回声消除器中自适应滤波器的研究

文中介绍了一种用于IP电话的自适应回声消除器,采用NLMS算法自适应滤波器来实现.首先通过MATLAB进行了NLMS算法的步长选择,并用生成的符合G.168要求的带限CSS测试信号对回声消除器的性能进行了测试.     

VoIP回声消除器设计及算法研究

本文提出了一种与线性预测编解码器相结合的新声学回声消除器,由去相关可变步长的NLMS自适应算法和基于回声路径失配方差的双端通话检测算法所组成.Matlab仿真结果表明,与Gordy所提出的回声消除算法相比,本文提出的算法在双端通话和回声路径改变时判别更准确,收敛速度更快;在收敛状态时,ERLE值平均提高了15dB,失调误差平均降低了10dB,具备更好的回声消除性能.     

一种改进的与声码器结合的声学回声消除器

提出了一种改进的与线性预测声码器结合的声学回声消除器,通过增加一个自适应辅助滤波器,避免了对信号路径中所有信号在任何时刻都需经去相关、恢复相关过程所带来的信号质量降低、失调误差增加等负面影响。计算机仿真结果表明,该结构明显提高了已有算法的回声抵消量。     

VoIP电话中的回声消除算法

声学回声是降低VoIP通话通信质量的重要问题之一,自适应回声抵消是抑制回声的最有效方法之一,其采用自适应滤波器评估回声路径。常用的NLMS(Normalized Least Mean Square)算法计算复杂度高,实用性差,本文利用FFT技术,实现了NLMS频域快速算法FDNLMS(Frequency Domain Normalized Least Mean Square),将自适应更新变换到频域,逐块进行累加更新,保证收敛性能的同时,极大的降低了运算复杂度。实验表明,在滤波器系数为1024阶时,FDNLMS算法的处理速度比NLMS快12倍。     

基于正交理论的声学回声消除及仿真

本文介绍了一种用于消除声学回声的自适应滤波器,利用NLMS(即归一化的最小均方误差)实现算法的收敛;同时在正交理论的基础上,通过互相关计算提出了一种新的双端检测方法;最后利用Matlab对算法进行了仿真,并给出了关键部分的仿真结果。     

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。IPN-LMS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。     

IP电话回声消除器的设计及其DSP实现

介绍了一种用于IP电话中的自适应回声消除器，采用归一化最小二乘（NLMS）自适应滤波器实现，包括语音模式检测器和粗略时延估计器。最后以TI公司的TMS320C5402 DSP芯片为平台，实现了该回声消除器，学对关键代码进行了分析。     

基于信号包络和互相关的双端通话检测算法研究

声学回声消除(AEC)系统依赖于自适应滤波器的回声路径估计,当麦克风接收的信号存在回声和近端信号时,可能出现模拟回声路径的自适应滤波器发散,导致回声消除性能下降,严重时影响双端通话质量。一个成熟的声学回声消除器应该包含有双端通话检测算法,针对该问题,提出一种准确度高、性能稳定的基于信号包络检测(Env)和归一化互相关(NCC)估计相结合的双端通话检测(DTD)算法。该算法先使用远端信号与麦克风信号的能量包络来判断双端通话是否发生,当包络检测不能准确地判断双端通话时,再使用远端信号与麦克风信号的互相关估计对双端通话进行最终判断,保证双端通话检测的准确性。仿真实验表明,提出的算法可以准确判断双端通话的开始与结束,并且受回声路径变化的影响较小,提升了AEC系统的性能。     

一种新的快速自适应滤波算法的研究

从信号前后时刻相关性角度出发,提出了一种新的快速自适应滤波算法,并证明了其最优权值是一种广义的维纳方程解。该算法结构简单,收敛速度快,稳态失调小,具有处理多种信号的能力,包括非相关和强相关信号,而其计算量与NLMS算法相当。仿真结果表明,对于非相关信号,新算法的稳态失调小于VS-NLMS算法, 收敛速度与参考文献已有算法相当,但快于L.E-LMS算法;对于强相关性信号,新算法的稳态失调小于NLMS 算法和DCR-LMS算法。     

基于凸组合的同步长最大均方权值偏差自适应滤波算法

针对NLMS和PNLMS滤波器对时变信道跟踪能力差的缺点,提出了一种同步长凸组合最大均方权值偏差(MSD,mean square deviation)算法。该算法将同步长的NLMS和PNLMS 2种不同类型的自适应滤波器进行凸组合,以最大均方权值偏差为准则,使新的滤波器能够在外界信道特性(稀疏、非稀疏和模糊态)时变的情况下,保持良好的随动性能,并在收敛的各个阶段均保持快速且稳定的均方特性。理论推导和仿真实验表明:该算法与NLMS、PNLMS和IPNLMS算法相比,在稀疏和非稀疏状态时能够保持四者中最快的收敛速度,并且在模糊状态时算法性能优于其余三者。另外,该算法仍保持较好的稳态均方性能。     

基于DSP的声回声抵消器的设计

回声抵消器是视频会议系统和电话会议系统中一种必不可少的设备。在讨论回声抵消器原理和算法的基础上,给出了基于DSP的回声抵消器的设计方案。在设计中采用了变步长LMS算法,利用TI公司的DSP TMS320VC5402作为处理单元实现了回声抵消。     

Acoustic echo cancellor with sub-band noise cancelling

An acoustic echo canceller with sub-band noise cancelling that employs a cascade configuration is proposed. The adaptation control adopted to match the occurrence of intermittent speech/echo and continuous room noise using the NLMS algorithm is very effective in echo and noise cancellation. Hardware is implemented and its performance evaluated through experiments. The noise cancellation significantly enhances overall echo-cancellation performance.<>     

Gabor expansion for adaptive echo cancellation

A good echo cancellation algorithm should have a fast convergence rate, small steady-state residual echo, and less implementation cost. The normalized least mean square (NLMS) adaptive filtering algorithm may not achieve this goal. We show that using the Gabor expansion is a way to achieve this goal. For direct digital signal processing compatibility the Gabor expansion introduced in this paper is for discrete-time signals, although the Gabor expansion also can be used for continuous-time signals. The Gabor expansion can be defined as a discrete-time signal representation in the joint time-frequency domain of a weighted sum of the collection of functions (known as the synthesis functions). There are several design issues in the echo canceller based on the Gabor expansion: the design of the analysis functions for the far-end speech, the design of the analysis functions for the near-end signal containing the echo plus the near-end speech, the design of the adaptive filters in the subsignal path, and the design of the synthesis functions. All the adaptive filters are designed using identical NLMS adaptive filtering algorithms     

Performance analysis of an echo-cancellation arrangement thatcompensates for frequency offset in the far echo

Analytical and experimental results are presented for the performance of one echo canceller arrangement. It consists of a data-driven echo canceller having a so-called cross-coupled structure, which is followed by a rotator and a phase-locked loop (PLL). A cross-coupled echo canceller structure without a PLL is analyzed first. Expressions for speed of convergence and achievable echo-return-loss enhancement (ERLE) in the presence of frequency offset are derived. These results are compared in previously published results for a noncross-coupling echo canceller structure. Specifically, it is shown that the cross-coupled structure converges twice as fast as the noncross-coupled structure and provide an achievable ERLE that is about 6 dB better. The joint adaptation of the echo canceller and the PLL is then studied. It is shown that it is always possible to choose design parameters for the echo canceller which are consistent with adaptation requirements under double-talking conditions, provided that the PLL is properly engineered. The sensitivity of the performance of PLL to the power level of the far echo, as well as possible solutions to this problem, are discussed     

Acoustic echo cancellation for stereophonic systems derived from pairwise panning of monophonic speech

An algorithm is introduced that performs stereophonic acoustic echo cancellation (SAEC) for systems using pairwise panning of a single monophonic source to provide the effect of spatialisation. The technique exploits the inherent high correlation between the loudspeaker signals, unlike other general SAEC techniques, which try to utilise any small uncorrelated features in the signals. The algorithm maintains a single aggregate echo path estimate that is updated using normalised least mean square (NLMS) and the knowledge of any change in the spatialisation. Consequently, it achieves a computational complexity that is of the same order as a single channel NLMS algorithm.     

A Cascadable VLSI Echo Canceller

A new single-chip echo canceller has been developed. In addition to offering improved performance over an earlier single-chip echo canceller [14], the new canceller has many desirable features not available before. The most important of these features is that multiples of the chip can be connected in cascade to synthesize very long impulse responses. This paper describes the new features implemented on the chip and characterizes its performance.     

The fast Newton transversal filter: an efficient scheme foracoustic echo cancellation in mobile radio

The authors investigate the recently suggested fast Newton family for adaptive filtering in the context of acoustic echo cancellation, with emphasis on the mobile radio case. A distinctive advantage of the fast Newton transversal filter (FNTF) is that it can offer high performance with speech inputs at low computational cost. They discuss possible implementations and compare the FNTF with classical schemes in terms of complexity. A complete numerically stabilized version is presented, and additional features for proper real-time operation with speech are discussed. Experimental comparisons using various signals and real situations show that in all cases, the FNTF behaves similarly to the standard fast RLS transversal filter (FTF) algorithm, whereas its complexity is only slightly higher than that of the normalized LMS (NLMS). Compared with the NLMS, the experiments show that in the context investigated, the latter exhibits inferior performance with respect to convergence and tracking. Thus, they demonstrate that the FNTF is an efficient scheme for acoustic echo cancellation in mobile radio     

A data-driven multitone echo canceller

A data-driven echo canceller for full-duplex data transmission with multitone modulation is presented. This multitone echo canceller (MTEC) is not impaired by eigenvalue-spread problems that are inherent in signal-driven echo cancellers-it has numerical performance and cancellation range that equals, and in most cases exceeds, that of data-driven echo cancellers used in data transmission with baseband or quadrature amplitude modulation. The method makes use of frequency-domain updating, but time-domain implementation of the canceller. It introduces no delay into the received signal path and presents no special difficulties for interframe interpolation between near-end echo, far-end echo, or far-end data signal. The authors also investigate the fast initialization and the special case of far-end frequency offset in the echo signal     

同频数字直放站回波干扰消除器的设计

针对无线同频直放站应用中急需解决的发送天线到接收天线之间的回波干扰问题,提出一种基于叠加PN序列的回波消除方案,简要介绍了基于FPGA的回波消除器的硬件设计,以及对回波消除直放站系统回波消除度测量的测试方案,并详细论述测试原理.测试结果表明,该回波消除直放站满足实际产品化要求.