计算机科学与技术

基于GSPAP的子带自适应声反馈消除算法

  • 梁维谦 ,
  • 郑方 ,
  • 陈朝阳 ,
  • 陈高鋆
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  • 1. 集美大学 信息工程学院, 海上通信与智能电子系统福建省高等学校重点实验室, 厦门 361021;
    2. 清华大学 信息技术研究院, 语音和语言技术中心, 北京 100084;
    3. 厦门莱亚特医疗器械有限公司, 厦门 361009

收稿日期: 2016-05-11

  网络出版日期: 2017-07-15

GSPAP based sub-band adaptive feedback cancellation algorithm

  • LIANG Weiqian ,
  • ZHENG Fang ,
  • CHEN Chaoyang ,
  • CHEN Gaojun
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  • 1. Key Laboratory of Maritime Communication and Intelligent Electronic Systems of Fujian Province, College of Information Engineering, Jimei University, Xiamen 361021, China;
    2. Center for Speech and Language Technologies, Research Institute of Information Technology, Tsinghua University, Beijing 100084, China;
    3. Xiamen LA and Associates Medical Equipment Co. LTD., Xiamen 361009, China

Received date: 2016-05-11

  Online published: 2017-07-15

摘要

声反馈抑制需求鲁棒、 高效的自适应滤波器。该文提出一种基于Gauss-Seidel伪放射投影(Gauss-Seidel pseudo affine projection,GSPAP)的子带自适应声反馈消除算法。通过加权重叠相加滤波器组进行子带划分,子带上采用参考信号的自相关矩阵取代能量对滤波器的自适应步长进行归一化;在自相关矩阵的迭代公式中引入基于输入信号和参考信号能量最大值的自整定系数,增强算法鲁棒性;选用二阶 GSPAP 迭代法对自相关矩阵解算求逆,以平衡算法性能与复杂度。实验结果表明:在相同滤波器长度的条件下,该文方法获得11~22 dB的稳态增益增量,比时域归一化最小均方误差(normalized least mean square,NLMS)方法提升20%~55%。

本文引用格式

梁维谦 , 郑方 , 陈朝阳 , 陈高鋆 . 基于GSPAP的子带自适应声反馈消除算法[J]. 清华大学学报(自然科学版), 2017 , 57(7) : 707 -712 . DOI: 10.16511/j.cnki.qhdxxb.2017.25.026

Abstract

Acoustic feedback cancellation needs robust and efficient adaptive filters. A Gauss-Seidel pseudo affine projection (GSPAP) sub-band adaptive filter based algorithm was developed using the auto-correlation matrix of the reference signal in the sub-band domain, instead of the power, to normalize the adaptation step size. A self-adjusting coefficient calculated from the maximum power of the input and output sub-band signals, was used to improve the robustness of the auto-correlation matrix. A two-order GSPAP algorithm was used to find the inverse of the auto-correlation matrix to balance the performance and complexity. Results from the added stable gain (ASG) experiment indicate that this algorithm gives 11 to 22 dB improved feedback cancellation for different types of hearing aids, which is 22% to 55% better than given by the time-domain normalized least mean square (NLMS) algorithm.

参考文献

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