簡易檢索 / 詳目顯示

研究生: 陳茂林
Mao-Lin Chen
論文名稱: 模糊邏輯卡爾曼濾波器語音強化辨識系統設計
The Fuzzy Logic Kalman Filter Speech Enhancement and Recognition System Design
指導教授: 施慶隆
Ching-Long Shih
口試委員: 徐佳銘
Jia-Ming Shyu
蔡清池
Ching-Chih Tsai
呂福生
Fu-Sheng Lu
許新添
Hsin-Teng Hsu
學位類別: 博士
Doctor
系所名稱: 電資學院 - 電機工程系
Department of Electrical Engineering
論文出版年: 2009
畢業學年度: 97
語文別: 中文
論文頁數: 140
中文關鍵詞: 語音辨識卡爾曼濾波器模糊邏輯理論模糊邏輯卡爾曼濾波器訊噪比
外文關鍵詞: Fuzzy logic theory, Kalman filter, Speech recognition, FLKF, SNR
相關次數: 點閱:409下載:2
分享至:
查詢本校圖書館目錄 查詢臺灣博碩士論文知識加值系統 勘誤回報

本論文改進傳統的語音濾波器無法消除中低頻雜訊與濾波器參數調整過高時會損害語音特色的缺憾,並將其應用在語音辨識上,然後以雙旋轉翼直昇機的音控來驗證設計結果。我們提出結合卡爾曼濾波器和模糊邏輯理論形成一個新型的模糊邏輯卡爾曼濾波器(Fuzzy Logic Kalman Filter, FLKF)來消除背景噪音並保留語音特色。當應用於吵雜環境時,本系統將特別有效,因系統會將卡爾曼濾波器的輸出訊號及誤差訊號變化量當作模糊邏輯系統的輸入,調變卡爾曼濾波器的 Q參數,以達到消除背景雜訊及語音強化的效果,來提升訊噪比與語音辨識率。
本論文設計一個語音辨識系統來控制雙旋轉翼直昇機。實作驗證內容計有(1)語音辨識的訊號前置處理效果;(2)使用波形分析與訊噪比計算,深入探討濾波器性能;(3)雙旋轉翼直昇機語音辨識音控驗證。經比較驗證本文所設計的新式模糊邏輯卡爾曼濾波器有較佳的雜訊消除能力與語音強化的效果,不但提高訊噪比,且可應用於設計語音辨識器的音控系統。


This thesis presents an improved speech filter ability to remove mid-low frequency noise and reserve the original speech characteristics. A speech recognition system is then designed to control the twin rotor helicopter system with voice input. A new fuzzy logic Kalman filter is proposed (FLKF) to filter out the noisy background and preserve speech characteristics. This system is particularly effective in collecting speech in a noisy environment. The parameter Q of the Kalman filter is tuned to achieve the strengthening effect of eliminating background noise and to promote the SNR and speech recognition rate.
A speech recognition system is designed to illustrate the twin rotor helicopter speech control function. The practical tests includes (1) the signal’s leading processing effect on speech recognition, (2) using wave analysis and signal noise ratio calculation to confer the filter’s function, and (3) verification of the speech recognition voice control for a twin rotor helicopter speech recognition controller. It is shown that the new Fuzzy Logic Kalman filter can perform better noise eliminating ability and speech enhancement function. In summary, the proposed method can improve the signal noise rate and have application in a speech recognition voice control system.

目 錄 中文摘要--------------------------------------------------------------------------------------I 英文摘要-----------------------------------------------------------------------------------------II 誌謝----------------------------------------------------------------------------------------------III 符 號 索 引------------------------------------------------------------------------------------VI 圖表索引---------------------------------------------------------------------------------------IX 表格索引--------------------------------------------------------------------------------------XI 第1章 緒論--------------------------------------------------------------------------------1 1.1研究動機----------------------------------------------------------------------1 1.2 文獻探討------------------------------------------------------------------------------ 1 1.3研究目的及其重要--------------------------------------------------------------------5 1.4 論文架構與研究內容----------------------------------------------------------------6 第2章 語音訊號濾波器設計------------------------------------------------------7 2.1 語音訊號處理-----------------------------------------------------------------------7 2.2語音線性預估模型------------------------------------------------------------------14 2.3線性預估係數解法------------------------------------------------------------------17 2.4語音強化與語音品質評量---------------------------------------------------------17 2.5濾波器演算法------------------------------------------------------------------------19 2.5.1卡爾曼濾波器-----------------------------------------------------------------20 2.5.2正規化最小均方適應性濾波器--------------------------------------------24 2.5.3遞迴式最小平方適應性濾波器--------------------------------------------25 2.5.4模糊邏輯卡爾曼濾波器------------------------------------------------------26 2.6實驗驗證---------------------------------------------------------------------30 2.7實驗結論-----------------------------------------------------------------------47 第3章 語音辨識應用-------------------------------------------------------------------------49 3.1 語音訊號前置處理-----------------------------------------------------------------50 3.2端點偵測及取音框-----------------------------------------------------------51 3.3特徵參數計算-------------------------------------------------------------55 3.4 動態時間校準法--------------------------------------------------------------------58 3.5實驗驗證------------------------------------------------------------------------------62 3.6實驗結論------------------------------------------------------------------------------69 第4章 音控雙旋轉翼直昇機應用---------------------------------------------------------70 4.1系統描述------------------------------------------------------------------------------70 4.2數學模式------------------------------------------------------------------------------71 4.3定位控制器設計---------------------------------------------------------------------77 4.3.1 PD控制非線性補償----------------------------------------------------------79 4.3.2 模糊PID控制-----------------------------------------------------------------79 4.4 實驗驗證-----------------------------------------------------------------------------80 4.4.1實驗室雙旋轉翼直昇機的物理參數與螺旋槳特性----------81 4.4.2 穩定位置控制實驗模擬----------------------------------------------------85 4.4.3音控辨識實驗模擬------------------------------------------------------------86 4.5實驗結論------------------------------------------------------------------------------91 第5章結論與建議----------------------------------------------------------------------------92 5.1結論------------------------------------------------------------------------------------93 5.2未來展望------------------------------------------------------------------------------95 參考文獻----------------------------------------------------------------------------------------96 作者簡介---------------------------------------------------------------------------------------104 論文登錄 授權書 符 號 索 引 脈衝訊號 視窗的加權 音框能量 過零率 自相關函數 變異函數 增益 預估誤差訊號 線性預估係數 自相關函數矩陣 全極點模型階數 語音自相關函數 功率頻譜密度 噪音能量 乾淨語音訊號的能量 SNR 訊噪比 k時刻的系統狀態 k時刻對系統的控制量 k時刻的測量值 測量矩陣 過程的噪音 測量的噪音 Q、R 變異數(Covariance) 上一狀態預測的結果 上一狀態最佳化結果 的轉置矩陣 輸出系統狀態估測 殘差序列 適應性濾波器係數 誤差訊號 收斂因數 正規化收斂因數 X(k)的功率估計值 遞迴總誤差量 增益向量 自相關係數反矩陣 遺忘因數 模糊誤差值 誤差變化量 模糊輸出值 、 、 歸屬函數 第 個權重 系統運動能量 系統慣量 主旋轉翼的螺旋槳輸入 尾旋轉翼的螺旋槳輸入 尾旋轉翼死區補償 主旋轉翼重力補償 俯仰角度初始均衡位置 黏滯摩擦係數 圖 表 索 引 圖2-1離散時間訊號圖-------------------------------------------------------------------------8 圖2-2加視窗及取音框-------------------------------------------------------------------------9 圖2-3語音訊號的能量曲線圖---------------------------------------------------------------11 圖2-4語音訊號過零率曲線------------------------------------------------------------------12 圖2-5語音發音模型圖-----------------------------------------------------------------------14 圖2-6卡爾曼濾波器目標物狀態與量測流程圖------------------------------------------20 圖2-7卡爾曼濾波器整體系統(u(k)=0)----------------------------------------------------23 圖2-8模糊邏輯卡爾曼濾波器架構圖-----------------------------------------------------27 圖2-9歸屬函數圖-----------------------------------------------------------------------------28 圖2-10吵雜語音訊的FLK濾波器解調 Q、R值依據圖--------------------------------31 圖2-11 FLK濾波器雜訊消除語音強化輸出波形----------------------------------------33 圖2-12波音747引擎聲干擾(0dB)下的各濾波器輸出訊號分析比較----------------35 圖2-13在波音747引擎聲干擾(0dB)下的濾波器輸出訊號波形圖比較-------------36 圖2-14直昇機螺旋槳聲干擾(5dB)下的濾波器輸出訊號分析比較------------------37 圖2-15直昇機螺旋槳聲干擾(5dB)下的濾波器輸出波形比較------------------------38 圖2-16汽車啟動引擎聲干擾(-3dB)下的濾波器輸出訊號分析比較-----------------39 圖2-17汽車啟動引擎聲干擾(-3dB)下的濾波器輸出波形比較-----------------------40 圖2-18 F16戰鬥機引擎聲干擾(-6dB)下的濾波器輸出訊號分析比較---------------41 圖2-19 F16戰鬥機引擎聲干擾(-6dB)下的濾波器輸出波形比較---------------------42 圖2-20吵雜語音6dB使用模糊解調與未用模糊解調的輸出訊號分析比較-------44 圖2-21使用模糊解調與未用模糊解調的語音強化輸出波形比較-------------------45 圖2-22使用模糊解調與未用模糊解調的雜訊消除輸出波形比較-------------------46 圖3-1語音辨識架構圖------------------------------------------------------------------------51 圖3-2語音辨識端點偵測取音框圖---------------------------------------------------------55 圖3-3 Itakura路逕限制-----------------------------------------------------------------------60 圖3-4「啟動、前進、後退、左右」語音訓練圖-------------------------------------------64 圖3-5待測語音端點取音框圖---------------------------------------------------------------64 圖3-6「0~9語句」語音訓練圖--------------------------------------------------------------67 圖3-7「0~9語句」語音辨識待測語音圖--------------------------------------------------67 圖4-1直昇機模型------------------------------------------------------------------------------71 圖4-2俯、仰角定位點脈波值定義圖-------------------------------------------------------71 圖4-3直昇機模型構造圖( )--------------------------------------------------74 圖4-4直昇機的靜止均衡位置圖------------------------------------------------------------77 圖4-5模糊PID 控制器-----------------------------------------------------------------------80 圖4-6主旋轉翼的螺旋槳特性曲線 ------------------------------------------------82 圖4-7尾旋轉翼的DC馬達角速度軌跡 -------------------------------------85 圖4-8尾旋轉翼的螺旋槳特性曲線 ------------------------------------------------85 圖4-9 PD與模糊PID定點位置穩定控制比較------------------------------------------86 圖4-10干擾下模糊PID定點穩定控制----------------------------------------------------86 圖4-11雙旋轉翼直昇機語音訓練圖-------------------------------------------------------90 圖4-12雙旋轉翼直升機語音辨識待測語音圖-------------------------------------------90 表 格 索 引 表2-1模糊輸出規則表------------------------------------------------------------------------29 表2-2吵雜語音FLK濾波器解調Q值的訊雜比選擇表--------------------------------32 表2-3波音747引擎聲干擾(0dB)下的SNR值比較表-----------------------------------37 表2-4直昇機螺旋槳聲干擾(5dB)下的SNR值比較表----------------------------------39 表2-5汽車啟動引擎聲干擾(-3dB)下的SNR值比較表---------------------------------41 表2-6 F16戰鬥機引擎聲干擾(-6dB)下的SNR值比較表-------------------------------43 表2-7對吵雜合成語音做雜訊消除及語音強化比較表---------------------------------46 表3-1「0~9語句」語音辨識動態時間校準法表------------------------------------------68 表4-1一個線性模糊控制規則表------------------------------------------------------------80 表4-2實驗室雙旋轉翼直昇機的物理參數表---------------------------------------------81 表4-3雙旋轉翼直升機語音辨識動態時間校準法表------------------------------------91

參考文獻

[1] 王小川,“語音訊號處理”,全華科技圖書公司,台灣、台北,2005年2月。
[2] 王立新,“自適應模糊系統與控制”,國防工業出版社,中國,北京,pp98-102,1995年10月。
[3] 余玉田,“雙自由度主動噪音控制之設計與實作”,國立中興大學機械工程研究所,碩士論文,2000年6月。
[4] 卓大靖、王聖鋐,“適應性模糊強跟蹤卡爾曼濾波器於導航系統之設計”,國立台灣海洋大學,通訊與導航工程系,碩士論文,2005年6月。
[5] 林士能,“具有線上輔助修正之前饋主動噪音控制器的設計與實作”,中興大學機械工程研究所,碩士論文,2001年6月。
[6] 林明鋒,“模型直昇機姿態控制”,碩士論文,國立中央大學電機工程研究所,2001年6月。
[7] 林鎮洲、徐偉群,“模糊卡爾曼濾波器應用於載具姿態判定”,國立臺灣海洋大學,機械與輪機工程學系,碩士論文,2003年6月。

[8] 楊憲東,“自動飛行控制原理與實務”,全華科技圖書,2000年5月。
[9] 楊鎮光,“Visual Basic 與語音辨識”,文魁資訊股份有限公司,臺灣、臺北,2002年6月。
[10] 葉智偉,“控制路徑頻譜整形於主動噪音控制之應用探討”, 國立中興大學機械工程研究所,碩士論文,2002年6月。
[11] 趙家賢,“結合第二路徑頻譜整形器之適應混合前饋式主動噪音控制”,國立中興,大學機械工程研究所,碩士論文,2003年6月。
[12] 顏坤銘、宋開泰,“家用機器人之語音辨識系統”,國立交通大學電機與控制工程學系,碩士論文,2002年7月。
[13] 羅子良,“含內部模式之 設計一維聲管主動噪音控制之應用與控制實作”,國立中興大學機械工程研究所,碩士論文,1998年6月。
[14] 羅華強,“訊號處理---MATLAB的應用”, 全華科技圖書公司,pp3-41,2003年8月。
[15] A. Harada, K. Nishikawa, H. Kiya, “A pipelined architecture for the normalized LMS adaptive digital filters,” IEEE Asia-Pacific Conference on Circuits and Systems, pp.73 -76, Nov, 1998.

[16] A. Subramanya, Z. Zhang, Z. Liu, A. Acero, “Speech Modeling with Magnitude-Normalized Complex Spectra and Its Application to Multisensory Speech Enhancement,” 2006 IEEE International Conference on Multimedia and Expo, pp.1157 -1160, July, 2006.

[17] A.S. Abutaleb, “Adaptive line enhancement using a random AR model,” IEEE Transactions on Acoustics, Speech, and Signal Processing, vol 38, No.7, pp.1211- 1215, July, 1990.

[18] B.T. Lilly, K.K. Paliwal, “Effect of speech coders on speech recognition performance,” ICSLP 96 Proceedings, Fourth International Conference on Spoken Language, vol. 4, pp. 2344 - 2347, Oct,1996.

[19] B. Widrow, S.D. Stearns, “Adaptive Signal Processing,” Englewood Cliffs, NJ: Prentice-Hall, 1984.
[20] C.L. Shih, M.L. Chen, J.Y. Wang, “Mathematical and Set-point Stabilizing Controller Design of a Twin Rotor MIMO System,” Asia Journal of Control, vol. 10, No. 1, February, 2008.
[21] C.L. Shih, M.L. Chen, “Mathematical Model and Stabilizing Controller Design of a Twin Rotor MIMO System,” IEEE ICSS2005 International Conference on Systems & Signals, pp934 - 939, April, 2005.
[22] C.T. Lin, “Single-channel speech enhancement in variable noise-level environment,” IEEE Transactions on Systems, Man and Cybernetics, vol. 33, No. 1, Part A, pp.137 - 143, Jan, 2003.
[23] D. Burshtein, S. Gannot, “Speech Enhancement Using a Mixture-Maximum Model,” IEEE Transaction on Vol. 10, No. 6, Speech And Audio Processing, pp. 326-332, September, 2002.
[24] E. Zavarehei, S.Vaseghi, Q. Yan, “Noisy Speech Enhancement Using Harmonic-Noise Model and Codebook-Based Post-Processing,” Audio vol 15, No. 4, pp.1194 - 1203, May, 2007.

[25] G. Chen, J. Wang, L.S. Shieh, “Interval Kalman filtering,” IEEE Transactions on Aerospace and Electronic Systems, vol. 33, No. 1, pp. 250 - 259, Jan, 1997.

[26] G. Kitagawa, W. Gersch, “A smoothness prior’s long AR model method for spectral estimation,” IEEE Transactions on Automatic Control, vol. 30, No. 1, pp.57 - 65, Jan, 1985.

[27] G. Sreenu, P. N. Girija, M. N. Prasad and M. Nagamani, “A Human Machine Speaker Dependent Speech Interactive System,” India Annual Conference, Proceedings of the IEEE INDICON, Dec, pp. 349 - 351, 2004.

[28] H. G.Hirsch, C. Ehrlicher, “Noise estimation techniques for robust speech recognition,” in Proc. 20th IEEE ICASSP’95, pp.153-156, May, 1995.
[29] H. Sorenson, Kalman Filtering, “Theory and Application,” IEEE Press, 1985.
[30] H. Suzuki, H. Zen, Y. Nankaku, C. Miyajima, K. Tokuda, and T. Kitamure, “Speech Recognition Using Voice-Characteristic Dependent Acoustic Models,” IEEE International Conference on Acoustics, Speech, and Signal Processing, vol .1, pp.I-740 - I-743, April, 2003.

[31] H. Y. Chang, N. K. Soo, S.Rahardja, “Kalman filtering speech enhancement incorporating masking properties for mobile communication in a car environment,” IEEE International Conference on Multimedia and Expo, vol. 2, pp.1343 - 1346, June, 2004.
[32] H.X. Li, G. Chen, “Fuzzy-PID Controller is a Quasi-Sliding Mode Controller with Dual Features, ” International Journal of Fuzzy Systems, vol. 5, No. 2, pp. 131-140, June, 2003.
[33] H.Y. Chang, S. Rahardja, N. K. soo, “Autoregressive Parameter Estimation for Kalman Filtering Speech Enhancement,” Acoustics, IEEE International Conference on Speech and Signal Processing, ICASSP 2007, vol. 4, pp.IV-913 - IV-916, April, 2007.
[34] I. Cohen, B. Berdugo, “Noise Estimation by Minima Controlled Recursive Averaging for Robust Speech Enhancement,” IEEE Signal Processing letters, vol. 9, No. 1, pp.12-15, January, 2002.
[35] I. Kokkinos, P. Maragos, “Nonlinear Speech Analysis Using Models for Chaotic System,” IEEE Transactions on Speech and Audio Processing, vol. 13, No. 6, pp.1098 - 1109, Nov, 2005.
[36] I. Mann, S. McLaughlin, “Synthesizing natural-sounding vowels using a nonlinear dynamical model,” Signal Process, vol. 81, pp. 1743 -1756, 2001.
[37] I. Cohen, B. Berdugo, “A Multi-Microphone Post-Filtering Approach for Speech Enhancement,” IEEE Signal Processing Letters, pp.312-322, March, 2002.
[38] I. Cohen, “Speech Enhancement Using a No causal A Priori SNR Estimator,” IEEE Signal Processing Letters, vol.11, No. 9, pp. 32 - 40, September, 2004.
[39] J. Huang, J. Zhao, Y. Xie, “Source classification using pole method of AR model,” 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP-97, vol. 1, April, pp.567 – 570, 1997.

[40] J. S. Lim, A. V. Oppenheim, “Enhancement and bandwidth compression of noisy speech,” Proc. IEEE, vol. 67, pp. 1586 -1604, Dec, 1979.
[41] J. S. Robert, L. H. Sandra, “Fundamentals of Digital Signal Processing Using MATLAB,” United States, Thomson, Feb, 2005.
[42] J. Wang, T. Zhang, “Degradation prediction method by use of autoregressive Algorithm”, IEEE International Conference on Industrial Technology, pp. 1 - 6, April, 2008.
[43] J.S. Hu, “Active Noise Cancellation in Ducts Using Internal Model-Based Control Algorithms,” IEEE Transactions on Control Systems Technology, vol.4, pp. 163-170, March, 1996.

[44] J.W. Pitton, K. Wang, B.H. Juang, “Time-frequency analysis and auditory modeling for automatic recognition of speech,” Proceedings of the IEEE, vol.84, No. 9, pp. 1199 - 1215, Sept, 1996.
[45] J.Z. Sasiadek, J. Khe, “ Sensor fusion based on fuzzy Kalman filter,” 2001 Proceedings of the Second International Workshop on Robot Motion and Control, pp.275 - 283, Oct,2001.

[46] J.T. Chien, P.Y. Lai, “Microphone Array Signal Processing for Far-Talking Speech Recognition,” Wireless Communications, 2001 IEEE Third Workshop on Signal Processing Advances, pp.322 - 325, March, 2001.

[47] J.Y. Wang, M.L. Chen, C.L. Shih, “Generalized Predictive Control in Flying Shear Equipment,” Journal of Advanced Computational Intelligence and Intelligent Informatics, vol.11 No.9, pp.1144 -1148, Nov, 2007.
[48] L. Singh, S. Sridharan, “Speech enhancement using pre-processing,” Speech and Image Technologies for Computing and Telecommunications’. Proceedings of IEEE vol. 2, pp.755 - 758, Dec, 1997.

[49] L.B. Fah, A. Hussain, S.A. Samad, “Speech enhancement by noise cancellation using neural network,” TENCON 2000. Proceedings vol.1, pp.39 - 42, Sept, 2000.

[50] L.X. Wang, J.M. Mendel, “Fuzzy Adaptive Filters, with Application to Nonlinear Channel Equalization,” IEEE Transactions on Fuzzy System, vol.1, No. 3, August 1993.
[51] L.X. Wang, “Self-Adaptive Fuzzy System Control,” national defense industry publisher, China Beijing, pp. 98 -102, Sept,1995.
[52] M. Arnold, “Reasoning about Non-Linear AR models using expectation maximization,” J. Forecast, vol.22, pp.479 - 490, Jan,2003.
[53] M. Banbrook, S. McLaughlin, and I. Mann, “Speech characterization and synthesis by nonlinear methods,” IEEE Transactions on Speech Audio Processing, vol.7, No.1, pp.1-17, Jan, 1999.
[54] M. Birgmeier, “A fully Kalman-trained radial basis function network for nonlinear speech modeling,” in Proc. Int. Conf. Neural Networks, pp.254 - 259, Nov, 1995
[55] M. Hayes, “Statistical Digital Signal Processing and Modeling,” John Wiley and Sons, 1996.
[56] M. K. Steven, “Modern Spectral Estimation: Theory and Application,” Prentice Hall, 1988.
[57] M. Montazeri, P. Duhamel, “A set of algorithms linking NLMS and block RLS algorithms,” IEEE Transactions on Signal Processing, vol. 43, No. 2, pp.444 -453, Feb, 1995.

[58] M. S. Brandstein, D. B. Ward, Eds., “Microphone Arrays: Signal Processing Techniques and Applications,” Springer-Verlag, Berlin, 2001.
[59] M.E. Dajer, J.C. Pereira, C.D. Maciel, “Nonlinear Dynamical Analysis of Normal Voices,” Seventh IEEE International Symposium on Multimedia, Dec, 2005.

[60] M. Ning, M. Bouchard, R. A. Goubran, “Speech enhancement using a masking threshold constrained Kalman filter and its heuristic implementations,” IEEE Transitions on Audio, Speech & Language Processing, vol. 14, No. 1, pp.19 -32, Jan, 2006.
[61] M. L. Chen, J. Y. Wang, C. L. Shih, “Dynamic system model predict and adaptive controller design,” Journal of Information & Optimization Sciences (EI), vol. 29, No.1, pp. 175-189, 2008.
[62] M. K. Hasan, M. R. Khan, “A Modified A Priori SNR for Speech Enhancement Using Spectral Subtraction Rules,” IEEE Signal Processing Letters, vol.11, No. 4, April, 2004
[63] N. Virag, “Single channel speech enhancement based on masking properties of the Human auditory system,” IEEE Transitions on Speech and Audio Processing, vol.7, No.2, pp. 126-137, 1999.
[64] P. Maragos, T. F. Quatieri, “Energy Separation in Signal Modulations with Application to Speech Analysis,” IEEE Transitions on Signal Processing, vol. 41, No. 10, October, 1993.
[65] P. C Loizou, “Signal processing for cochlear implants and low-rate speech coding,” IEEE Workshop on Speech Coding, Sept, pp. 68-78, 2000.

[66] P.M.T. Broersen; S. de Waele, “Automatic identification of time-series models from long autoregressive models,” IEEE Transactions on Instrumentation and Measurement, vol. 54, No. 5, pp. 1862 - 1868, Oct, 2005
[67] Q. Fan, K. Nonami, M. Nnkano, “Active Noise Control of Exhaust Duct Using Two-Degree-of-Freedom Control System with Model Matching,” JSME International Journal, Series C, Mechanical systems, machine elements and manufacturing, vol.40, No.2, 1997.
[68] R. Flynn, E. Jones, “Robust connected digit recognition using speech enhancement and an auditory model front-end,” 2007,6th International Conference on Information, Communications & Signal Processing, pp.1-4, Dec, 2007
[69] R. Martin, “Noise power spectral density estimation based on optimal smoothing and minimum statistics,” IEEE transactions on speech and audio signal processing, vol. 9, No. 5, pp.504 -512, July, 2001.
[70] R. C. Hendriks, R. Heusdens, J. Jensen, “Adaptive Time Segmentation for Improved Speech Enhancement,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 14, No. 6, pp. 2064 - 2074, Nov, 2006.

[71] R. G. Brown, P.Y.C. Hwang, “Introduction to random signals and applied Kalman filtering,” 3rd Edition, Wiley, New York, 1997.

[72] S. Haykin, “Adaptive Filter Theory,” Fourth Edition, PRENTICE HALL, Information and System Sciences Series, 2002.
[73] S. Gazor, Wei Zhang; “Speech enhancement employing Laplacian-Gaussian mixture,” IEEE Transactions on Speech and Audio Processing, vol. 13, No. 5, Part 2, pp.896 - 904, Sept, 2005
[74] S. Haykin, “Adaptive Filter Theory,” Englewood Cliffs, NJ: Prentice-Hall, 1996.
[75] S. K. Katsikas, S. D. Likothanassis, D. G. Lainiotis, “AR model identification with unknown process order,” IEEE Transactions on Acoustics, Speech and Signal Processing, vol. 38, No. 5, pp.872 - 876, May 1990.
[76] S. K. Tso, Y. H. Fung, “Methodological development of fuzzy-logic controllers from multivariable linear control,” IEEE Transactions on Systems, Man and Cybernetics, vol. 27, No. 3, pp. 566 -572, 1997.
[77] S. M. Kuo, D. R. Morgan, “Action Noise Control System: Algorithms and DSP Implementations,” 605 Third Avenue, New York, pp. 0012 - 0158, 1996.
[78] S. Yeldener, J. H. Rieser, “A background noise reduction technique based on sinusoidal speech coding systems,” IEEE International Conference on Acoustics, Speech, and Signal Processing , vol. 3, pp. 1391 - 1394, June, 2000.

[79] S. Gazor, W. Zhang, “Speech Enhancement Employing Laplacian-Gaussian Mixture,” IEEE Transactions on Speech and Audio Processing, vol. 13, No. 5,Part 2, pp. 896 - 904, Sept, 2005.
[80] S. T. Zhang, X. Y. Wei, “Fuzzy adaptive Kalman filtering for DR/GPS,” 2003 International Conference on Learning and Cybernetics Machine, vol. 5, Nov, pp. 2634 - 2637, Sept, 2003.

[81] T. Matsuura, H. Togiishi, “Two dimensional AR model of signing process and its application to on-line signature verification,” IEEE International Conference on Electronics, Circuits and Systems vol. 2, pp. 545 - 548, Sept,1998.

[82] T. Yamamoto, S. Omatu, M. Kaneda, “A Design Method of Self-Tuning PID controllers,” American Control Conference, vol. 3, 29 July, pp.3263 - 3267, 1994.
[83] T. H. Li, J. D. Gibson, “Speech Analysis and Segmentation by Parametric Filtering,” IEEE Transactions on Speech and Audio Processing, vol. 4, No.3, pp. 203 - 213, May, 1996.
[84] T. W. Lee, K. Yao, “Speech Enhancement By Perceptual Filter with Sequential Noise Parameter Estimation,” 2004 IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 1, pp. 693-699, May,2004.

[85] Twin Rotor MIMO System--Advanced Teaching Manual 1, Feedback Instruments Ltd, UK,2001.
[86] V. G. Reju, T. Y. Chow, “A Computationally Efficient Noise Estimation Algorithm for Speech Enhancement,” The IEEE Asia-Pacific Conference on Circuits and Systems, December, 2004.
[87] V. Stahl, A. Fischer, R. Bippus, “Quartile based noise estimation for spectral subtraction and Wiener filtering,” IEEE International Conference on Acoustics, Speech and Signal Processing, vol.3, pp. 1875-1878, May,2000.
[88] W. Jin, M. S. Scordilis, “Speech Enhancement by Kalman Filtering with Residual Noise Clipping,” SoutheastCon, Proceedings. IEEE, April, pp.225 -228, 2005.

[89] W. Qiao, M. Mizumoto, “PID type fuzzy controller and parameters adaptive method,” Fuzzy Sets and Systems, vol. 78, pp. 23-35, 1996.
[90] W. H. Lan, C. L. Shih, M. L. Chen, “A Patter Free Algorithm to Inspect Open-Shorts of a Flex Circuit,” the 13th throughout the country automation science and technology workshop, pp.17-18, June, 2004.
[91] W. R. Wu, P. C. Chen,“Adaptive AR modeling in white Gaussian noise,” IEEE Transactions on Signal Processing, vol. 45, No. 5, pp.1184 - 1192, May, 1997.
[92] X. Cui, A. Alwan, “Noise Robust Speech Recognition Using Feature Compensation Based on Polynomial Regression of Utterance SNR,” IEEE Transitions on Speech and Audio Processing, vol.13, No.6, pp.324-334, May, 2006.
[93] X. Hu, A. q. Hu, L. Zhao, “A robust adaptive speech enhancement system,” Neural Networks and Signal Processing, Proceedings of the 2003 International Conference on vol. 1, Dec, pp. 814 - 817, 2003.

[94] X. Huang, A. Acero, and H. W. Hon, “Spoken language Processing, a guide to theory, algorithm and system development,” Prentice Hall, 2001.
[95] Y. Cai, “A Forecasting procedure for nonlinear autoregressive time series models,” J. Forecast, vol. 24, pp335 - 351, May, 2005.
[96] Y. Ephraim, D. Malah, “Speech enhancement using a minimum mean-square error short-time spectral amplitude estimator,” IEEE Transitions on Accost, speech, signal Proc, vol.1, No.1, pp. 1109-1121, 1984.
[97] Y. Hu, C. L. Philipos, “Speech Enhancement Based on Wavelet Thresholding the Multitaper Spectrum,” IEEE Transitions on Speech and Audio Processing, vol. 12, No.1, pp. 312-412, January, 2004.
[98] Y. Itoh, “A matching algorithm between arbitrary sections of two speech data sets for speech retrieval,” 2001 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2001. Proceedings, vol. 1, pp. 593 - 596, May,2001.

[99] Y. Li, “Some Results of Fuzzy Turing Machines,” 2006.WCICA The Sixth World Congress On Intelligent Control and Automation, vol. 1, pp. 3406 -3409, 2006.

[100] Y. Nagata, T. Fujioka, M. Abe, “Speech enhancement based on auto gain control,” IEEE Transactions on Audio, Speech and Language Processing, vol. 14, No. 1, pp.177 - 190, Jan, 2006.

[101] Y. Tsuda, T. Shimamura, “An improved NLMS algorithm for channel equalization,” IEEE International Symposium on Circuits and Systems, vol. 5, pp. V-353 - V-356, May, 2002.

[102] Y. Hu, P. C. Loizou, “A subspace approach for enhancing speech corrupted by colored noise,” Signal Processing Letters, IEEE, vol. 9, No.7, pp. 204 - 206, July, 2002.
[103] Y. Zheng, Z. Lin, “Time-varying autoregressive system identification using wavelets,” Acoustics, Speech, and Signal Processing, vol. 1, pp.572 - 575, June, 2000.

[104] Y. J. Chen, C. C. Wang, G. J. Jong, and B. W. Wang, “The Separation System of the Speech Signals Using Kalman Filter with Fuzzy Algorithm,” Proceedings of the First International Conference on Innovative Computing, Information and Control, pp.321-329, 2006.
[105] Y. Lee, “Channel Prediction with Cascade AR Modeling,” Telecommunications, International Conference on Internet and Web Applications and Services/Advanced, pp.40-48, Feb, 2006.

[106] Y. Lee, “Channel Prediction with Cascade AR Modeling,” AICT-ICIW '06. International Conference on Internet and Web Applications and Services/Advanced International Conference on Telecommunications, pp.40-50, Feb, 2006.

[107] Z. Goh, K. C. Tan, B.T.G Tan, “Kalman-filtering speech enhancement method based on a voiced-unvoiced speech model,” IEEE Transactions on Speech and Audio Processing, vol. 7, No. 5, pp.510 - 524, Sept, 1999.
[108] Z. Wang, X. Zhang, “A high performance speech enhancement algorithm based on double-channel adaptive noise canceling,” IEEE International Symposium on Microwave, Antenna, Propagation and EMC Technologies for Wireless Communications Proceedings, pp.983 - 986, 2005.

無法下載圖示
全文公開日期 本全文未授權公開 (校外網路)
全文公開日期 本全文未授權公開 (國家圖書館:臺灣博碩士論文系統)
QR CODE