| Project/Area Number |
07457395
|
|---|
| Research Category |
Grant-in-Aid for Scientific Research (B)
|
| Allocation Type | Single-year Grants |
|---|
| Section | 一般 |
|---|
| Research Field |
Otorhinolaryngology
|
|---|
| Research Institution | TOHOKU UNIVERSITY |
|---|
Principal Investigator |
SONE Toshio Res.Inst.Elect.Comm, Tohoku University Prof., 電気通信研究所, 教授 (50005245)
|
|---|
| Co-Investigator(Kenkyū-buntansha) |
KAWASE Tetsuaki School of Medicine, Tohoku University Lecturer, 医学部・付属病院, 講師 (50169728)
TAKASAKA Tomonori School of Medicine, Tohoku University Prof., 医学部, 教授 (80004646)
TAKANE Shouichi Res.Inst.Elect.Comm, Tohoku University Res.Assoc., 電気通信研究所, 助手 (90236240)
OZAWA Kenji Res.Inst.Elect.Comm, Tohoku University Res.Asoc., 電気通信研究所, 助手 (30204192)
SUZUKI Yoiti Res.Inst.Elect.Comm, Tohoku University Assoc.Prof., 電気通信研究所, 助教授 (20143034)
池田 勝久 東北大学, 医学部, 助教授 (70159614)
|
|---|
| Project Period (FY) |
1995 – 1996
|
| Project Status |
Completed (Fiscal Year 1996)
|
| Budget Amount *help |
¥5,700,000 (Direct Cost: ¥5,700,000)
Fiscal Year 1996: ¥2,100,000 (Direct Cost: ¥2,100,000)
Fiscal Year 1995: ¥3,600,000 (Direct Cost: ¥3,600,000)
|
|---|
| Keywords | hearing aid, / digital hearing aid / environmental noise / noise suppression / wavelet transform / beam-forming / power spectrum / 1マイクロホン型環境音抑圧 / 環境音抑圧 / フーリエ変換 / 基本ウェーブレット / 減衰正弦波 / 2マイクロホン型環境音抑圧 |
|---|
| Research Abstract |
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.
|
