基于归一化互相关法的声学回声消除及仿真

全双工免提通信系统中,要获得好的声学回声消除效果,提高语音质量,关键要解决双端发声问题,双端发声检测的准确性直接影响声学回声消除效果。由于基于能量和基于互相关双端发声检测算法存在门限值设置难,以及检测统计量对回声信道变化敏感的问题。对归一化互相关法进行了研究,得出此算法理论上不存在以上问题,通过采集真实语音信号,计算机仿真,从实验证明了此方法确实可行,并具有非常好的声学回声消除效果。     

采用变步长算法的无双端检测回声系统

提出了一种新型无双端检测的自适应回声消除系统,在与传统的回声消除系统比较过程中,本系统表现出了良好的性能,不但结构简单,计算量也小.模拟双端通信实验表明,该自适应滤波器工作正常,回声消除性能好,减少了采用传统的回声消除算法中双端检测错误造成语音切音现象和回声消除不干净的状况.     

一种使用双滤波器的回声消除算法

首先讨论了回声消除中的双端发声问题，指出传统的双端发声检测方法用于回声消除时，其不可避免的误检会导致回声消除系统性能严重恶化，使系统不稳定。本文提出一种使用双滤波器的回声消除算法。该算法使用自适应滤波器跟踪回声信道，使用一个辅助滤波嚣和自适应滤波器一起完成回声消除。实验结果表明．谊算法与传统算法相比，运算量相当;但在双端发声期间，该算法稳定性更高，回音往返损耗增强（Echo return loss enhancement，ERLE）有明显的改善。     

对消处理技术在宽带频谱感知中的应用研究

在高斯噪声背景假设条件下,能量检测的频谱感知性能最优且易于工程实现,但在非高斯噪声背景下,其感知性能大大下降甚至无效。针对这一问题,利用对消处理方法来提高能量检测在非高斯噪声下的频谱感知性能,通过将检测统计量与先验背景噪声进行对消预处理,在降低噪声非高斯度的同时提高了统计量的信噪比,从而提高了检测概率,并进一步提高了宽带频谱感知性能。利用USRP、GNURADIO和MATLAB设计并实现了频谱感知平台,同时验证了该算法的可行性。     

基于FLOM和LSTM的频谱感知算法

为解决现有频谱感知检测方法在Alpha噪声中性能下降的问题，提出一种基于FLOM(fractional low order moment, FLOM)和LSTM(long short-term memory, LSTM)神经网络的频谱感知算法。利用分数低阶矩在解决非高斯噪声下感知性能退化的强大能力以及长短期记忆神经网络在解决时序特性问题上的强大处理能力，设计一个频谱感知算法。不同于现有的基于能量和协方差矩阵等二阶统计量的频谱感知，利用FLOM对数据进行分数低阶预处理后，LSTM通过提取分数低阶协方差矩阵的特征进行决策。仿真结果表明，该算法比传统的频谱感知算法具有更高的检测概率。在低信噪比下，基于分数低阶矩阵感知的LSTM检测方案的检测概率比其它基于数据驱动的检测方法改善了至少15%。     

基于门限自适应的分布式检测融合算法

贝叶斯检测融合策略是一种比较传统的分布式检测融合方法,必须给定待检测现象的先验概率和各局部传感器的虚警概率和漏检概率,而在现实应用中,统计量是未知的或者是随时间变化的.因此,研究了一种纽曼一皮尔逊准则下的门限自适应分布式检测系统的融合算法.算法可根据观测数据,自动在线调整门限,使得局部传感器检测达到最佳,从而提高系统的检测性能.计算机仿真的结果表明,算法能较快地收敛,相对局部传感器,融合中心的检测性能也明显地有了提高.     

基于自动删除算法的分布式CFAR检测算法

研究分布式恒虚警(CFAR)检测系统在非均匀干扰背景中进行优化检测.针对多传感器分布式恒虚警检测系统在非均匀干扰背景中容易出现检测概率下降或者虚警率提高的问题,提出了一种基于自动删除算法的分布式恒虚警检测算法.算法是一种基于局部检测统计量的分布式CFAR检测算法,充分利用了局部检测器的观测信息,提高了检测性能,同时采用...     

多径线性调频信号的循环相关检测及性能分析

针对复线性调频信号在多径条件下的循环相关检测问题进行研究。利用循环平稳信号的循环相关特性，分析了多径条件下复线性调频信号的循环相关特性，在此基础上从连续和离散两方面构造了基于循环自相关包络的检测统计量，然后分析了检测性能，推导了检测统计量的输出信噪比表达式，从中可看出输出信噪比较输入信噪比和单径条件下的输出信噪比都有了较大提高。最后采用计算机从输出信噪比和虚警概率两方面进行了仿真，得到了基于循环自相关包络检测统计量的检测性能曲线，仿真试验验证了本文的结论。     

基于绝对值累积的主用户动态到达频谱感知算法

在认知无线电网络的主用户动态到达频谱感知场景中,针对拉普拉斯脉冲噪声干扰导致频谱检测性能下降的问题,提出基于绝对值累积(AVC)的频谱感知算法。假设接收到的主用户信号服从泊松分布,对接收信号进行AVC处理抑制脉冲噪声干扰,并将处理信号累积求和作为判决统计量,得到判决统计量的均值与方差,求出判决门限理论表达式以判断主用户是否动态到达,从而实现频谱感知。理论分析与仿真结果表明,该算法在不同虚警概率、信噪比及累积求和采样点数量下的检测概率均优于改进的能量检测算法。     

一种计算复杂度低的双端通话检测算法

在声学回音消除中,近端语音的出现会导致模拟回音路径的自适应滤波器发散,一个成熟的声学回音消除器应该包含有双端通话检测算法。针对这个问题,提出了一种计算复杂度低的、基于麦克风信号与误差信号的互相关双端通话检测算法。同时,该算法与滤波器系数缓存机制相结合以进一步提高系统的鲁棒性。实验结果表明,该算法具有良好的检测性能,可以对双端通话的出现和消失做出快速响应,同时能显著提高系统在双端通话环境下的回音消除效果。     

A Variable Step-Size Affine Projection Algorithm Designed for Acoustic Echo Cancellation

The adaptive algorithms used for acoustic echo cancellation (AEC) have to provide 1) high convergence rates and good tracking capabilities, since the acoustic environments imply very long and time-variant echo paths, and 2) low misadjustment and robustness against background noise variations and double-talk. In this context, the affine projection algorithm (APA) and different versions of it are very attractive choices for AEC. However, an APA with a constant step-size parameter has to compromise between the performance criteria 1) and 2). Therefore, a variable step-size APA (VSS-APA) represents a more reliable solution. In this paper, we propose a VSS-APA derived in the context of AEC. Most of the APAs aim to cancel $p$ (i.e., projection order) previous a posteriori errors at every step of the algorithm. The proposed VSS-APA aims to recover the near-end signal within the error signal of the adaptive filter. Consequently, it is robust against near-end signal variations (including double-talk). This algorithm does not require any a priori information about the acoustic environment, so that it is easy to control in practice. The simulation results indicate the good performance of the proposed algorithm as compared to other members of the APA family.      

适用于即时翻译系统的改进仿射投影算法 *

针对即时翻译系统应用中存在双端对讲干扰和模型噪声的问题,提出了一种适用于便携式即时翻译系统的改进变步长仿射投影算法。新算法在收敛步长中引入近端信号能量统计量和滤波器收敛程度统计量,根据统计量的改变实时调整步长参数,防止算法发散。仿真结果表明,与传统自适应滤波算法和改进仿射投影算法相比,所提出的算法不但可以有效克服双端对讲干扰,而且在收敛速度、稳态失调等方面也有明显改善。     

一种低延时双端发音检测方法

回声消除是提高通信中语音信号质量的关键技术。其主要难题是回声路径估计的自适应算法的控制逻辑。为了达到较好的回声消除效果,自适应滤波器需要在双端发音模式下缓慢更新或停止更新,而在其他模式快速学习。现有的双端发音检测算法没有考虑检测延时问题,使得滤波器在停止更新前已经发散,严重影响了回声消除的效果。针对该问题,在滤波器收敛时回声消除至少达到10 dB的假设前提下,对传统的能量比较法进行改进,提出低延时的解决方案。实验结果表明,该方法比相关比较法的检测延时减少了35毫秒以上。     

Robust extended multidelay filter and double-talk detector for acoustic echo cancellation

We propose an integrated acoustic echo cancellation solution based on a novel class of efficient and robust adaptive algorithms in the frequency domain, the extended multidelay filter (EMDF). The approach is tailored to very long adaptive filters and highly auto-correlated input signals as they arise in wideband full-duplex audio applications. The EMDF algorithm allows an attractive tradeoff between the well-known multidelay filter and the recursive least-squares algorithm. It exhibits fast convergence, superior tracking capabilities of the signal statistics, and very low delay. The low computational complexity of the conventional frequency-domain adaptive algorithms can be maintained thanks to efficient fast realizations. We also show how this approach can be combined efficiently with a suitable double-talk detector (DTD). We consider a corresponding extension of a recently proposed DTD based on a normalized cross-correlation vector whose performance was shown to be superior compared to other DTDs based on the cross-correlation coefficient. Since the resulting DTD also has an EMDF structure it is easy to implement, and the fast realization also carries over to the DTD scheme. Moreover, as the robustness issue during double talk is particularly crucial for fast-converging algorithms, we apply the concept of robust statistics into our extended frequency-domain approach. Due to the robust generalization of the cost function leading to a so-called M-estimator, the algorithms become inherently less sensitive to outliers, i.e., short bursts that may be caused by inevitable detection failures of a DTD. The proposed structure is also well suited for an efficient generalization to the multichannel case.     

Improved variable step-size NLMS adaptive filtering algorithm for acoustic echo cancellation

Acoustic echo canceller (AEC) is used in communication and teleconferencing systems to reduce undesirable echoes resulting from the coupling between the loudspeaker and the microphone. In this paper, we propose an improved variable step-size normalized least mean square (VSS-NLMS) algorithm for acoustic echo cancellation applications based on adaptive filtering. The steady-state error of the NLMS algorithm with a fixed step-size (FSS-NLMS) is very large for a non-stationary input. Variable step-size (VSS) algorithms can be used to decrease this error. The proposed algorithm, named MESVSS-NLMS (mean error sigmoid VSS-NLMS), combines the generalized sigmoid variable step-size NLMS (GSVSS-NLMS) with the ratio of the estimation error to the mean history of the estimation error values. It is shown from single-talk and double-talk scenarios using speech signals from TIMIT database that the proposed algorithm achieves a better performance, more than 3 dB of attenuation in the misalignment evaluation compared to GSVSS-NLMS, non-parametric VSS-NLMS (NPVSS-NLMS) and standard NLMS algorithms for a non-stationary input in noisy environments.     

基于声回波抵消两路算法的改进更新逻辑

声回波抵消两路算法被广泛用来检测系统双向通话;基于声回波抵消两路算法,提出了一种改进的控制更新逻辑。此更新逻辑通过比较滤波器的回波返回损失(ERLE),判断是否对滤波器进行更新。此改进更新逻辑能正确检测系统双向通话,避免滤波器的错误更新,并提高两路算法的收敛速度,减小存储器资源和计算量。仿真结果证实了此更新逻辑的有效性。     

An improved new error estimation algorithm for optimal filter lengths for stereophonic acoustic echo cancellation

In this paper, we propose an new error estimate algorithm (NEEA) for stereophonic acoustic echo cancellation (SAEC) that is based on the error estimation algorithm (EEA) in [Nguyen-Ky T, Leis J, Xiang W. An improved error estimate algorithm for stereophonic acoustic echo cancellation system. In: International conference on signal processing and communication systems, ICSPCS’2007, Australia; December 2007]. In the EEA and NEEA, with the minimum error signal fixed, we compute the filter lengths so that the error signal may approximate the minimum error signal. When the echo paths change, the adaptive filter automatically adjusts the filter lengths to the optimum values. We also investigate the difference between the adaptive filter lengths. In contrast with the conclusions in [Khong AWH, Naylor PA. Stereophonic acoustic echo cancellation employing selective-tap adaptive algorithms. IEEE Trans Audio, Speech, Lang Process 2006;14(3):785-96, Gansler T, Benesty J. Stereophonic acoustic echo cancellation and two channel adaptive filtering: an overview. Int J Adapt Control Signal Process 2000;4:565-86, Benesty J, Gansler T. A multichannel acoustic echo canceler double-talk detector based on a normalized cross-correlation matrix. Acoust Echo Noise Control 2002;13(2):95-101, Gansler T, Benesty J. A frequency-domain double-talk detector based on a normalized cross-correlation vector. Signal Process 2001;81:1783-7, Eneroth P, Gay SL, Gansler T, Benesty J. A real-time implementation of a stereophonic acoustic echo canceler. IEEE Trans. Speech Audio Process 2001;9(5):513-23, Gansler T, Benesty J. New insights into the stereophonic acoustic echo cancellation problem and an adaptive nonlinearity solution. IEEE Trans. Speech Audio Process 2002; 10(5):257-67, Benesty J, Gansler T, Morgan DR, Sondhi MM, Gay SL. Advances in network and acoustic echo cancellation. Berlin: Springer-Verlag; 2001], our simulation results have shown that the filter lengths can be different. Our simulation results also confirm that the NEEA is better than EEA and SM-NLMS algorithm in terms of echo return loss enhancement.