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Background noise suppression for signal enhancement by novelty filtering

IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS 2000年第1期36卷 102-113页

作者： Ko, H Arozullah, M Korea Univ Sch Elect Engn Sungbuk Ku Seoul 136701 South Korea Catholic Univ Amer Dept Elect Engn Washington DC 20065 USA

The enhancement of weak signals in the presence of background and channel noise is necessary to design a robust automatic signal detection and recognition system. The autoassociative property of neural networks can be used to map the identifying characteristics of input source waveforms or their spectra. This paper is directed at the exploitation of such neural network properties for novelty filtering that improves the detection probability of weak signals hy learning and subsequent subtraction of noise background from the input waveform. A neural-network-based preprocessor that learns to selectively filter out the background noise without significantly affecting the signal will be highly useful in solving practical signal enhancement problems. An analytical basis is established for the operation of neural-network-based novelty filters that enhance the signal detectability in the presence of noise background and channel noise.

关键词： backpropagation feedforward neural nets filtering theory matched filters signal detection tracking filters Haar wavelet LMS algorithm autoassociative property automatic signal recognition system background noise suppression channel noise channel noise matched filters feedforward neural nets back propagation BACKGROUND NOISE tracking filters least mean square algorithm signal acquisition filtering theory signal enhancement

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Low complexity implementation of LMS algorithm

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ELECTRONICS LETTERS 2002年第15期38卷 836-837页

作者： Fletcher, P Dean, M QinetiQ Ltd Malvern WR14 3PS Worcs England

Results are presented for the least mean squares (LMS) algorithm for array antenna control. The implementation is of reduced complexity, which offers significant advantages for adoption in ultra-low-cost systems.

关键词： Adoption Complexity Antenna arrays least mean square least mean square algorithm Costing systems

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ADAPTIVE ZERO-TRACKING COMPLEMENTED BY LMS WEIGHT UPDATE FOR GUARANTEED CONVERGENCE

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IEEE TRANSACTIONS ON ANTENNAS AND PROPAGATION 1992年第3期40卷 345-349页

作者： KO, CC BACHL, R Department of Electrical Engineering National University of Singapore Singapore

A low-complexity zero-tracking algorithm with fast and guaranteed convergence behavior is proposed and investigated. The algorithm implements a zero-tracking algorithm and a least mean square (LMS) weight update algorithm in parallel, with the former adjusting the zeros of the array in a time-multiplexed manner to achieve fast asymptotic and tracking behavior, while the latter improves the initial transient behavior of the algorithm and guarantees its convergence to the global optimum. By comparing their output powers, the relative performance of the two component algorithms are monitored, and re-initialization of one algorithm by the other may occur periodically. This renders the new algorithm to have both fast and guaranteed convergence behavior, even though the zeros are directly available and the implementation complexity is only about two times that for the conventional LMS algorithm.

关键词： Converge Zero reinitialization algorithms Fasting global optimum least mean square algorithm TWO-COMPONENT Transient performance

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Two-stage ML-based group detection for direct-sequence CDMA systems

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JOURNAL OF COMMUNICATIONS AND NETWORKS 2003年第1期5卷 33-42页

作者： Buzzi, S Lops, M Univ Cassino DAEIMI I-03043 Cassino FR Italy

In this paper a two-stage maximum-likelihood (NIL) detection structure for group detection in DS/CDMA systems is presented. The first stage of the receiver is a linear filter, aimed at suppressing the effect of the unwanted (i.e., out-of-group) users' signals, while the second stage is a non-linear block, implementing a NIL detection rule on the set of desired users signals. As to the linear stage, we consider both the decorrelating and the minimum mean square error approaches. Interestingly, the proposed detection structure turns out to be a generalization of Varanasi's group detector, to which it reduces when the system is synchronous, the signatures are linearly independent and the first stage of the receiver is a decorrelator. The issue of blind adaptive receiver implementation is also considered, and implementations of the proposed receiver based on the LMS algorithm, the RLS algorithm and subspace-tracking algorithms are presented. These adaptive receivers do not rely on any knowledge on the out-of-group users' signals, and are thus particularly suited for rejection of out-of-cell interference in the base station. Simulation results confirm that-the proposed structure achieves very satisfactory performance in comparison with previously derived receivers, as well as that the proposed blind adaptive algorithms achieve satisfactory performance.

关键词： Linear independence Direct-sequence code-division multiple access Computer Programming Languages adaptive receiver linear filters Receivers mean square error least mean square algorithm proposed structure

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Variable Step-size LMS algorithm Based on Hyperbolic Tangent Function

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CIRCUITS SYSTEMS AND SIGNAL PROCESSING 2023年第7期42卷 4415-4431页

作者： Li, Long Zhao, Xuesong Harbin Univ Sci & Technol Sch Elect & Elect Engn Harbin 150080 Peoples R China

To solve the problem that the least mean square (LMS) algorithm cannot balance the convergence rate and steady-state mean square error (MSE) well and require excessive manual adjustment of parameters, this paper proposes a parameter-free variable step-size LMS algorithm based on the hyperbolic tangent function. First, the algorithm uses the hyperbolic tangent function to establish a nonlinear relationship between the error and the step size. Based on this, the average value of the error signal is used to update the step size. Further, the accumulated error value is multiplied by its average value and added to the input signal energy. Then, the algorithm is normalized by using this value to prevent the algorithm divergence due to the sudden increase in the signal power. The convergence proof of the proposed algorithm is provided in this paper. The simulation results indicate that the proposed algorithm can automatically adjust the parameters. The convergence rate and the steady-state MSE are better than those of other algorithms for both high and low signal-to-noise ratios.

关键词： least mean square algorithm Hyperbolic tangent function System identification mean square error

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Shunt active power filtering with reference current generation based on dual second order generalized integrator and LMS algorithm

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ENERGY REPORTS 2022年 8卷 886-893页

作者： Sant, Amit, V Pandit Deendayal Energy Univ Sch Technol Dept Elect Engn Gandhinagar India

For reducing the carbon emissions from utility sector, it is important to improve the efficiency of network and electrical loads. Current harmonics are responsible for the degradation of the system as well as load efficiency. The propagation of current harmonics into the grid can be prevented with shunt active power filters (SAPFs). Reference current generation (RCG) is critical for SAPF operation. This paper presents the control of SAPF with RCG based on least mean square (LMS) and dual second order generalized integrator (SOGI). The two SOGIs separately process grid voltage and load current to determine the respective fundamental in-phase component (FPC) and fundamental quadrature component (FQC). FPC and FQC of grid voltage are used to determine the phase angle of fundamental grid voltage. With LMS algorithm the distorted current can be decomposed into fundamental active (FA), fundamental reactive (FR) and harmonic components. However, the dynamic response of LMS based estimation is poor when the same learning rate is used for all components. FQC of load current, estimated with SOGI, can be processed by LMS algorithm to separate FA and FR components. RCG can be implemented with the computed phase angle and FA component. In LMS algorithm, as only two components are to be separated, two different learning rates can be easily specified. This results in faster dynamic response for LMS algorithm and has consequent positive impact on RCG and SAPF operation. Also, the computational complexity is considerably reduced. The performance of SAPF with the proposed extraction algorithm is analyzed for different operating conditions and comparative analysis is carried out. (C) 2022 The Author. Published by Elsevier Ltd.

关键词： Current harmonics least mean square algorithm Second order generalized integrator Shunt active power filter

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Improvements in intelligibility of noisy reverberant speech using a binaural subband adaptive noise-cancellation processing scheme

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2001年第6期110卷 3232-3242页

作者： Shields, PW Campbell, DR Univ Paisley Sch Informat & Commun Technol Elect Engn & Phys Div Paisley PA1 2BE Renfrew Scotland

This article reports on the performance of an adaptive subband noise cancellation scheme, which performs binaural preprocessing of speech signals for a hearing-aid application. The multi-microphone subband adaptive (MMSBA) signal processing scheme uses the least mean squares (LMS) algorithm in frequency-limited subbands. The use of subbands enables a diverse processing mechanism to be employed, splitting the two-channel wide-band signal into smaller frequency-limited subbands, which can be processed according to their individual signal characteristics. The frequency delimiting used a linear- or cochlear-spaced subband distribution. The effect of the processing scheme on speech intelligibility was assessed in a trial involving 15 hearing-impaired volunteers with moderate sensorineural hearing loss. The acoustic material consisted of speech and speech-shaped noise signals, generated using simulated and real-room acoustic environments, at signal-to-noise ratios (SNRs) in the range -6 to +3 dB. The results show that the MMSBA scheme delivered average speech intelligibility improvements of 11.5%, with a maximum of 37.25%, in noisy reverberant conditions. There was no significant reduction in mean speech intelligibility due to processing, in any of the test conditions. (C) 2001 Acoustical Society of America.

关键词： broadband signal Scheme Noise subband Subbands intelligibility Speech Intelligibility Speech noise signal least mean square algorithm Signaling In Band Self tuning

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Channel estimation and tracking for MC-CDMA signals

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EUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS 2004年第3期15卷 249-258页

作者： Sanguinetti, L Morelli, M Mengali, U Univ Pisa Dept Informat Engn I-56100 Pisa Italy

This paper investigates the impact of channel estimation errors in the uplink and downlink of a multicarrier code-division multiple-access (MC-CDMA) system. Pilot blocks are periodically inserted into the downlink data stream to perform a least-squares channel estimation. It is shown that a single-user minimum-mean-square-error (MMSE) detector endowed with the proposed channel estimator is robust to estimation errors even under full-loaded conditions. In the uplink scenario channel estimates are computed through a least-mean-square algorithm and are passed to a linear multi-user MMSE detector or a parallel interference cancellation (PIC) receiver. Simulations indicate that in this case the system performance depends heavily on the quality of the channel estimates, meaning that they are particularly critical. Copyright (C) 2004 AEI.

关键词： Multicarrier code division multiple access System performance up link Error analysis channel estimation tracking Mini Mental State Examination Downlink least mean square algorithm

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A FINITE PRECISION ANALYSIS OF THE BLOCK-GRADIENT ADAPTIVE DATA-DRIVEN ECHO CANCELER

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IEEE TRANSACTIONS ON COMMUNICATIONS 1992年第5期40卷 940-946页

作者： CIOFFI, JM HO, MN Department of Electrical Engineering Information Laboratory University of Stanford Stanford CA USA Dept. of Electr. Eng. Stanford Univ. CA USA

In this paper, we analyze the block-gradient (block LMS) algorithm's finite-precision performance in the data-driven echo canceller application. From both our analysis and our simulation results, we find that the block LMS requires significantly less precision than the standard LMS algorithm. Our analysis also shows how higher convergence and tracking speeds may be traded for an improvement in precision requirements. We derive formulae that can be used to accurately compute echo rejection levels as a function of precision, signal powers, step-size, block length, and echo canceller length. Finally, we illustrate the utility of the formulae by showing performance levels for a typical V.32bis full-duplex voiceband modem operating at a transmission rate of 14.4 kbps (7200 Hz sampling rate), and for a high-speed digital subscriber line echo canceller operating at a sampling rate of 400 KHz.

关键词： Echo cancellers data driven TRANSMISSION RATE formula least mean square algorithm sampling rate Cancellers Precision High-rate Digital Subscriber lines

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Broadband control of plate radiation using a piezoelectric, double-amplifier active-skin and structural acoustic sensing

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JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA 2000年第2期107卷 876-884页

作者： Johnson, BD Fuller, CR Virginia Polytech Inst & State Univ Vibrat & Acoust Labs Dept Mech Engn Blacksburg VA 24061 USA

The potential of a piezoelectric, double-amplifier active-skin with structural acoustic sensing (SAS) is demonstrated for the reduction of broadband acoustic radiation from a clamped, aluminum plate. The active-skin is a continuous covering of the vibrating portions of the plate with active, independently controllable piezoelectric, double-amplifier elements and is designed to affect control by altering the continuous structural radiation impedance rather than structural vibration. In simulation, acoustic models are sought for the primary and secondary sources that incorporate finite element methods. Simulation indicates that a total radiated power attenuation in excess of 10 dB may be achieved between 250 and 750 Hz with microphone error sensing, while under SAS the radiated power is reduced by nearly 8 dB in the same frequency range. In experiment, the adaptive feed forward filtered-x LMS (least mean square) algorithm, implemented on a Texas Instruments C40 DSP, was used in conjunction with the 6I6O control system. With microphone error sensing, 11.8-dB attenuation was achieved in the overall radiated power between 175 and 600 Hz. while inclusion of SAS resulted in a 7.3-dB overall power reduction in this frequency band. (C) 2000 Acoustical Society of America. [S0001-4966(00)00702-5].

关键词： broadband Aluminum sheet Microphone Device Component Frequency bands structural vibration frequency ranges Synthetic aperture sonar Acoustic imagery power decay least mean square algorithm

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