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Environmental noise suppression algorithms for hearing aids

助听器环境噪声抑制算法

基本信息

批准号：
07457395
负责人：
SONE Toshio
金额：
$ 3.65万
依托单位：
TOHOKU UNIVERSITY
依托单位国家：
日本
项目类别：
Grant-in-Aid for Scientific Research (B)
财政年份：
1995
资助国家：
日本
起止时间：
1995 至 1996
项目状态：
已结题

项目摘要

Digital signal processing algorithms to suppress environmental noise were studied for advanced hearing aid. Algorithms with one microphone and two microphones were examined.As to digital signal processing algorithm to suppress environmental noise with one microphone, the spectral subtraction method is well known and widely used. In the method, averaged frequency spectrum of environmental noise is subtracted from the frequency spectrum of input signal. This method usually uses Fourier transform to calculate the frequency spectrum. The conventional spectrum subtraction with Fourier transform, however, gives insufficient performance. We, therefore, used wavelet transform, which was considered to be similar to the frequency analysis process in the auditory system. Using various kinds of basis functions for the wavelet analysis, hearing experiments were carried out to evaluate their performances. The result showed that the intelligibility was better restored with non-orthogonal basis functions than with orthogonal ones.Digital signal algorithms with two microphones were examined next. When multiple microphones are used, beam-forming techniques are the most promising. Though conventional beam-forming technique is effective for a free or almost free sound field, it is not very effective in a live or reverberant room. Since such deterioration is caused by the phase relation between of direct sound and reflections, we developed a new beam-forming algorithm which did not use phase relation of the input signal but focused only on its power spectrum. The new algorithm showed better performance in a dispersive sound field than the conventional alrorithm.

研究了用于高级助听器的抑制环境噪声的数字信号处理算法。在单麦克风和双麦克风的数字信号处理算法中，谱减法是一种常用的环境噪声抑制算法。在该方法中，从输入信号的频谱中减去环境噪声的平均频谱。这种方法通常使用傅里叶变换来计算频谱。然而，传统的傅里叶变换的频谱减法给出了不足的性能。因此，我们使用小波变换，这被认为是类似的频率分析过程中的听觉系统。采用不同的小波基函数，进行了听觉实验，以评估其性能。结果表明，非正交基函数比正交基函数能更好地恢复语音的可懂度。当使用多个麦克风时，波束形成技术是最有前途的。虽然传统的波束形成技术对于自由或几乎自由的声场是有效的，但它在现场或混响室中不是很有效。由于这种恶化是由直达声和反射声之间的相位关系引起的，我们开发了一种新的波束形成算法，该算法不使用输入信号的相位关系，而只关注其功率谱。在频散声场中，新算法比传统算法具有更好的性能。