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检索条件"主题词=NLMS algorithm"

共 108 条记录，以下是81-90 订阅

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Application of adaptive filtering for weak impulsive signal recovery for bearings local damage detection in complex mining mechanical systems working under condition of varying load

Application of adaptive filtering for weak impulsive signal ...

引用

Conference on Mechatronic Systems, Mechanics and Materials

作者： Zimroz, Radoslaw Bartelmus, Walter Wroclaw Univ Technol Diagnost & Vibro Acoust Lab PL-50370 Wroclaw Poland

ISBN: (纸本)9783037852743

The paper shows application of an adaptive filter as a pre-processor for impulsive cyclic weak signal recovery from raw vibration signals captured from complex mechanical systems used in the industry (namely bearings used in pulleys parts of driving units for belt conveyors). Periodic/cyclic impulses are related to local faults which cause impulse/concentric forces/stresses in kinematic pairs. Typical examples of such local faults which cause mechanical system condition change are spall/pitting on bearings elements: outer/inner races and/or rolling elements. For analyzed objects, impulses associated with local faults are masked by other signal sources. In the first part of the paper objects are presented for the better understanding of mechanical phenomena that exist in the system, then preliminary signal analysis will be performed (in time, frequency and time-frequency domain) for the identification of signal nature. Next the idea of an adaptive system and the brief description of Normalized Least Mean Square (nlms) algorithm will be presented. Application of nlms is better than classical LMS due to stability of the adaptation. In the last section the results of adaptive filtering for signals from bearings is discussed. The authors show the application of nlms (for the first time in literature) for the case when signals are received from machines working in industrial condition. There were made only trails when the machines were investigated in laboratory conditions.

关键词： rolling element bearings local damage detection vibration adaptive filter nlms algorithm weak signal extraction

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Blind Speech Enhancement and Acoustic Noise Reduction by SFTF Adaptive algorithm 5

Blind Speech Enhancement and Acoustic Noise Reduction by SFT...

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5th International Conference on Electrical Engineering - Boumerdes (ICEE-B)

作者： Rahima, Henni Djebari, Mustapha Mohamed, Djendi Univ Blida 1 Detect Informat & Commun DIC Lab Blida 09000 Algeria Univ Blida 1 Lab Signal Proc & Imaging LATSI Route SoumaaBP 270 Blida 09000 Algeria

ISBN: (纸本)9781538606865

In this paper, we address the problem of noise reduction and speech enhancement by adaptive filtering algorithms using the forward blind source separation structure (FBSS), which is often combined with adaptive algorithms to efficiently cancel the acoustic noise at the output. In this paper, we propose to combine the FBSS with the Simplified Fast Transversal Filter (SFTF) algorithm, where the adaptation gain is obtained only from the forward prediction. The performances of the proposed SFTF algorithm are compared with the Normalized Least Mean Square (nlms) algorithm in different noisy conditions. This comparison is evaluated in terms of Cepstral Distance (CD), the System Mismatch (SM) and the Segmental Signal to Noise Ratio (SegSNR) criteria.

关键词： Speech enhancement SFTF algorithm FBSS structure nlms algorithm

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Automatic Tuning of Probe Noise for Continuous Acoustic Feedback Cancelation in Hearing Aids 24

Automatic Tuning of Probe Noise for Continuous Acoustic Feed...

引用

24th European Signal Processing Conference (EUSIPCO)

作者： Akhtar, Muhammad Tahir Nishihara, Akinori COMSATS Inst Informat Technol Dept Elect Engn Pk Rd Islamabad 44000 Pakistan Tokyo Inst Technol Dept Human Syst Sci Grad Sch Decis Sci & Technol Meguro Ku Tokyo Japan

ISBN: (纸本)9780992862657

In this paper, we propose a probe signal-based method for continuous acoustic adaptive feedback cancelation (AFC) in digital hearing aids. The main idea is to incorporate a time varying gain with the probe signal, such that a high level probe noise is injected during the transient state, and a low level probe signal is used after the system has converged. The proposed method is essentially based on two adaptive filters working in tandem. The weights between these two adaptive filters are exchanged by an efficient weight-transfer strategy, such that both adaptive filters give a good estimate of the acoustic feedback path. Simulation results demonstrate that the proposed method achieves good modeling accuracy, preserves good speech quality, and maintains high output SNR at the steady-state.

关键词： Hearing aids acoustic feedback nlms algorithm probe noise

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A Channel-Estimate-Based Decision Feedback Equalizer Robust under Impulsive Noise

A Channel-Estimate-Based Decision Feedback Equalizer Robust ...

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Underwater Communications networking (UComms)

作者： Pelekanakis, Konstantinos Chitre, Mandar Natl Univ Singapore Acoust Res Lab Singapore 119223 Singapore

ISBN: (纸本)9781479975785

A channel-estimate-based decision feedback equalizer (CEB DFE) robust under impulsive noise is presented for single-input multiple-output (SIMO) underwater acoustic communications. Channel estimation is performed via the improvedproportionate M-estimate affine projection algorithm (IPMAPA), a linear complexity algorithm which is robust against impulsive interference and exploits channel sparseness. The superiority of IPMAPA to the normalized sign algorithm (NSA) and the normalized least-mean-square (nlms) algorithm is demonstrated by processing data transmitted at 9 kbps over a 1.2km shallow water environment contaminated by snapping shrimp noise.

关键词： channel estimation decision feedback equalisers impulse noise least mean squares methods underwater acoustic communication wireless channels CEB DFE IPMAPA nlms algorithm NSA SIMO underwater acoustic communication bit rate 9 kbit/s channel estimation-based decision feedback equalizer data processing distance 1.2 km improved proportionate M-estimate affine projection algorithm impulsive interference impulsive noise linear complexity algorithm normalized least mean square algorithm normalized sign algorithm shallow water environment shrimp noise snapping single input multiple output underwater acoustic communication Baseband Channel estimation Decision feedback equalizers Feedforward neural networks Noise Receivers Robustness decision feedback equalisers Impulse noise channel estimation impulsive interference Least squares method underwater acoustic communication Baseband feedforward neural nets wireless channels Least mean square algorithm Receivers Noise shallow-water environment Robustness

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Active noise control algorithm robust to noisy inputs and measurement impulsive noises 20

Active noise control algorithm robust to noisy inputs and me...

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20th International Conference on Control, Automation and Systems (ICCAS)

作者： Park, Taesu Kim, Minsu Gwak, Minseon Cho, Taesung Park, PooGyeon Pohang Univ Sci & Technol Dept Elect Engn Pohang 37673 South Korea

ISBN: (数字)9788993215205

ISBN: (纸本)9788993215199

This paper proposes an algorithm that can effectively remove noise when Gaussian noise is introduced into a reference microphone and impulsive noise is introduced into a target point in an active-noise-cancellation (ANC) system. We applied the normalized-least-mean-square (nlms) algorithm, the most used in the adaptive filter algorithm, to the ANC environment. In the ANC environment, a compensation vector was calculated to compensate for the bias that occurs when Gaussian noise flows into the nlms algorithm. In addition, a step-size scaler was proposed to prevent false update of the adaptive filter when impulsive noise occurred at the target point. Simulation results show that the proposed algorithm has better performance in the ANC environment than other algorithms.

关键词： Active noise cancellation Adaptive filter nlms algorithm Impulsive noise

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Hardware Implementation of Adaptive Filters

Hardware Implementation of Adaptive Filters

引用

IEEE Student Conference on Research and Development

作者： Mollaei, Yaghoub Monash Univ Sch Elect & Comp Syst Engn Dept Engn Bandar Sunway Malaysia

ISBN: (纸本)9781424451876

Adaptive filter is one of the most important areas in digital signal processing. This paper seeks to use this area to remove noise from noise corrupted audio signals. An adaptive FIR filter with normalized LMS algorithm is designed to cancel the noise. A SIMULINK model is created and linked to TI TMS320C6711 digital signal processor through embedded target for TI C6000 SIMULINK toolbox and REAL-TIME workshop to perform hardware adaptive noise cancellation. The result is a noise free audio signal in the DSP board output.

关键词： Adaptive Filter Digital Signal Processor FIR Filter nlms algorithm SIMULINK TI TMS320C6711 REAL-TIME Workshop Code Composer Studio (CCS)

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FEATURE NORMALIZED LMS algorithmS 53

FEATURE NORMALIZED LMS ALGORITHMS

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53rd Asilomar Conference on Signals, Systems, and Computers (ACSSC)

作者： Yazdanpanah, Hamed Apolinario, Jose A., Jr. Univ Sao Paulo Dept Comp Sci Sao Paulo SP Brazil Mil Inst Engn IME Program Elect Engn Rio De Janeiro Brazil Mil Inst Engn IME Program Def Engn Rio De Janeiro Brazil

ISBN: (纸本)9781728143002

Recently, a great effort has been observed to exploit sparsity in physical parameters;however, in many cases, sparsity is hidden in relations between parameters, and some appropriate tools should be utilized to expose it. In this paper, a family of algorithms called feature normalized least-mean-square (F-nlms) algorithms is proposed to exploit hidden sparsity. The major key of these algorithms is a so-called feature matrix that transforms non-sparse systems to new systems containing sparsity;then the revealed sparsity is exploited by some sparsity-promoting penalty function. Numerical results demonstrate that the F-nlms algorithms can reduce the steady-state mean-squared-error (MSE) and/or improve the convergence rate of the learning process significantly.

关键词： nlms algorithm lowpass system highpass system hidden sparsity l(1)-norm

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Adaptive Subband Filtering Method for MEMS Accelerometer Noise Reduction

Adaptive Subband Filtering Method for MEMS Accelerometer Noi...

引用

Nanotechnology Conference and Trade Show (Nanotech 2008)

作者： Pietrzak, P. Pekoslawski, B. Makowski, M. Napieralski, A. Tech Univ Lodz Dept Microelect & Comp Sci PL-90924 Lodz Poland

ISBN: (纸本)9781420085051

Silicon microaccelerometers can be considered as an alternative to high-priced piezoelectric sensors. Unfortunately, relatively high noise floor of commercially available MEMS (Micro-Electro-Mechanical Systems) sensors limits the possibility of their usage in condition monitoring systems of rotating machines. The solution of this problem is the method of signal filtering described in the paper. It is based on adaptive subband filtering employing Adaptive Line Enhancer. For filter weights adaptation, two novel algorithms have been developed. They are based on the nlms algorithm. Both of them significantly simplify its software and hardware implementation and accelerate the adaptation process. The paper also presents the software (Matlab) and hardware (FPGA) implementation of the proposed noise filter. In addition, the results of the performed tests are reported. They confirm high efficiency of the solution.

关键词： MEMS accelerometer adaptive subband filtering nlms algorithm vibration measurement hardware filter

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Adaptive prediction of time-varying channels for coded OFDM systems

Adaptive prediction of time-varying channels for coded OFDM ...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Schafhuber, D Matz, G Hlawatsch, F Vienna Univ Technol Inst Commun & Radio Frequency Engn A-1040 Vienna Austria

ISBN: (纸本)0780374029

We propose adaptive channel predictors for orthogonal frequency division multiplexing (OFDM) communications over time-varying channels. Successful application of the normalized least-mean-square (nlms) and recursive least-squares (RLS) algorithms is demonstrated. We also consider the use of adaptive channel predictors for delay-free equalization, thereby avoiding the need for regular transmission of pilot symbols. Simulation results demonstrate the good performance of the proposed techniques.

关键词： adaptive estimation time-varying channels OFDM modulation prediction theory least mean squares methods recursive estimation adaptive equalisers adaptive prediction time-varying channels coded OFDM systems adaptive channel predictors orthogonal frequency division multiplexing OFDM communications normalized least-mean-square algorithm nlms algorithm recursive least-squares algorithm RLS algorithm delay-free equalization performance

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TWO-ADAPTIVE FILTER-BASED METHOD USING GAIN CONTROLLED PROBE NOISE FOR ACOUSTIC FEEDBACK NEUTRALIZATION IN DIGITAL HEARING AIDS 16

TWO-ADAPTIVE FILTER-BASED METHOD USING GAIN CONTROLLED PROBE...

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16th International Workshop on Acoustic Signal Enhancement (IWAENC)

作者： Akhtar, Muhammad Tahir Nishihara, Akinori Nazarbayev Univ Dept Elect & Comp Engn Sch Engn Kabanbay Batyr Ave 53 Astana Kazakhstan Tokyo Inst Technol Meguro Ku Tokyo 1528552 Japan

ISBN: (纸本)9781538681510

In digital hearing aids, the acoustic feedback occurs due to a close proximity of the input microphone and loudspeaker. The acoustic feedback may severely degrade the performance of hearing aids. This paper focusses on adaptive filtering approach for acoustic feedback cancelation (AFC). The proposed method comprises two adaptive filters which are adapted using a delay-based normalized least mean square (nlms) algorithm. In order to improve the convergence, we introduce (recently proposed) maximum Versoria-criterion (MVC)-based gradient information into the update algorithm. Computer simulation results are presented which demonstrate effectiveness of the proposed method.

关键词： Hearing aids acoustic feedback cancelation nlms algorithm probe noise maximum Versoria-criterion

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