首页> 外文期刊>Instrumentation and Measurement, IEEE Transactions on >Modified LMS-Based Feedback-Reduction Subsystems in Digital Hearing Aids Based on WOLA Filter Bank
【24h】

Modified LMS-Based Feedback-Reduction Subsystems in Digital Hearing Aids Based on WOLA Filter Bank

机译:基于WOLA滤波器组的改进的基于LMS的数字助听器反馈减少子系统

获取原文
获取原文并翻译 | 示例

摘要

Digital hearing aids usually suffer from acoustic feedback. This feedback corrupts the speech signal, causes instability, and damages the speech intelligibility. To solve these problems, an acoustic feedback reduction (AFR) subsystem using adaptive algorithms such as the least mean square (LMS) algorithm is needed. Although this algorithm has a reduced computational cost, it is very unstable. To avoid this situation, other AFR subsystems based on modifications of the LMS algorithm are used. Such algorithms are given as follows: 1) normalized LMS (NLMS); 2) filtered-X LMS (FXLMS); and 3) normalized FXLMS (NFXLMS). These algorithms are tested in three digital hearing aid categories: 1) in the ear (ITE); 2) in the canal (ITC); and behind the ear (BTE). The first and second categories under study suffer from great feedback effects due to the short distance between the loudspeaker and the microphone, whereas the third category suffers from these effects due to the high signal level at the hearing aid output; thus, robust AFR subsystems are needed. The added stable gains (ASGs) over the limit gain when AFR subsystems are working in the digital hearing aids are studied for all the categories. The ASG is determined as a tradeoff between two measurements: 1) segmented signal-to-noise ratio (objective measurement) and 2) speech quality (subjective measurement). The results show how the digital hearing aids that work with AFR subsystems adapted with the NLMS or the NFXLMS algorithms can achieve up to 18 dB of increase over the limit gain. After analyzing the results, it is observed that the subjective measurement always limits the achieved ASG, but when the NLMS algorithm is used, it is appreciated that the objective measurement is a good approximation for estimating the maximum achieved ASG. Finally, taking into consideration the hearing aid performances and the computational cost of each AFR subsystem implementation, an AFR subsystem based on the NLMS algorithm to adapt feed-nback-reduction filters that are 128 coefficients long is proposed.
机译:数字助听器通常遭受声学反馈。该反馈破坏语音信号,导致不稳定,并损害语音清晰度。为了解决这些问题,需要使用诸如最小均方(LMS)算法之类的自适应算法的声学反馈减少(AFR)子系统。尽管该算法具有降低的计算成本,但是它非常不稳定。为了避免这种情况,使用了基于LMS算法修改的其他AFR子系统。这样的算法如下:1)归一化LMS(NLMS); 2)过滤的X LMS(FXLMS);和3)标准化的FXLMS(NFXLMS)。这些算法已在三种数字助听器类别中进行了测试:1)入耳式(ITE); 2)在运河(ITC)中;和耳后(BTE)。由于扬声器和麦克风之间的距离短,正在研究的第一和第二类受到很大的反馈影响,而第三种则由于助听器输出处的高信号电平而受到这些影响。因此,需要强大的AFR子系统。针对所有类别,研究了AFR子系统在数字助听器中工作时超过极限增益的增加的稳定增益(ASG)。确定ASG是两个测量之间的权衡:1)分段信噪比(客观测量)和2)语音质量(主观测量)。结果表明,与采用NLMS或NFXLMS算法的AFR子系统配合使用的数字助听器如何能够在极限增益上实现多达18 dB的增加。在分析结果之后,可以观察到主观测量始终会限制所获得的ASG,但是当使用NLMS算法时,应该意识到,客观测量是估计最大已实现ASG的良好近似。最后,考虑到每个AFR子系统实现的助听器性能和计算成本,提出了一种基于NLMS算法的自适应AFR子系统,以适应128系数长的反馈减噪滤波器。

著录项

相似文献

  • 外文文献
  • 中文文献
  • 专利
获取原文

客服邮箱:kefu@zhangqiaokeyan.com

京公网安备:11010802029741号 ICP备案号:京ICP备15016152号-6 六维联合信息科技 (北京) 有限公司©版权所有
  • 客服微信

  • 服务号