A Signal Decorrelation PNLMS Algorithm for Double-Talk Acoustic Echo Cancellation

Acoustic echo cancellation (AEC) in voiced communication systems is used to eliminate the echo which corrupts the speech signal and reduces the efficiency of signal transmission. Usually, the performance of AEC system based on the adaptive filtering degrades seriously in the presence of speech issued from the near-end speaker (double-talk). In typical AEC scenarios, double-talk detector (DTD) must be added to AEC for improving speech quality. One of the main problems in AEC with DTD is that the DTD errors can result in either large residual echo or distorting the near-end input speech. Considering the strong correlation property of speech signals, this paper presents a novel proportionate decorrelation normalized least-mean-square (PDNLMS) adaptive AEC without DTD for echo cancellation as an interesting alternative to the typical AEC with DTDs. Unlike traditional AEC with a DTD, the proposed PDNLMS uses the difference of near-end speech as the residual error to update adaptive echo channel filter during the periods of double-talk, which can efficiently reduce the double-talk influence on the AEC adaptation process. The experimental results show that not only the proposed PDNLMS without DTD illustrate better stability and faster convergence rate, but it is also of a lower steady-state misalignment and better residual signal than current methods with DTDs at a lower computational cost.     

DSSS中基于重叠变换的自适应干扰抑制算法及仿真分析

提出了一种基于重叠变换的自适应干扰抑制算法，利用调制重叠变换（MLT）把接收机信号映射到变换域进行抗干扰处理，它可以有效地把干扰能量集中在有限的变换域子带（Transform Bins）中，与基于DFT、DCT等块变换相比有更好的滤波效果。由于可以使用基于重叠变换的快速算法，因此它结构简单，节省计算量。仿真结果表明，基于MLT的变换域处理方案可以有效地改善系统性能。     

基于深度学习的两阶段联合声学回波和混响抑制技术

在现代通信系统中，通信语音的质量和可懂度会被回波与混响严重损害，人与人之间的交流因此会被严重干扰。为了同时消除回波与混响的负面影响，本文提出了一种基于深度学习的两阶段联合声学回波和混响抑制系统。该系统逐步地消除加性声学回波与多径效应产生的混响干扰，从而获得目标语音。系统首先使用基于理想比值掩蔽（Ideal Ratio Mask，IRM）的模型去除与目标信号不相关的声学回波，紧接着对于与目标信号强相关的混响干扰，系统通过利用一个基于“隐掩蔽”的谱映射模型将其去除。两阶段模型最后进行联合训练以获得更好的系统性能。一系列不同声学环境下的实验结果表明，本文所提出的系统可显著地消除回波与混响干扰，从而极大地增强了目标语音的语音质量与可懂度。      

Biorthogonal and nonuniform lapped transforms for transform codingwith reduced blocking and ringing artifacts

New lapped transforms are introduced. The lapped biorthononal transform (LBT) and hierarchical lapped biorthogonal transform (HLBT) are appropriate for image coding, and the modulated HLBT biorthogonal transform (MMLBT) and nonuniform modulated lapped biorthogonal transform (NMLBT) are appropriate for audio coding. The HLBT has a significantly lower computational complexity than the lapped orthogonal transform (LOT), essentially no blocking artifacts, and fewer ringing artifacts than the commonly used discrete cosine transform (DCT). The LBT and HLBT have transform coding gains that are typically between 0.5 and 1.2 dB higher than that of the DCT. Image coding examples using JPEG and embedded zerotree coders demonstrate the better performance of the LET and HLBT. The NMLBT has fewer ringing artifacts and better reproduction of transient sounds than the MLT, as shown in audio coding examples. Fast algorithms for both the HLBT and the NMLBT are presented     

一种简化的自适应回声抵消算法及其应用分析

声学回声抵消器可视为一横向自适应滤波器,声学回声抵消技术在现代信号处理中得到越来越广泛的应用。在讨论了声学回声抵消基本原理后,提出了一种基于自适应滤波技术的声学回声抵消简化频域FDAF—MDFTP算法,并对该算法特性以及对声学回声返回损失增加度的影响进行了分析。最后,基于该频域算法对室内声学回声抵消进行了模拟计算,可得到很好的回音抵消效果。     

一个基于盲信号分离的多路回波消除结构

多路回波消除问题的本质难点是由于多路输入信号之间的强相关性而导致的自适应滤波器解不唯一的问题.目前许多文献都是试图通过对输入信号预处理来解决这个问题,这些方法都对原输入信号作了变化,从而影响了通话质量.本文提出了一个简化的盲信号分离模型的立体声回波消除结构,在这个结构中,立体声回波消除问题被转化为两个单路的回波消除问题,从而避免了自适应滤波器解不唯一的问题.仿真结果证明了新结构的有效性.     

VoIP通信中的回声消除方法研究

介绍了语音通信声学回声产生模型和自适应AEC回声消除算法原理,分析了AEC应用于VoIP语音通信中存在的问题,设计了一种基于短时能量的非线性回声消除方法,在NGN网络的VoIP通信中,使用该方法实现了极高的回声抑制比。测试结果表明该方法的消回声效果、算法稳定性和实现复杂度等指标明显优于自适应AEC算法,适合于嵌入式VoIP通信终端设备的开发。     

A Nonlinear Acoustic Echo Canceller Using Sigmoid Transform in Conjunction With RLS Algorithm

Nonlinearity of amplifiers and/or loudspeakers gives rise to nonlinear echo in acoustic systems, which seriously degrades the performance of speech and audio communications. Many nonlinear acoustic echo cancellation (AEC) methods have been proposed. In this paper, a simple yet efficient nonlinear echo cancellation scheme is presented by using an adaptable sigmoid function in conjunction with a conventional transversal adaptive filter. The new scheme uses the least mean square (LMS) algorithm to update the parameters of sigmoid function and the recursive least square (RLS) algorithm to determine the coefficient vector of the transversal filter. The proposed AEC is proved to be convergent under some mild assumptions. Computer simulations show that the proposed scheme gives a superior echo cancellation performance over the well known Volterra filter approach when the echo path suffers from the saturation-type nonlinear distortion. More importantly, the new AEC has a much lower computational complexity than the Volterra-filter-based method.      

不需双端检测的声回波对消算法研究

声回波对消中双端对讲情况下的近端话音对自适应算法有很大影响。为避免双端话音检测,在滤波型LMS算法基础上,用远端信号和误差输出信号的和代替远端信号去激励预测误差滤波器,降低近端话音的影响。另为进一步提高算法抗近端干扰的能力,做了变步长的改进,首先将步长反比于输出信号预测误差的短时功率,其次将步长正比于预测误差的互相关系数。实验表明,文中提出的两算法在近端话音出现时表现出较好的性能,其中第二种有更好的稳态失调。     

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.     

Extended lapped transforms: properties, applications, and fastalgorithms

The family of lapped orthogonal transforms is extended to include basis functions of arbitrary length. Within this new family, the extended lapped transform (ELT) is introduced, as a generalization of the previously reported modulated lapped transform (MLT). Design techniques and fast algorithms for the ELT are presented, as well as examples that demonstrate the good performance of the ELT in signal coding applications. Therefore, the ELT is a promising substitute for traditional block transforms in transform coding systems, and also a good substitute for less efficient filter banks in subband coding systems     

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。     

An approach to data-driven echo cancellation in OQAM-basedmulticarrier data transmission

This paper presents a method of echo cancellation in offset-QAM based multicarrier data transmission. The method makes use of inherent T/2 amplitude modulation signaling structure of the orthogonally frequency-division multiplexed mutually T/2-staggered quadrature subchannel signals. The echo cancellation convergence is analyzed and a particular two-stage strategy to effectively accelerate adaptation process is presented     

一种利用人耳听觉特性的双声道回波抵消改进算法

本文提出了一种利用人耳听觉特性的双声道回波抵消改进算法.在该系统中,利用人耳的听觉特性给输入信号加上一个小的扰动来改善输入信号的自相关特性和条件数,从而达到改善系统收敛特性的作用.本文讨论了在双声道回波抵消系统中遇到的输入信号强相关从而引起系统性能恶化的问题,提出了采用输入信号加扰来改善系统性能的算法.并且利用人耳听觉特性使这种扰动信号隐藏在语音信号中,使其对语音信号的干扰最小.从本文的仿真试验来看,该方法对双声道回波抵消系统的性能有一定的改善.     

A Robust Step-Size Control Technique Based on Prop ortionate Constraints on Filter Up date for Acoustic Echo Cancellation

One of the m ain di?culties in Acoustic echo cancellation (AEC) is that the filter adaptation needs to vary according to different situations such as near-end interferences and echo path changes. In this paper, we pro-pose a robust step-size control algorithm in frequency do-main. The proposed method is based on the optimization of the square of the bin-wise a posteriori error. Constraint on the filter update is applied, which contributes to ro-bustness to near-end interferences of the algorithm. The learning rate formula is derived first and then the relation-ship between the proposed algorithm and a robust statis-tics based approach is revealed. The method is extended to the Multidelay block frequency domain adaptive filter (MDF) so as to meet the demand of low delay in prac-tical application. Moreover, the values of the constraints are designed to be updated proportionately to improve the convergence property. Simulation results demonstrate the superiority of the proposed algorithm.     

Cascaded power symmetric IIR filter banks and continuityconstrained adaptive algorithms for acoustic echo cancellation insubbands

Teleconferencing systems and hands-free mobile terminals use acoustic echo cancellation (AEC) for high-quality full-duplex speech communication. The problem of aliasing in subband AEC is addressed. Filter banks with implicit notch filtering are derived from cascaded power symmetric-infinite impulse response (CFS-IIR) filters. It is shown that adaptive filters used with these filter banks must be coupled via continuity constraints to reduce the aliasing in the residual echo. A continuity constrained NLMS algorithm is therefore proposed and evaluated     

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。     

Filter bank design for a subband adaptive filtering structure with critical sampling

Subband adaptive filtering structures are attractive in applications such as acoustic echo cancellation and channel equalization, due to their properties of decorrelating the input signal and reducing the computational complexity. Recently, a new adaptive filtering structure with critical sampling was proposed. In this paper, we describe an optimization procedure to select the analysis and synthesis filter banks of this new subband structure, so that minimum steady-state mean square error or fastest convergence rate can be achieved. Such filter-bank design method is based on a theoretical analysis of the convergence properties of the adaptation algorithm and uses a nonlinear optimization routine. Computer simulations illustrate the convergence improvements that can be obtained with the filter banks designed by the proposed method.     

Combined acoustic echo cancellation,dereverberation and noise reduction: a two microphone approach

This paper presents new algorithms for acoustic echo cancellation and noise reduction which use two (or possibly more) microphone signals. In contrast to the single microphone method the multimicrophone approach can exploit the spatial coherence properties of sound fields which arise from noise and reverberated speech. Besides the standardfir echo canceller the proposed algorithms comprise an adaptive filter to eliminate non coherent signal components. The combined system achieves better Erle than thefir echo canceller alone, attenuates ambient noise, dereverberates near end speech, and possibly leads to implementations with reduced complexity. The paper analyzes the acoustical properties of typical environments, presents the algorithms and experimental results.     

一种基于二阶盲信号分离的多路声回波抑制模型

本文提出了一个基于二阶盲信号分离的多路声回波抑制模型.该模型回避了多路声回波对消中因声回波源信号间的强互相关性所致的固有的解的非唯一性问题,而是充分利用了这种互相性来去除声回波.模型仅添加一个辅助麦克风并巧妙置位即可对各路麦克风信号中的多路声回波进行有效的分离和抑制.为了实时处理的目的,提出了一个计算复杂度低且收敛稳健的二阶频域盲信号分离算法来检验该模型.实验结果充分确认了提出模型的有效性.