音频场景分析作为助听器的控制系统

M. Roch, T. Huang, Jing Liu, R. Hurtig
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引用次数: 1

摘要

众所周知,简单的扩音并不能帮助许多听力受损的听众。因此,已经为数字助听器提出了许多信号增强算法。这些算法中的许多只在某些环境中有效。快速和正确地检测听觉场景元素的能力可以允许从可用例程库中选择/参数化增强算法。在这项工作中,作者研究了一种频域压缩算法的实时参数化,该算法保留了形成峰比,从而提高了一些严重感音神经性听力损失患者在2-3 kHz范围内的语音理解能力。最佳压缩比取决于声信号的质量。本文简要介绍了频率压缩技术,并描述了一种高斯混合模型分类器,它可以根据广泛的声学类别动态设置频率压缩比,我们称之为队列。我们讨论了在通用计算机上实现的原型模拟器的结果。
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Audio scene analysis as a control system for hearing aids
It is well known that simple amplification cannot help many hearing-impaired listeners. As a consequence of this, numerous signal enhancement algorithms have been proposed for digital hearing aids. Many of these algorithms are only effective in certain environments. The ability to quickly and correctly detect elements of the auditory scene can permit the selection/parameterization of enhancement algorithms from a library of available routines. In this work, the authors examine the real time parameterization of a frequency-domain compression algorithm which preserves formant ratios and thus enhances speech understanding for some individuals with severe sensorineural hearing loss in the 2-3 kHz range. The optimal compression ratio is dependent upon qualities of the acoustical signal. We briefly review the frequency-compression technology and describe a Gaussian mixture model classifier which can dynamically set the frequency compression ratio according to broad acoustic categories which we call cohorts. We discuss the results of a prototype simulator which has been implemented on a general purpose computer.
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