基于正交基函数的声回波系统递归识别

Lester S. H. Ngia
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引用次数: 12

摘要

在免提电话或视频会议应用中,在封闭环境中,扬声器和麦克风之间存在声反馈耦合,从而产生声回波。FIR滤波器因其结构简单而被广泛应用于声学回波消除器中。然而,在本文中,Kautz和Laguerre滤波器结构被证明是比FIR滤波器更有效的回波消除器,因为它们可以用更少的参数准确地描述声回波系统。这些滤波器是由它们各自的标准正交Kautz和Laguerre基函数构建的。理论和数值结果表明,时变声回波路径主要是由其时变的零点而不是时变的声学极点引起的。因此,估计了Kautz和Laguerre滤波器的极点,并且可以保持固定或在需要时偶尔更新。用批处理高斯-牛顿算法估计极点。然后,可以用大多数适合线性回归模型的递归算法来估计Kautz和Laguerre滤波器的系数,例如归一化LMS算法。研究表明,所提出的Kautz和Laguerre滤波器作为声回波消除器中的滤波器结构,比FIR和IIR滤波器具有更好的收敛和跟踪性能。
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Recursive identification of acoustic echo systems using orthonormal basis functions
In hands-free telephone or video conference application, there exists an acoustic feedback coupling between the loudspeaker and microphone in an enclosed environment, which creates the acoustic echo. FIR filters are commonly used in acoustic echo cancellers because of their simple structure. However, in this paper, the Kautz and Laguerre filter structures are shown to be more efficient echo cancellers than the FIR filters, because they can describe accurately the acoustic echo system with fewer parameters. These filters are built from their respective orthonormal Kautz and Laguerre basis functions. The proposal is motivated by some theoretical and numerical results that the time-varying acoustic echo path is basically due to its time-varying zeros and not its time-invariant acoustical poles. Therefore, the poles of the Kautz and the Laguerre filters are estimated, and can be kept fixed or updated occasionally if required. The poles are estimated by a batch Gauss-Newton algorithm. Then, the coefficients of the Kautz and Laguerre filters can be estimated by most recursive algorithms that are suitable for linear regression models, e.g., the normalized LMS algorithm. Generally, it is shown that the proposed Kautz and Laguerre filters, as the filter structures in an acoustic echo canceller, have better convergence and tracking properties than the FIR and IIR filters.
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