Robust adaptive processing of microphone array data for hearing aids

M. Hoffman
{"title":"Robust adaptive processing of microphone array data for hearing aids","authors":"M. Hoffman","doi":"10.1109/ASPAA.1993.379992","DOIUrl":null,"url":null,"abstract":"The problem of adaptively combining the outputs of an array of microphones as a single input for a hearing aid is investigated. A robust processor based on a constrained minimum variance optimization approach is used. One fundamental criteria employed in designing this robust beamformer limits the amount of cancellation of the desired signal. The results presented include the effects of acoustic headshadow, small room reverberation, microphone placement uncertainty, and desired speaker location uncertainty. Performance improvement is measured as a predicted change in the speech reception threshold (SRT) between single microphone and multi-microphone conditions. Performance improvements are demonstrated relative to the \"best\" single microphone in the array for block optimum and adaptive spatial filters. The performance of the block optimum arrays is shown to be attainable with adaptive implementations. A fast-attack, slow release input signal power averager allows the adaptive processor to avoid instabilities commonly experienced with nonstationary, impulsive inputs such as speech.<<ETX>>","PeriodicalId":270576,"journal":{"name":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","volume":"73 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"1993-10-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"3","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/ASPAA.1993.379992","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 3

Abstract

The problem of adaptively combining the outputs of an array of microphones as a single input for a hearing aid is investigated. A robust processor based on a constrained minimum variance optimization approach is used. One fundamental criteria employed in designing this robust beamformer limits the amount of cancellation of the desired signal. The results presented include the effects of acoustic headshadow, small room reverberation, microphone placement uncertainty, and desired speaker location uncertainty. Performance improvement is measured as a predicted change in the speech reception threshold (SRT) between single microphone and multi-microphone conditions. Performance improvements are demonstrated relative to the "best" single microphone in the array for block optimum and adaptive spatial filters. The performance of the block optimum arrays is shown to be attainable with adaptive implementations. A fast-attack, slow release input signal power averager allows the adaptive processor to avoid instabilities commonly experienced with nonstationary, impulsive inputs such as speech.<>
助听器麦克风阵列数据鲁棒自适应处理
研究了一组麦克风输出自适应组合为助听器单输入的问题。采用了一种基于约束最小方差优化方法的鲁棒处理器。设计这种鲁棒波束形成器时采用的一个基本准则限制了期望信号的抵消量。给出的结果包括声头阴影、小房间混响、麦克风放置不确定性和期望扬声器位置不确定性的影响。性能改进是通过预测单麦克风和多麦克风条件下语音接收阈值(SRT)的变化来衡量的。对于块优化和自适应空间滤波器,相对于阵列中“最佳”单麦克风,性能得到了改进。通过自适应实现,可以达到块最优数组的性能。快速攻击,慢释放输入信号功率平均器允许自适应处理器避免非平稳,脉冲输入(如语音)通常经历的不稳定性
本文章由计算机程序翻译,如有差异,请以英文原文为准。
求助全文
约1分钟内获得全文 求助全文
来源期刊
自引率
0.00%
发文量
0
×
引用
GB/T 7714-2015
复制
MLA
复制
APA
复制
导出至
BibTeX EndNote RefMan NoteFirst NoteExpress
×
提示
您的信息不完整,为了账户安全,请先补充。
现在去补充
×
提示
您因"违规操作"
具体请查看互助需知
我知道了
×
提示
确定
请完成安全验证×
copy
已复制链接
快去分享给好友吧!
我知道了
右上角分享
点击右上角分享
0
联系我们:info@booksci.cn Book学术提供免费学术资源搜索服务,方便国内外学者检索中英文文献。致力于提供最便捷和优质的服务体验。 Copyright © 2023 布克学术 All rights reserved.
京ICP备2023020795号-1
ghs 京公网安备 11010802042870号
Book学术文献互助
Book学术文献互助群
群 号:481959085
Book学术官方微信