppression techniques. Basically there have been two types of method for narrowband interference suppression estimator/subtracter approaches and transform domain approaches. In this paper the focus is on the type of estimator/subtracter approaches. However, the binary direct sequence (DS) signal, that acts as noise in the prediction process is highly non-Gaussian. The case of a Gaussian interferer with known in an autoregressive (AR) signal or a digital signal and also in a sinusoidal signal (Tone) that included in is paper. The proposed NBI suppression is presence in an adaptive IIR notch filter for lattice structure and more powerful by using a variable step-size algorithm. The simulation results show that the proposed algorithm can significantly increase the convergence rate and improved system performance when compare with adaptive least mean square algorithm (LMS).

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