助听器麦克风阵列数据鲁棒自适应处理

M. Hoffman
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引用次数: 3

摘要

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
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Robust adaptive processing of microphone array data for hearing aids
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.<>
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Multidimensional scaling analysis of head-related transfer functions Robust adaptive processing of microphone array data for hearing aids Local silencing of room acoustic noise using broadband active noise control Computationally efficient compression of audio signals by means of RIQ-DPCM A simplified source/filter model for percussive sounds
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