A Phase Adaptive Structure for Echo Cancellation

Echo cancellation as an effective technique for echo control on telephone circuits is burdened by the appearance of carrier frequency offsets and phase jitter in some echo paths. In this paper we present a digital phase-adaptive structure for cancellation which will automatically compensate for echo path frequency offsets as large as 0.001 of the sampling rate without any reduction in return/loss enhancement.     

自适应回声消除算法的仿真研究

介绍几种自适应回声消除算法:NLMS,PNLMS,PNLMSDU,IPNLMS,MPNLMS,并在Matlab环境下进行了仿真比较.采用高斯噪声作为输入,分别在回声路径稀疏、回声路径中等稀疏、回声路径完全非稀疏的条件下进行仿真.仿真结果表明,PNLMS算法是针对稀疏回声路径的有效快速算法,MPNLMS算法做了进一步的优化.IPNLMS算法和PNLMSDU算法融合了NLMS算法和PNLMS算法的优点,既保证有较快的初始收敛速度,又保证后阶段收敛速度不明显下降.     

基于盲信号分离的自适应回声抵消算法

在视频会议和免提通信系统中,扬声器和麦克风之间的声耦合严重影响语音通信系统的质量。文中提出了一种用基于盲信号分离（BSS）的自适应声回波对消（AEC）方法,可有效解决回声和噪声对近端语音信号的影响。该方法不仅能减少背景噪声对回波对消的影响,而在双向通话时,可直接利用盲信号分离技术分离出近端语音。     

自适应回音抵消技术分析与探讨

对回音抵消器中的关键技术--自适应滤波技术及双方同时说话/回音路径跳变检测与控制技术进行了介绍、分析与比较,探讨了现有技术的优缺点,并提出了该技术领域进一步的研究方向.     

A Single-Channel Acoustic Echo Cancellation Scheme Using Gradient-Based Adaptive Filtering

In this paper, a single-channel acoustic echo cancellation (AEC) scheme is proposed using a gradient-based adaptive least mean squares (LMS) algorithm. Unlike the conventional dual-channel problem, by considering a delayed version of the echo-suppressed signal as a reference, a modified objective function is formulated and thereby an LMS update equation is derived. It is shown that the resulting update equation converges to the optimum Wiener–Hopf solution. Based on the commonly used assumption of negligibility of cross correlation between the reference and the current speech signals, a multi-step stopping criterion is introduced, which not only provides efficient control of the LMS update sequence but also ensures a faster convergence. The proposed control criterion is validated by considering all possible scenarios which arise due to the variation of speech properties at the reference and current samples of the adaptive filter. From extensive experimentation on several real-life echo-corrupted speech signals in different acoustic environments, it is found that the proposed algorithm can efficiently handle the problem of single-channel AEC and provide satisfactory performance in terms of both subjective and objective measures.     

改进子带自适应滤波算法及其在回波抵消中的应用

本文提出了两种基于多带结构的仿射投影符号子带自适应滤波器(Affine Projection Sign Subband Adaptive Filter, AP SSAF)的改进方法。针对稀疏系统的系统识别,设计了两种子带自适应滤波器。首先给出了AP SSAF的变正则化参数更新方程,文中采用随机梯度下降法来更新正则化参数,来使系统的均方偏差最小化,该方法能同时兼顾快速收敛及低稳态失调。其次将权重分布矩阵引入AP SSAF得到系数比例AP SSAF,该方法能够利用系统的稀疏性提高AP SSAF的收敛性能。仿真中将本文所提算法用于一般系统识别以及回波抵消,实验结果验证了本文的算法对脉冲噪声具有稳健性,具有较好的跟踪性能,并具有较快的收敛速度及低稳态失调。      

基于LMS算法自适应回波抵消器的Simulink仿真分析

讨论了回波抵消的原理,并利用Simulink平台建立了基于LMS算法自适应回波消除器的仿真模型,对系统进行了仿真演示,给出了较为详细的模型构建和仿真波形,并对显示结果进行了性能分析。仿真的结果证实了该系统的有效性,较好地反映了系统的动态工作过程,且失真较小。     

带有旋转因子的多路回波消除自适应滤波算法

本文提出一种带有旋转因子的归一化多路回波消除算法NLMS-RF,并对算法进行了分析和几何解释.进一步地,我们将所提算法扩展成一种带有旋转因子的多路回波消除仿射算法APA-RF.NLMS-RF算法的计算量远小于Sankaran(1999)所提的多路回波消除算法,而APA-RF的计算量与Benesty(1996)所提出的算法的计算量相当.在收敛精度和速度上,NLMS-RF和APA-RF分别较Sankaran(1999) 和Benesty(1996)所提算法有较大的改善.仿真结果表明所提算法的有效性.     

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。     

一种改进的自适应滤波算法及其在回波消除中的应用

该文改进了一种自适应滤波算法,并对新算法进行了收敛性分析。作为应用,将新算法应用于混沌调制通信系统在回波信道中的回波消除问题。仿真结果表明,改进后的算法比原算法具有更好的收敛速度和收敛精度;在低信噪比下,新算法更为稳定。     

基于LMS的自适应直放站回波消除算法研究

针对无线同频直放站转发器输出的大功率射频信号可使整个系统会自激这一问题,提出了基于LMS的回波消除系统算法.如果不用回波消除,会造成增益不可增大以及系统容易自激,若整个系统自激,可造成系统不稳定,进而可能损坏末级放大器.采用回波抑制技术后,可以有效改善信号的回波对系统的影响,提高系统的稳定性.仿真结果表明该算法可有效地抑制回波.     

Adaptive Echo Cancellation Based on a Multipath Model of Acoustic Channel

Usually Acoustic Echo Cancellers (AECs) are realized by adaptive Finite duration Impulse Response (FIR) filter having large number of coefficients and Least Mean Square (LMS) as an adaptive algorithm resulting in slow convergence speed and poor tracking performance of these adaptive filters. In this paper, we have proposed a Multiple Sub-filter (MSF) parallel structure based on multipath acoustic echo model using the basis that each sub-filter will compensate the echo contributed by each path of multipath acoustic channel. To realize the MSF, modified Generalized Autocorrelation-based Estimator (MAE) has been used to estimate time delay associated with each path while the order of each sub-filter has been estimated using Power Spectral Density (PSD) method. Accuracy Percentage (AP) performance measure has been used to characterize the performance of the estimator. Simulation results show that the performance of the MAE improves with the increase in SNR and/or decrease in number of multipath. Using these estimates MSF based AEC is constructed. The convergence performance of MSF-based AEC has been studied, via computer simulation, and compared with the conventional Single Long length adaptive Filter (SLF)-based canceller for different SNRs and number of multipath. The results of MSF have been found to be very encouraging in almost all of the various situations considered. Subsequently, the tracking behavior has also been studied with variation in the channel parameters of the multipath model. The proposed MSF can track variations in the channel parameters of the multipath model faster as compared to the conventional echo canceller.     

A Modified Adaptive FIR Equalizer for Multipath Echo Cancellation in FM Transmission

Multipath propagation is a severe problem in conventional FM broadcasting since this phenomenon causes nonlinear distortions after demodulation. Several proposals for the correction of multipath corrupted FM signals were given in some recent papers, but up to now there exist no satisfactory solutions with sufficiently low hardware complexity for commercial application. The main idea of the present paper is to take advantage of some a priori knowledge of the multipath transfer function with the goal to develop a nonrecursive equalizer structure with minimum hardware expense. For adaptive adjustment of the equalizer, we use a recently introduced constant modulus algorithm (CMA) in a modified form. Instead of direct coefficient adjustment, we regard the parameters of the transmission channel as unknowns which are updated adaptively. The equalizer coefficients are uniquely defined by the channel parameters and can be determined in one step. The advantage of this method is a large improvement of the convergence behavior in comparison to existing solutions. The methods presented here are tested by means of an FM hardware system including FM transmitter, multipath simulation, FM demodulator, and an FIR multipath equalizer, which allows real-time investigations of the equalizer performance under well-defined multipath conditions.     

基于LMS算法的自适应直放站回波消除算法研究

针对无线同频直放站转发器输出的大功率射频信号可使整个系统会自激这一问题,提出了基于LMS的回波消除系统算法。如果不用回波消除,会造成增益不可增大以及系统容易自激,若整个系统自激,可造成系统不稳定,进而可能损坏末级放大器。采用回波抑制技术后,可以有效改善信号的回波对系统的影响,提高系统的稳定性。仿真结果表明该算法可有效地抑制回波。     

基于DSP Builder的回波抵消器设计与实现

针对通信中的回波问题,基于自适应滤波的LMS算法,设计了自适应回波抵消器。并基于利用FPGA芯片,在DSP Builder平台上,有效结合MatLab／Simulink和Quartus II设计工具,根据模块化设计思想实现了LMS算法自适应回波抵消器硬件电路设计。软件仿真和系统FPGA硬件实测结果表明,该设计方法使回波抵消器的FPGA硬件实现更加简便快捷。     

卫星通信中的回波抑制与回波抵消

本文叙述了卫星通信中回波的产生及其抑制的原理，分析了ＪＵＥ－４５Ａ Ｍ Ⅱ ＩＮＭＡＲＳＡＴ－Ａ船站中回波抑制器电路，阐述了一种用横向滤波器实现的回波抵消的原理。     

电话语音回声消除的研究

为了消除电话语音系统中的反射回声,本文通过对电话信道的研究,提出通过估算信道冲击响应,得到提示语音经过信道卷积后的结果,从而抑制回声的算法.与LMS自适应算法相比,它可获得很好的回声抑制和对原始信号的保真效果.回声抵消后,语音识别正确率和准确率分别提高约29%和60%.     

Adaptive Combination of Proportionate NSAF with the Tap-Weights Feedback for Acoustic Echo Cancellation

To obtain fast convergence rate and low steady-state error in acoustic echo cancellation, a convex combination scheme of the improved proportionate normalized subband adaptive filter algorithm is proposed. Instead of the gradient method in the conventional combination theory, the mixing parameter is adapted by using the normalized gradient method which makes it more robust to the variations of subband error signals. Also, to implement a smooth transition from the fast filter to the accurate filter, a cyclic feedback of the overall tap-weights giving to all component filters is applied.     

一种新自适应滤波快速算法及其 在多路回波消除中的应用

本文提出了一种新的自适应滤波算法,该算法结构简单、计算量适中且收敛速度快,弥补了一般变步长LMS自适应算法计算量小但收敛速度欠佳,以及仿射投影算法(APA)收敛速度快但计算量非常大的缺陷.该算法计算量与一般LMS算法相当,而收敛速度却与APA算法相当,其结构比APA及相应的改进算法要简单得多.我们不仅对所提算法的收敛性及性能进行了分析,而且将它用于多路回波消除中获得了成功,仿真结果表明,该算法与Sankaran(1997)所提NLMS-OCF算法及Benesty(1996)所提APA-MC算法比较,在收敛速度和收敛精度相当的情况下,其计算复杂度大大减少.从而新算法具备更好的实时性.     

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…