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DSP algorithms for digital hearing instruments.

机译:用于数字助听器的DSP算法。

摘要

A new digital filter bank design and a new compression algorithm that can improve the performance of hearing instruments located completely in the ear canal (CIC) are developed in the thesis. In order to assess state-of-the-art hearing instruments employing advanced signal processing techniques the DynamEQ-II analog hearing instrument developed by the Gennum Corporation was studied extensively. A sophisticated SIMULINK model, involving the use of audio files, was developed to evaluate the performance characteristics of the strategies and algorithms used in the DynamEQ-II. The RangeEar algorithm employed in the DigiFocus hearing instrument from the Oticon Company was also studied using SIMULINK in a similar manner. Two recommended improvements for a new hearing instrument are presented. The first improvement involves the use of an eight-band digital filter bank based on an interpolated finite impulse response (IFIR) prototype filter that has been optimized using delay elements to give a maximally flat overall magnitude response. The resulting group delay is a constant and less than the value where self-hearing and u22lip readingu22 problems occur. The second improvement uses a new compression algorithm based on a model of the human auditory system. The new algorithm replaces the existing constant homomorphic multiplication algorithms with an acoustic signal intensity weighted multiplication. The resulting nonlinear compression ratio expands low level signals and compresses high level signals in such a manner so as to improve noise immunity and increase the intelligibility of the sound. The MIT hearing loss simulator was employed to evaluate the effectiveness of the new proposed filter bank and compression algorithm by analysis of and listening to actual test audio files.Dept. of Electrical and Computer Engineering. Paper copy at Leddy Library: Theses u26 Major Papers - Basement, West Bldg. / Call Number: Thesis2001 .O53. Source: Masters Abstracts International, Volume: 41-04, page: 1157. Adviser: W. C. Miller. Thesis (M.A.Sc.)--University of Windsor (Canada), 2001.
机译:本文提出了一种新的数字滤波器组设计和一种新的压缩算法,可以改善完全位于耳道内的助听器的性能。为了评估采用先进信号处理技术的最先进的助听器,对由Gennum Corporation开发的DynamEQ-II模拟助听器进行了广泛的研究。开发了涉及音频文件使用的复杂SIMULINK模型,以评估DynamEQ-II中使用的策略和算法的性能特征。 Oticon公司的DigiFocus助听器中使用的RangeEar算法也以类似的方式使用SIMULINK进行了研究。提出了针对新的助听器的两项建议的改进。第一项改进涉及使用基于内插有限脉冲响应(IFIR)原型滤波器的八频段数字滤波器组,该滤波器已使用延迟元件进行了优化,以提供最大平坦的整体幅度响应。所得的组延迟是一个常数,并且小于发生自听和“唇读”问题的值。第二个改进是使用基于人类听觉系统模型的新压缩算法。新算法用声音信号强度加权乘法代替了现有的恒定同态乘法算法。所得到的非线性压缩比以这样一种方式扩展低电平信号并压缩高电平信号,从而提高抗噪性并增加声音的清晰度。通过分析和收听实际的测试音频文件,使用MIT听力损失模拟器评估新提出的滤波器组和压缩算法的有效性。电气和计算机工程系。莱迪图书馆的纸质副本:论文主要论文-西楼地下室。 /电话号码:Thesis2001 .O53。资料来源:Masters Abstracts International,第41-04卷,第1157页。顾问:W。C. Miller。论文(硕士)-温莎大学(加拿大),2001。

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    Onat Erkan.;

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