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研究生:張智興
研究生(外文):Chih-Hsinh Chang
論文名稱:旁波帶適應性濾波器結合波束構成器於助聽器背景噪音之衰減
論文名稱(外文):A NOISE REDUCTION TECHNIQUE BY ADOPTING SUBBAND ADAPTIVE FILTERING AND BEAMFORMING FOR HEARING AIDS SYSTEM
指導教授:戴政祺
指導教授(外文):Cheng-Chi Tai
學位類別:碩士
校院名稱:國立成功大學
系所名稱:電機工程學系碩博士班
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2002
畢業學年度:90
語文別:英文
論文頁數:70
中文關鍵詞:適應性濾波器波束構成器旁波帶適應性濾波器
外文關鍵詞:beamformeradaptive filteringsubband adaptive filtering
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旁波帶適應性濾波器結合波束構成器於助聽器背景噪音之衰減

張智興* 戴政祺**
國立成功大學電機工程學系
摘要

於此論文,我們應用噪音消除的技術於助聽器系統上。在演算法方面,我們提出以旁波帶適應性濾波器結合波束構成器做為中心主軸。傳統助聽器的結構相較於以前述方法所成的數位助聽器簡單。傳統助聽器同時放大周遭的聲音,包括語音訊號和噪音。因為噪音和語音訊號同時被放大,所以助聽器使用者在吵雜的環境下不能清楚的聽到語音訊號。我們以聲音來源的方向判別語音訊號或噪音,並配合適應性噪音消除器和適應性波束構成器來做語音訊號的保留和噪音的消除。於是建構了一套以適應性噪音消除器為基礎的系統,而此系統只保留感測器正前方的語音訊號,並將其他方向來的噪音做衰減。
此系統也使用旁波帶適應性濾波器的方法,以減少系統的演算負擔和功率消耗,且因為頻帶的切割,使得系統在運算處理上有更大的彈性。我們以模擬和真實的噪音來測試系統的響應,並從測試結果中可看出確實能夠達到將正前方語音訊號保留,並消除其他方向噪音的需求。


關鍵字:適應性濾波器,波束構成器,旁波帶適應性濾波器
A NOISE REDUCTION TECHNIQUE BY ADOPTING SUBBAND ADAPTIVE FILTERING AND BEAMFORMING FOR HEARING AIDS SYSTEM


Chih-Hsing Chang* Cheng-Chi Tai**
Department of Electrical Engineering
National Cheng Kung University, Tainan Taiwan R.O.C.


ABSTRACT

In this paper, a noise cancellation technique is proposed for hearing aids system. The proposed algorithm adopted subband adaptive filtering with combination of beamformer technique. The structure of conventional hearing aids is relatively simple. They amplify ambient sounds that include speech signal as well as noise. Because noise and human speech signal are amplified at the same time, hearing aid users can’t clearly hear speech signal in noisy environment.
The direction of sound can be used to discriminate speech signal from noise by combining adaptive noise canceller and adaptive beamformer. A system is developed that based on the constrained adaptive noise canceller to preserve speech signal from straight ahead and minimize background noise arriving from other directions.
This system also uses subband adaptive filtering method to reduce the requirement for computation, power consumption and enhance the flexibility of the system. The performance of this system is illustrated using simulated and real-world noises. The results show that the developed system can reserve the right ahead speech signal and substantially reject noises from the other directions.


Key word : adaptive filtering,beamformer,subband adaptive filtering
Contents

ABSTRACT ………………………………………………………… I
Chapter 1 INTRODUCTION ……………………………………... 1
1-1 Background ………………………………………………… 1
1-2 Organization of this thesis …………………………………. 2
Chapter2 THEORETICAL BACKGROUND ………………… 3
2-1 NLMS (Normalized LMS) algorithm ……………………… 3
2-2 Array signal process ………………………………………….. 4
2-3 Griffiths-Jim beamformer .…………………………………… 6
2-4 Griffiths-Jim beamformer for the binaural type hearing aids .. 8
2-5 Subband adaptive filtering …………………………………. 10
CHAPTER3 HARDWARE SYSTEM AND SOFTWARE DESIGN ………………………………………………………… 15
3-1 System structure …………………………………………….. 15
3-2 Introduction of C5402 DSK and C54x DSP ……………… 16
3-3 Software development ……………………………………… 21
CHAPTER4 RESULTS ………………………………………….. 25
4-1 Beampattern ………………………………………………. 25
4-2 Direction testing …………………………………………….. 28
4-3 Noise reduction experiments ……………………………… 31
4-3-1 NLMS adaptive beamformer experiments and results ………………………………………. 31
4-3-2 FLS adaptive beamformer experiments and results …... 35
4-3-3 Subband adaptive beamformer experiments and results .. 39
4-3-4 Real-world noise reduction experiments ……………. 50
4-3-5 Human speech experiments …………………………... 52
CHAPTER5 CONCLUSIONS …………….………………... 55
5-1 Conclusion …………………………………………………... 55
5-2 Further development in the future …………………………... 56
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