Fast algorithm for avoiding incidental cancellation problems in partially adaptive arrays

The problem of signal cancellation in partially adaptive arrays can be avoided by using a fast source separation algorithm to track individual sources in the environment and adding back the desired signal component to the system output if the latter is detected to have been cancelled.<>     

多径回波信道条件下回波抵消技术研究

主要研究移动通信数字同频直放站的回拨抵消技术。通过对基于最小均方误差原理的自适应滤波器的收敛条件进行分析,发现即使在外信道为单径信道或只考虑多径信道的主径回波信道时,现有利用时延去相关的自适应滤波算法,在原理上存在着不能保证满足收敛条件的问题。针对这一问题,本文提出了将发送信号进行延时后再送入自适应估计器的改进措施,有条件地改善了回波抵消效果,并进一步提出了消除多径回波的新型技术原理。     

An adaptive algorithm for separating and tracking multipledirectional sources in linear arrays

An algorithm for spatially filtering out, enhancing, and tracking individual directional sources in an adaptive array is proposed and investigated. In this algorithm, the sources are separated by using an adaptive beamformer whose outputs are processed by using the LMS algorithm to track distinct sources individually. From the LMS weights used, the source locations can be estimated. Whenever significant changes in these are detected, the beamformer is updated so that its outputs will be due to different sources in the steady state. With this algorithm, the problems of look-direction errors in look-direction constrained arrays and of large signal power in power inversion arrays are eliminated, and the enhancement of multiple moving sources becomes a natural process. Furthermore, because the sources are individually tracked and the beamformer is only updated occasionally, the algorithm possesses fast tracking behavior, and its implementation complexity is comparable to that of beamformer-based adaptive arrays using the LMS algorithm     

峭度自适应学习率的盲信源分离

本文提出了一种自适应学习率盲信源分离的自然梯度算法,自适应学习率仅依赖于神经网络输出峭度平方和的负指数.开始阶段由于小的峭度,学习率大收敛速度快.之后,随着峭度变大,学习率慢慢变小,产生小的稳态误差.在线性无记忆混合的情况下,用欠高斯信源进行的模拟实验表明,与固定学习率相比,本文提出的峭度自适应学习率盲信源分离算法具有收敛速度快和稳态误差小的特点.     

一种条件自适应滤波算法及其在回声消除中的应用

对回声消除问题进行分析后,提出了一种条件自适应滤波算法。新算法并不要求自适应滤波器达到最优值,而是在满足一定条件时才进行系数更新。这种算法可以用在回声消除领域,很好地解决信道变化和双端发声的区分问题。文章还对这种条件自适应滤波算法的收敛性能做了严格分析,并且给出了和分析结果相吻合的计算机仿真结果。     

基于小波变换的声学回波消除系统

声学回波消除技术一直是语音通信领域的研究热点。在声学回波消除系统中,通过估计回波路径中的固定时延区域来提高自适应滤波算法的收敛速度。提出了一种基于小波变换的固定时延估计算法以及基于小波变换的声学回波消除系统,解决传统时延估计算法在声学回波消除系统中估计误差大、抗干扰能力弱的问题。测试结果表明,算法稳健性、有效性等指标明显优于传统时延估计算法,基于小波变换的声学回波消除系统具有良好的消回波性能。     

基于离散余弦变换的旁瓣对消技术研究

自适应旁瓣对消是一种有效抑制有源干扰的措施。研究了自适应旁瓣对消和合成孔径雷达（SAR）有源遮盖式干扰的基本原理,详细推导了基于离散余弦变换的DCT-LMS频域自适应方法,并将其应用于SAR的旁瓣对消系统中。通过与其他自适应算法的对比实验,证明了DCT-LMS算法兼有收敛速度快,计算量小的优点。最终模拟实际环境中的干扰源,利用SAR的实际数据进行了仿真实验。实验结果表明,DCT-LMS算法能有效地抑制有源干扰噪声,确保SAR接收机正常工作,具有较高的干扰对消比。     

基于组合滤波算法IP电话的回声消除

网络之间互连的协议（IP,Internet Protocol）电话回声消除问题一直都是研究的热点,通常使用自适应滤波算法来消除回声,但其收敛速度和稳态失调之间的矛盾是回声消除需要解决的一个重要问题。研究一种组合比例自适应滤波算法,按照一定的比例组合两种具有一定互补性能的算法,能够有效解决收敛速度和稳态失调。通过MATLAB仿真分析,证明了组合比例自适应滤波算法具有更快的收敛速度和良好的稳态特性。     

频域快速自适应干扰对消算法研究及仿真

针对于提高干扰机收发隔离度的自适应干扰对消系统 ,利用快速FFT技术实现了一种替代时域LMS算法的频域快速LMS自适应算法 ,分析表明该算法不但具有同时域LMS算法近似的收敛特性 ,而且计算量大幅度减少 ,有利于对消系统的实时实现。计算机仿真证实了分析的正确性和该算法的可行性     

集成多种自适应滤波算法的回声消除器

如何选择自适应算法的步长,从而有效解决收敛速度和稳态失调之间的矛盾是回声消除中的一个重要问题。论文提出一种集成多种自适应滤波算法的回声消除框架,以挖掘不同自适应滤波算法以及不同步长选择之间的互补性,来获得稳定的消除效果。所提算法可以分析同一时刻不同算法的误差,并始终选择一种最好的算法。通过对LMS、NLMS、PNLMS和IPNLMS这四种自适应算法的结合实验,显示了该算法可以集合各种算法以及步长选择的优点,具有更快的收敛速度和良好的稳态特性。     

The behavior of a multiple channel active control system

The convergence behavior of an adaptive feedforward active control system is studied. This adjusts the outputs of a number of secondary sources to minimize a cost function comprising a combination of the sum of mean-square signals from a number of error sensors (the control error) and the sum of the mean-square signals fed to the secondary sources (the control effect). A steepest descent algorithm which performs this function is derived and analyzed. It is shown that some modes not only converge slowly but also require an excessive control effort for complete convergence. This ill-conditioned behavior can be controlled by the proper choice of the cost function minimized. Laboratory experiments using a 16-loudspeaker 32-microphone control system to control the harmonic sound in an enclosure are presented. The behavior of the practical system is accurately predicted from the theoretical analysis of the adaptive algorithm. The effect of errors in the assumed transfer matrix used by the steepest descent algorithm is briefly discussed     

An approach to analog mitigation of RFI

Among all noise sources present in wireline transmission systems we focus on one special type: narrowband radio frequency interference generated by radio amateurs (RAM) and broadcast radio stations. This disturbance, characterized by high power and narrow bandwidth, has the potential of overloading the receiver's analog-to-digital converter (ADC). Once the ADC is in saturation, any countermeasure taken in digital domain will fail. A viable way to face this problem is cancellation using the common-mode signal as a reference. This paper describes in detail an adaptive, mixed-signal, narrowband interference canceller employing a modified recursive least-squares algorithm, which is split into an analog and a digital part. The mixed-signal approach enables the circuit to generate an interference-cancelling signal of several MHz while operating the adaptive algorithm at some kilohertz. Simulation as well as measurement results show a steady-state disturbance suppression of about 35 dB. The convergence speed is high enough to protect the ADC from overloading due to time-variant HAM interference     

基于子带分解和比例归一化算法的回声消除器设计与仿真

针对稀疏信道条件下的网络回声抵消问题。提出了一种比例归一化子带自适应滤波算法。该算法基于子带分解结构,并利用网络中回声路径的稀疏特性,使得各个系数的步长与该系数的绝对值成比例,加快了活动系数的收敛速度,从而改善了子带分解算法在稀疏信道条件下的性能。仿真结果表明：将所提算法应用于网络回声消除器,能够获得很快的收敛速度和很低的稳态失调。     

Estimation-based synthesis of H∞-optimal adaptive FIR filtersfor filtered-LMS problems

This paper presents a systematic synthesis procedure for H∞-optimal adaptive FIR filters in the context of an active noise cancellation (ANC) problem. An estimation interpretation of the adaptive control problem is introduced first. Based on this interpretation, an H∞ estimation problem is formulated, and its finite horizon prediction (filtering) solution is discussed. The solution minimizes the maximum energy gain from the disturbances to the predicted (filtered) estimation error and serves as the adaptation criterion for the weight vector in the adaptive FIR filter. We refer to this adaptation scheme as estimation-based adaptive filtering (EBAF). We show that the steady-state gain vector in the EBAF algorithm approaches that of the classical (normalized) filtered-X LMS algorithm. The error terms, however, are shown to be different. Thus, these classical algorithms can be considered to be approximations of our algorithm. We examine the performance of the proposed EBAF algorithm (both experimentally and in simulation) in an active noise cancellation problem of a one-dimensional (1-D) acoustic duct for both narrowband and broadband cases. Comparisons to the results from a conventional filtered-LMS (FxLMS) algorithm show faster convergence without compromising steady-state performance and/or robustness of the algorithm to feedback contamination of the reference signal     

Interference cancellation in respiratory sounds via amultiresolution joint time-delay and signal-estimation scheme

This paper is concerned with the problem of cancellation of heart sounds from the acquired respiratory sounds using a new joint time-delay and signal-estimation (JTDSE) procedure. Multiresolution discrete wavelet transform (DWT) is first applied to decompose the signals into several subbands. To accurately separate the heart sounds from the acquired respiratory sounds, time-delay estimation (TDE) is performed iteratively in each subband using two adaptation mechanisms that minimize the sum of squared errors between these signals. The time delay is updated using a nonlinear adaptation, namely the Levenberg-Marquardt (LM) algorithm, while the function of the other adaptive system-which uses the block fast transversal filter (BFTF)-is to minimize the mean squared error between the outputs of the delay estimator and the adaptive filter. The proposed methodology possesses a number of key benefits such as the incorporation of multiple complementary information at different subbands, robustness in presence of noise, and accuracy in TDE. The scheme is applied to several cases of simulated and actual respiratory sounds under different conditions and the results are compared with those of the standard adaptive filtering. The results showed the promise of the scheme for the TDE and subsequent interference cancellation     

基于梯度向量投影的自适应滤波算法的改进及其在多路回波消除中的应用

该文首先对Lim(2000)的基于梯度向量正交投影的算法(OGA)进行了分析和改进,在此基础上获得了一种新的自适应滤波算法(MOGA)。新算法使用时变遗忘因子对误差进行指数加权平均来估计均方误差,并使用该因子改变自适应迭代过程中滤波器系数向量的更新方向.然后将改进后的新算法扩展成两路回波消除算法用于多路回波的消除中,获得了良好的效果。仿真结果表明, MOGA不仅对时变或时不变系统具有很好的跟踪能力,克服了Lim(2000)所提算法收敛性不佳甚至有时发散的缺陷,而且应用于多路回波消除时具有计算量小,收敛速度快和精度高等特点,其收敛速度和精度优于J.Benesty(1996)和G.Sankaran(1999)的相应结果。     

Adaptive Reference Echo Cancellation

The AREC (adaptive reference echo cancellation) algorithm is presented for an echo canceler used in full-duplex two-wire digital transmission on digital subscriber loops. The AREC algorithm incorporates a decision-directed estimation of and compensation for the far-end signal which is a source of interference to the conventional echo canceler adaptation algorithm. The AREC algorithm thus offers much faster convergence and shorter coefficient wordlengths than the conventional algorithm. Analysis and simulation of the performance and convergence of both AREC and conventional echo canceler adaptation algorithms are carried out. Included in the analysis is the effect of receiver delay and coefficient wordlength requirements. A simple and robust startup procedure is proposed and investigated by simulation.     

Adaptive arrays with main beam constraints

Initial applications of adaptive array theory to the radar sidelobe jamming problem ignored the problem of incidental cancellation of the desired signal returns. In more recent applications, longer transmitted waveforms have combined with returns from extended clutter and/or strong targets to create a more serious signal cancellation problem. There are several ways in which the adaptive processor can be constrained from responding to desired main lobe target returns while maintaining good cancellation of interference in the sidelobes. This paper examines the major techniques for constraining the response of the adaptive processor, including methods of controlling the response of the array in the absence of external interference. Time domain and frequency domain techniques are discussed. The majority of the discussion is devoted to angle domain techniques such as pilot signals, preadaption spacial filtering, and control loop spatial filtering. Analysis is presented showing the relationship between these techniques. Finally, examples are given showing the effects of these constraints as well as control of the quiescent array pattern.     

An Adaptive Acoustic Echo Cancellation Algorithm Based on Subband Decomposition and Conjugate Gradient Method

1 Introduction In speech communication applications ,the presence ofcoupling fromloudspeaker to the microphone often re-sults in undesired acoustic echo that seriously degradesspeech quality.Current solutions for removingthis echoare based on the real ti me identification of the acoustici mpulse response by using adaptive filtering or AdaptiveEcho Cancellation (AEC) filter techniques . Several AEC algorithms have been proposed for thisproblem. An acoustic echo canceller based upon inputort…     

Maximizing jammer effectiveness for evaluating the performance of adaptive nulling array antennas

A theory is developed which is used to find interference source distributions which maximize consumption of the degrees of freedom forN-channel adaptive nulling arrays with arbitrary element positions. For a given number of interference sources, after proper positioning, these sources represent a maximally stressed environment for the adaptive array degrees of freedom. The interference covariance matrix eigenvalues are shown to have a direct bearing on the number of degrees of freedom consumed as well on the adaptive cancellation. Numerical examples are given showing that certain source geometries produce the situation where little or no adaptive cancellation is possible due to the available degrees of freedom being severely taxed.