跳到主要內容

臺灣博碩士論文加值系統

(216.73.216.56) 您好!臺灣時間:2025/12/09 22:45
字體大小: 字級放大   字級縮小   預設字形  
回查詢結果 :::

詳目顯示

: 
twitterline
研究生:陳建男
研究生(外文):Chen, Chien-Nan
論文名稱:助聽器回授路徑測量及適應性的噪音消除演算法
論文名稱(外文):Measurement of feedback paths and adaptive noise cancellation algorithms in hearing aids
指導教授:桑梓賢
指導教授(外文):Sang, Tzu-Hsien
學位類別:碩士
校院名稱:國立交通大學
系所名稱:電子工程系所
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:2009
畢業學年度:98
語文別:英文
論文頁數:39
中文關鍵詞:助聽器回授路徑噪音消除
外文關鍵詞:hearing aidsfeedback pathnoise cancellation
相關次數:
  • 被引用被引用:0
  • 點閱點閱:452
  • 評分評分:
  • 下載下載:110
  • 收藏至我的研究室書目清單書目收藏:2
本篇論文主要探討助聽器回授路徑(feedback path)的測量及使用適應性濾波器(adaptive filter)消除噪音的演算法模擬。

我們首先用一個耳內型(ITC)助聽器,裡面無放大電路僅有接收器(receiver)及麥克風(microphone),接上脈衝產生器(pulse generator)使用掃頻(sweep stimulus)的方式驅動助聽器的接收器並接收麥克風的聲音得到回授路徑的頻率響應(frequency response),再由程式將之轉回時域的脈衝響應(impulse response),我們的測量結果可供消除回授演算法的參考及實現。

卡爾曼濾波器(Kalman filter)是一種有效率的適應性濾波器且可應用在時變系統上。尤其是更新估記狀態時僅需計算前一個狀態估記值及新得到的資料,所以只有前個狀態需要儲存。因此我們考慮將卡爾曼濾波器列入助聽器消除噪音演算法的可能性。我們做了一些卡爾曼濾波器在單一頻帶消除白雜訊的學習,至於在分頻濾波上加上卡爾曼濾波器因為可能增加大量運算故先暫時予以保留。
In this thesis, we focus on the measurement of feedback path of hearing aids and the simulation of adaptive filter algorithm for noise cancellation.

First of all we put an ITC hearing aid embodying only a receiver and a microphone in the artificial ear in the anechoic chamber. We use the pulse generator to inject the sweep signal to the receiver and receive the sound from the microphone to get the frequency response. Then make use of Matlab to transform it into impulse response. Our measurement result may supply to the realization of the feedback cancellation algorithm.

Kalman filter is an efficient adaptive filter and can be used for time-varying system. It only needs the estimated state from the previous time step and the current measurement to compute the estimate for the current state, so only the previous estimate requires storage. Therefore we consider the possibility of using the Kalman filter in hearing aids for noise cancellation. The single-band Kalman filter for white noise cancellation is studied while the multi-band Kalman filter is kept aside due to the possible surge in computational cost.
Chapter 1 Introduction 1
1.1 Overview of hearing aids 1
1.2 Introduction to hearing aid systems 3
Chapter 2 Measurement of feedback paths in hearing aids 7
2.1 Introduction to the feedback path 7
2.2 ITC digital hearing instrument 9
2.3 Sweep stimulus method 11
2.4 Modeling the EFP in the time-domain 15
Chapter 3 Adaptive noise cancellation algorithms in hearing aids 17
3.1 Introduction 17
3.2 Kalman filtering for white noise cancellation 19
3.3 Simulation 22
3.4 Discussion 30
Chapter 4 Conclusions and future work 33
4.1 Conclusions 33
4.2 Future work 34
Bibliography 35
About the Author 39
[1] Grzegorz Szwoch, Bozena Kostek, “Waveguide model of the hearing aid earmold system,” Diagnostic Pathology, May 2006.
[2] Jingbo Yang, Meng Tong Tan and Joseph S. Chang, “Modeling External Feedback Path of an ITE Digital Hearing Instrument for Acoustic Feedback Cancellation,” Circuits and Systems, 2005. ISCAS 2005. IEEE International Symposium on 23-26 May 2005 Page(s):1326-1329 Vol.2
[3] Hsiang-Feng Chi, Shawn X. Gao, Sigfrid D. Soli and Abeer Alwan, “Band-limited feedback cancellation with a modified filtered-X LMS algorithm for hearing aids,” Speech Communication 39 (2003) 147-161.
[4] Ann Spriet, Geert Rombouts, Marc Moonen, Member, IEEE, and Jan Wouters, “Combined Feedback and Noise Suppression in Hearing Aids,” IEEE Transactions on audio, speech, and language processing, Vol. 15, No. 6, August 2007.
[5] D. K. Bustamante et. al. , “ Measurement and adaptive suppression of acoustic feedback in hearing aids,” Proc. Int. Conf. Acoustics, Speech, Signal Processing, pp.2017-2020,1989
[6] S.F. Lybarger, “Acoustic Feedback Control, ” The Vanderbilt Hearing-Aid Report edited by G.A. Studebaker, 1989
[7] M.R. Stison et. al., “Effects of handset proximity on hearing aid feedback,” J. Acoust. Soc. Am. 115, 1147,2004
[8] D.P. Egolf, “Simulating the open-loop transfer function as a means for understanding acoustic feedback in hearing aid,” JASA. 85(1),1989

[9] J. Kates, “A Time-Domain Digital Simulation of Hearing Aid Response,” J. Rehb. Res. Dev., vol. 27, issue 3, 1990
[10] J.S. Lim, “Evaluation of a correlation subtraction method for enhancing speech degraded by additive white noise,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-26, pp. 471-472, Oct. 1978
[11] S. Boll, “Suppression of acoustic noise in speech using spectral subtraction,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-27, pp. 113-120, Oct. 1979
[12] R. J. Mcaulay and M. L. Malpass, “Speech enhancement using sorf-decision noise suppression filter,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-28, pp. 137-145, Apr. 1980
[13] Y. Ephraim and D. Malah, “Speech enhancement using minimum mean-square error short-time spectral amplitude estimator,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-32, pp. 1109-1121, Dec. 1984
[14] J.S. Lim and A. V. Oppenheim, “All-pole modeling of degraded speech,” IEEE Trans. Acoust., Speech, Signal Processing, vol. ASSP-26, pp. 197-210, Oct. 1978
[15] J. H. L. Hansen and M. A. Clement, “Constrained iterative speech enhancement with application to speech recognition,” IEEE Trans. Signal Processing, vol. 39, pp. 795-805, Apr. 1991
[16] T. V. Sreenivas and P. Kirnapure, “Codebook constrained Wiener filtering for speech enhancement,” IEEE Trans. Speech, Audio Processing, vol. 4, pp. 383-389, Sept. 1996
[17] Y. Cheng and D. O’Shaughnessy, “Speech enhancement based conceptually on auditory evidence,” IEEE Trans. Signal Processing, vol.39, pp. 1943-1954, Sept. 1991
[18] J. Chen and A. Gersho, “Adaptive postfiltering for quality enhancement of coded speech,” IEEE Trans. Speech Audio Processing, vol.3, pp. 59-71, Jan. 1995
[19] Y. Ephraim and H. L. van Tree, “A signal subspace approach for speech enhancement,” IEEE Trans. Speech Audio Processing, vol.3, pp. 251-266, July 1995
[20] S. H. Jensen, P. H. Hansen, S. D. Hansen, and J.A. Sorensen, “Reduction of broad-band noise in speech by truncated QSVD,” IEEE Trans. Speech Audio Processing, vol.3, pp.439-448, Nov. 1995
[21] Y. Ephraim, “A Bayesian estimation approach for speech enhancement using hidden Markov models,” IEEE Trans. Signal Processing, vol.40, pp. 725-735, Apr. 1992
[22] K. Y. Lee and K. Shirai, “Efficient recursive estimation for speech enhancement in color noise,” IEEE Signal Processing Lett., vol. 3, pp. 196-199, July 1996
[23] K. K. Paliwal and A. Basu, “A speech enhancement method based on Kalman filtering,” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing, pp. 177-180, Apr. 1987
[24] J. D. Gibson, B. Koo, and S. D. Grey, “Filtering of colored noise for speech enhancement and coding,” IEEE Trans. Signal Processing, vol.39, pp. 1732-1741, Aug. 1991
[25] B. Lee, K. Y. Lee, and S. Ann, “An EM-base approach for parameter enhancement with an application to speech signals,” Signal Process., vol. 46, no. 1, pp. 1-14, Sept. 1995
[26] M. Nied´zwiecki and K. Cisowski, “Adaptive scheme for elimination of broadband noise and impulsive disturbance from AR and ARMA signals,” IEEE Trans. Signal Processing., vol. 44, pp. 528-537, Mar. 1996
[27] Wen-Rong Wu, and Po-Cheng Chen, “Subband Kalman filtering for speech enhancement,” IEEE Trans. On circuits and systems-II: Analog and Digital Signal Processing, vol. 45, no.8, Aug. 1998


[28] Y. T. Kuo, T. J. Lin, Y. T. Li, W. H. Chang, C. W. Liu ,and S. T Young, “Design of ANSI S1.11 Filter Bank for Digital Hearing Aids,” Electronics, Circuits and Systems,2007. ICECS 2007. 14TH IEEE International Conference, pp. 242-245, Dec. 2007
[29] http://www.hearingconsultants.com.au/body_products.html, “How hearing aids work today,”
[30] Trench W. F., “An algorithm for the inversion of finite Toeplitz matrices,” J. Soc. Indust. Appl. Math., vol.12, pp. 515-522, 1964
[31] Mingsian R. Bai, Ping-Ju Hsieh, and Kur-Nan Hur, “Optimal design of minimum mean-square error noise reduction algorithms using the simulated annealing technique, ” J. Acoust. Soc. Am. 125 934 (2009)
[32] ITU-T Rec. P.835, “Subjective test methodology for evaluating speech communication systems that include noise suppression algorithm,” International Telecommunications Union, Geneva, Switzerland, 2003
QRCODE
 
 
 
 
 
                                                                                                                                                                                                                                                                                                                                                                                                               
第一頁 上一頁 下一頁 最後一頁 top