• 제목/요약/키워드: Adaptive Signal Processing

검색결과 479건 처리시간 0.029초

VLSI Implementation for the MPDSAP Adaptive Filter

  • Choi, Hun;Kim, Young-Min;Ha, Hong-Gon
    • 융합신호처리학회논문지
    • /
    • 제11권3호
    • /
    • pp.238-243
    • /
    • 2010
  • A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

Precoder Distribution and Adaptive Codebook in Wideband Precoding

  • Long, Hang;Kim, Kyeong Jin;Xiang, Wei;Wang, Jing;Liu, Yuanan;Wang, Wenbo
    • ETRI Journal
    • /
    • 제34권5호
    • /
    • pp.655-665
    • /
    • 2012
  • Based on wideband precoding (WBP) in the multiple-input multiple-output orthogonal frequency division multiplexing system, an adaptive nonuniform codebook is presented in this paper. The relationship between the precoder distribution and spatial correlation is analyzed at first. A closed-form expression based on overlapped isosceles triangles is proposed as an approximation of the precoder distribution. Then, the adaptive codebook design is derived with the approximate distribution to minimize quantization errors. The capacity and bit error rate performance demonstrate that the adaptive codebook with WBP outperforms the conventional fixed uniform codebook.

능동 신호 처리 및 시간 주파수 해석을 이용한 기어의 이상 진단 (Fault Diagnosis in Gear Using Adaptive Signal Processing and Time-Frequency Analysis)

  • 이상권
    • 소음진동
    • /
    • 제8권4호
    • /
    • pp.749-756
    • /
    • 1998
  • 기어에서 충격성 진동 및 소음은 치차의 이상과 연관이 있다. 따라서 충격 진동 및 소리는 기어의 이상 진단에 사용되어 질 수 있다. 또한 이들 충격파를 조기에 정확하게 탐지하여 기어의 이상을 진단하면 완전 파손을 방지할 수 있다. 그러나 주변 소음 및 노이즈 신호 때문에 객관적이 충격파의 탐지가 어렵기 때문에, 본 논문은 이러한 숨겨진 충격 신호를 능동 신호 처리 기법을 이용하여 조기에 찾아내고 이것을 시간-주파수 영역에서 해석하였다.

  • PDF

디지탈 신호처리에 의한 실시간 태아 심전도 감시 시스템 (Real-time FECG monitoring system using digital signal processing)

  • 김남현;김원기;윤대희;박상희
    • 제어로봇시스템학회:학술대회논문집
    • /
    • 제어로봇시스템학회 1990년도 한국자동제어학술회의논문집(국내학술편); KOEX, Seoul; 26-27 Oct. 1990
    • /
    • pp.580-585
    • /
    • 1990
  • This paper presents a real time FECG signal monitoring system in which an adaptive multichannel noise canceller is implemented using a Texas Instruments TMS32020 digital signal processor. Abdominal ECG signal is applied as the desired output and 3 chest ECG signals as the reference input signals of the adaptive multichannel noise canceller whose coefficients are updated using the LMS algorithms.

  • PDF

고해상도 LCD TV를 위한 적응형 콘트라스트 향상 장치의 설계 및 구현 (The Design and Implementation of Adaptive Contrast Enhancement Device for High Resolution LCD TV)

  • 서범석;권병헌;황병원
    • 한국항행학회논문지
    • /
    • 제11권3호
    • /
    • pp.319-328
    • /
    • 2007
  • 본 논문에서는 고해상도 FPD(Flat Panel Display)에서 동영상의 화질 향상을 위한 콘트라스트 향상 알고리즘을 제안하고 이를 구현하였다. 또한 입력되는 영상신호의 밝기 분포에서 영상의 평균과 분산을 이용하여 적응적으로 처리하는 적응형 콘트라스트 제어 기법을 제안하였다. 설계된 Contrast Enhancer는 기존화면과 비교하여 정량적으로 측정하였으며, 실제 유용성을 검증하기 위해 30인치 TFT-LCD TV에 인터페이스하여 구현하였다.

  • PDF

Linearly Constrained Adaptive Array Processing with Alternate Mainbeam Nulling

  • Chang, Byong-Kun;Jeon, Chang-Dae;Song, Dong-Hyuk
    • Journal of electromagnetic engineering and science
    • /
    • 제8권2호
    • /
    • pp.52-58
    • /
    • 2008
  • This paper concerns with signal cancellation problem in a linearly constrained adaptive array processor in coherent environment. Alternate mainbeam nulling approach was proposed to prevent the signal cancellation phenomenon. The linearly constrained LMS algorithm with a unit gain constraint and that with a null constraint in the direction of the desired signal is alternately implemented to reduce the signal interaction between the desired signal and the interferences, which is the main cause of the signal cancellation. It is shown that the proposed method performs better than a conventional method.

개선된 시스템 제어기를 사용한 능동소음제어의 실시간 구현 특성 (Characteristics of Real-time Implementation using the Advanced System Controller in ANC Systems)

  • 문학룡;손진근
    • 전기학회논문지P
    • /
    • 제64권4호
    • /
    • pp.267-272
    • /
    • 2015
  • Active noise control (ANC) is a method of cancelling a noise signal in an acoustic cavity by generating an appropriate anti-noise signal via canceling loudspeakers. The continuous progress of ANC involves the development of improved adaptive signal processing algorithms, transducers, and DSP hardware. In this paper, the convergence behavior and the stability of the FxLMS algorithm in ANC systems with real-time implementation is proposed. Specially, The advanced DSP H/W with dual core(DSP+ARM) and API(application programming interface) S/W programming was developed to improve the real-time implementation performance under the FxLMS algorithms of input noise such as road noise environment. The experimental results are found to be in good agreement with the theoretical predictions.

TMS320C30을 이용한 단일채널 적응잡음제거기 구현 (Implementation of the single channel adaptive noise canceller using TMS320C30)

  • 정성윤;우세정;손창희;배건성
    • 음성과학
    • /
    • 제8권2호
    • /
    • pp.73-81
    • /
    • 2001
  • In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

  • PDF

웨이브렛 패킷 변환을 이용한 적응알고리듬의 수렴속도 향상 (Enhancement of Convergence Speed of Adaptive Algorithm using Wavelet Packet Transform)

  • 박서용;김대성
    • 정보학연구
    • /
    • 제2권2호
    • /
    • pp.127-138
    • /
    • 1999
  • 최근 들어 신호처리 분야에서 웨이브렛 변환을 이용한 연구가 활발히 진행되고 있다. 본 논문에서는 웨이브렛 변환 영역에서의 적응 알고리듬을 구현하고 비 정제적 신호에 대한 성능을 평가하였다. 입력 신호를 웨이브렛 패킷 변환하여 다해상도 분해하고 NLMS알고리듬을 이용하여 부 밴드에서의 적응 알고리듬을 구현하였다. 제안한 방법을 화이트 가우시안 잡음이 섞인 도플러 신호의 잡음 제거에 적용하여 그 성능을 평가하였다.

  • PDF