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14th International IEEE Conference on Intelligent Transportation Systems (ITSC)

作者： Yamamoto, Ken Ohno, Kohei Itami, Makoto Tokyo Univ Sci Dept Appl Elect 2641 Yamazaki Noda Chiba 2788510 Japan

ISBN: (纸本)9781457721977

In this paper, an adaptive circular array antenna is applied to DS/SS (Direct Sequence / Spread Spectrum) IVC (Inter-Vehicle Communication) under shadowing environment. The DS/SS scheme is an attractive technique for the IVC system because it has advantages for CDMA (Code Division Multiple Access) capability and so on. However, the near-far problem is occurred in the DS/SS scheme. Shadowing the radio paths by large vehicles is also significant problem for the IVC system. To solve those problems, a lms (Least Mean Square) adaptive circular array antenna is proposed. The array antenna can be controlled its directivity when the receiving vehicle knows the information of the target vehicle. The PER (Packet Error Rate) and the throughput performances are evaluated. And the performances are compared with the ordinary system by using omini-directional antenna. From the simulation results, the adaptive circular array antenna is effective for the near-far problem and influence of shadowing in the DS/SS IVC system.

关键词： Inter-Vehicle Communication Adaptive Circular Array Antenna Shadowing DS/SS lms algorithm

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Application of adaptive line enhancer with lms to separate heart sound signal from lung sound signal at real time

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Research Journal of Applied Sciences, Engineering and Technology 2015年第6期9卷 448-453页

作者： Sathesh, K. Muniraj, N.J.R. Department of ECE Tejaa Shakthi Institute of Technology for Women Coimbatore Tamilnadu India

This study presents a technique for separation of Heart Sound Signal (HSS) from Lung Sound Signal (LSS) using Adaptive Line Enhancer (ALE) with Least Mean Square (lms) algorithm with real time recorded sound signal. While recording lung sounds, an incessant noise source takes place owing to heart sounds. This noise source severely contaminates the breath sound signal and interferes in the analysis of lung sounds. The proposed system is applied with different filter order and the results show the error rate of the Desired Sound Signal (DSS), Signal to Noise Ratio (SNR) and execution time. © Maxwell Scientific Organization, 2015.

关键词： Error rate Execution time Heart sound signal lms algorithm Lung sound signal Signal to noise ratio

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A Zero-Attracting Mixed Norm lms For Sparse Acoustic Room System Estimation

A Zero-Attracting Mixed Norm LMS For Sparse Acoustic Room Sy...

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22nd IEEE Signal Processing and Communications Applications Conference (SIU)

作者： Eleyan, Gulden Salman, Mohammad Shukri Mevlana Univ Elekt & Elekt Muhendisligi Bolumu Selcuklu Konya Turkey

ISBN: (纸本)9781479948741

Recently, many adaptive filtering proposals that discuss the sparsity of the system have been appeared These proposals are, mainly, based on the least-mean-square (lms) algorithm. In this paper we propose two algorithms that exploit the sparsity of the system and based on the mixed norm lms (MN-lms) algorithm. The first algorithm is proposed by adding I-1-norm penalty to the cost function of the MN-lms algorithms. This term enables us to attract the near-to-zero filter coefficients into zero very fast However, when the system is highly non sparse, the algorithm almost fails. Because of that, we propose another algorithm that uses an approximation of thel(0)-norm penalty term in the cost function of the MN-lms algorithm. This provides high performance even if the system is highly non sparse system. The performances of the proposed algorithms are compared to those of the lms and MN-lms algorithms in an acoustic sparse system identification setting. The proposed algorithms provide significant performances compared to the others.

关键词： lms algorithm MN-lms algorithm Sparse System

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23-mW 50-MS/s 10-bit Pipeline A/D Converter with Nonlinear lms Foreground Calibration

23-mW 50-MS/s 10-bit Pipeline A/D Converter with Nonlinear L...

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IEEE International Symposium on Circuits and Systems (ISCAS 2009)

作者： Oshima, Takashi Takahashi, Tomomi Yamawaki, Taizo Hitachi Ltd Cent Res Lab Kokubunji Tokyo 1858601 Japan

ISBN: (纸本)9781424438273

A novel nonlinear foreground calibration of pipeline AID converters based on lms algorithm has been proposed and verified by macro-based simulation as well as measurement of test chip. The prototype 23-mW 50-MS/s 10-bit pipeline ADC in 0.13-mu m CMOS has confirmed that the proposed calibration can accurately and rapidly correct the inaccuracy caused both by heavy nonlinearity of low-gain op-amps and by their incomplete settling without requiring any dedicated voltage references, complicated structures or timing sequence and hence can be used for various applications.

关键词： Pipeline ADC foreground calibration lms algorithm

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Improving Performance of Inter-Vehicle Communication using lms Adaptive Circular Array Antenna

Improving Performance of Inter-Vehicle Communication using L...

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IEEE Intelligent Vehicles Symposium (IV)

作者： Yamamoto, Ken Ohno, Kohei Itami, Makoto Tokyo Univ Sci Dept Appl Elect Noda Chiba 2788510 Japan

ISBN: (纸本)9781424478668

In this paper, lms (Least Mean Square) adaptive array antenna is applied to IVC(Inter-Vehicle Communication) in order to reduce interference which is caused by surrounding vehicles. The lms adaptive array antenna has a characteristic that the control of directivity is possible according to surrounding radio wave condition, if the receiving vehicle knows the information of the target vehicle. Under the case where the adaptive array antenna is used in IVC systems, packet error rate characteristic, throughput performance and improvement in near-far problem are evaluated by comparing with the performance of the IVC system using omni-directional antenna.

关键词： Inter-Vehicle Communication Adaptive Array Antenna DS/CDMA lms algorithm

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On performance bounds for an affine combination of two lms adaptive filters

On performance bounds for an affine combination of two LMS a...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Bershad, N. J. Bermudez, J. C. M. Tourneret, J. -Y Univ Calif Irvine 1621 Santiago Dr Newport Beach CA 92660 USA Univ Fed Santa Catarina Florianopolis SC Brazil IRIT ENSEEIHT TeSA F-31071 Toulouse France

ISBN: (纸本)9781424414833

This paper studies the statistical behavior of an affine combination of the outputs of two lms adaptive filters that simultaneously adapt using the same white Gaussian input. The purpose of the combination is to obtain an lms adaptive filter with fast convergence and small steady-state mean-square error (MSE). The linear combination studied is a generalization of the convex combination, in which the combination factor is restricted to the interval (0, 1). The viewpoint is taken that each of the two filters produces dependent estimates of the unknown channel. Thus, there exists a sequence of optimal affine combining coefficients which minimizes the MSE. The optimal unrealizable affine combiner is studied and provides the best possible performance for this class. Then, a new scheme is proposed for practical applications. It is shown that the practical scheme yields close-to-optimal performance when properly designed (as suggested by the theoretical optimal).

关键词： adaptive filters affine combination convex combination lms algorithm stochastic algorithms

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lms adaptive filter single-user receiver

LMS adaptive filter single-user receiver

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16th International Conference on Microelectronics (ICM 2004)

作者： Marouane, H Kachouri, A Kamoun, L Natl Engineers Sch Sfax Lab Elect & Informat Technol LETI Sfax 3038 Tunisia

ISBN: (纸本)0780386566

Software radio technology makes possible the implementation Of highly flexible receivers architectures that employ sophisticated signal processing algorithms such as those required for equalization. The implementation of advanced DS-CDMA receivers based on adaptive filter principles is becoming an area of considerable interest. The focus of the paper is the introduction of a dynamically reconfigurable architecture and an adaptive linear receiver for CDMA systems in mobile communication. This paper, presents a flexible single-user receiver that can support a variety of applications and analyses the convergence behaviour of the least mean square (lms) filter when used in an adaptive code division multiple access (CDMA) detector consisting of a tapped delay line with adjustable tap weights.

关键词： adaptive FIR filters code division multi-access lms algorithm

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Comparison of lms and Nlms Adaptive Filters with a Non-Stationary Input

Comparison of LMS and NLMS Adaptive Filters with a Non-Stati...

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44th Asilomar Conference on Signals, Systems and Computers

作者： Eweda, Eweda Ajman Univ Sci & Technol Dept Elect Engn Ajman U Arab Emirates

ISBN: (纸本)9781424497218

The tracking performances of the lms and Nlms algorithms are compared when the input of the adaptive filter is nonstationary. The analysis is done in the context of tracking a Markov plant. A periodic scenario of the variation of the input power is considered. The steady-state peak mean square deviation is used as the tracking performance measure. It is found that one algorithm outperforms the other depending on the values of the rate of variation of the input power, the minimum input power, the noise variance, and the mean square plant parameter increments.

关键词： Adaptive Filtering lms algorithm Nlms algorithm Tracking

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Low-power Implementation of lms Adaptive Filters Using Scalable Rounding 27

Low-power Implementation of LMS Adaptive Filters Using Scala...

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27th IEEE International Conference on Electronics, Circuits and Systems (IEEE ICECS)

作者： Di Meo, Gennaro De Caro, Davide Napoli, Ettore Petra, Nicola Strollo, Antonio G. M. Univ Naples Federico II Dept Informat Technol & Elect Engn Naples Italy

ISBN: (纸本)9781728160443

Least Mean Square (lms) filters are the most used adaptive filters with applications ranging from channel equalization to system identification and noise cancellation. An lms adaptive filter includes two main parts: a FIR filter and a block for coefficients updating that exploits the lms algorithm. The hardware implementation of lms filter requires a significant number of multipliers, adders and registers, resulting in power consumption issues. In this paper we propose a novel approximate, low-power implementation of the coefficients update block. In the proposed approach, the signal precision is dynamically scaled by using a time-variable rounding. The circuit can select between three levels of precision: no rounding, light rounding and strong rounding. An observation block decides at runtime the rounding level, based on the magnitude of the lms error signal. In this way, it is possible to minimize the convergence error while significantly reducing the switching activity when the algorithm is close to the convergence. VLSI implementation in TSMC 28nm CMOS technology shows that proposed approach results in a maximum power saving of 27% with respect to a standard lms, with negligible degradation of error performances and limited area overhead.

关键词： Low-power digital circuits lms algorithm Approximate Computing programmable rounding

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Analyses of Kagura musical signals using lms-based Fourier analyzer

Analyses of Kagura musical signals using LMS-based Fourier a...

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IEEE Region 10 Conference (TENCON)

作者： Ishibashi, Satoru Kudoh, Norimasa Kamaya, Hiroyuki Tadokoro, Yoshiaki NIT Hachinohe Coll Elect Informat Engn Course Hachinohe Aomori Japan NIT Hachinohe Coll Dept Elect & Comp Engn Hachinohe Aomori Japan Toyohashi Univ Technol Toyohashi Aichi Japan

ISBN: (纸本)9781509025978

Musical transcription requires much burden even for specialists in the musical field therefore, many trials of automatie transcription have been being done actively. As musical signals are substantially time varying, we have investigated an lms-based Fourier analyzer to accommodate time varying characteristics. In this article, frequency response of the above adaptive algorithm is briefl reviewed in order to provide insight of the adaptive algorithm. And then, the analyzing system of Kagura Japanese folk music musical signals, which converts analyzed results to the MIDI format, is described. Finally, results in MIDI format are converted to musical scores by using commercially available software in order to verify the validity of the system.

关键词： lms algorithm musical transcription system

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