研究生: |
賴昭賢 Zhao-xian Lai |
---|---|
論文名稱: |
多聲道音響系統接收特性之即時適應性調整 Real-Time Adaptation of Reception Characteristics for Multi-Channel Audio Systems |
指導教授: |
柳宗禹
Tzong-yeu Leou |
口試委員: |
邱炳樟
Bin-chang Chieu 林敬舜 Ching-shun Lin |
學位類別: |
碩士 Master |
系所名稱: |
電資學院 - 電子工程系 Department of Electronic and Computer Engineering |
論文出版年: | 2009 |
畢業學年度: | 97 |
語文別: | 中文 |
論文頁數: | 97 |
中文關鍵詞: | 串音現象 、等化器 、雜訊消除 、聽覺效應 、調適演算法 |
外文關鍵詞: | Crosstalk Canceling, Equalizer, Noise Canceling, Auditory Effect, Adaptive Filter Theory |
相關次數: | 點閱:244 下載:4 |
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本論文應用具多喇叭與多麥克風之多聲道音響空間傳播模型偵測特定空間內聲音傳播特性,並據此建構多聲道音響控制系統,使用調適濾波器技術使得位於多個接收端之訊號能依事先規劃之接收特性達成自動調整的功能。多聲道音響控制系統於各信號源至各喇叭間均加入一獨立調適濾波器用以補償修正各接收端所合成之音訊特性。由於人耳對各頻段的敏感度不同,因此調適濾波器所使用的誤差函數已根據人耳聽覺效應在各個頻帶間進行加權,使得誤差評估方式能接近人耳聽覺特性,因而產生較佳的聽覺效果。
基於本論文所建構聲音傳播模型與誤差評估方式,吾等利用MATLAB進行各種不同音訊傳播與特性調整之模擬,當中包含:(1)補償音訊信號源至各接收點間聲音傳播特性;(2)消除各音訊信號在各接收點所產生之串音現象;(3)消除空間中噪音源傳播至各接收點的噪音。甚至可同時結合以上三種音訊功能,系統仍可正確無誤的進行調整。而其實驗結果顯示此音訊架構對於以上各音訊功能皆能有效的調整,並且在許多相類似的音訊功能都可利用此音訊架構完成。
This thesis is mainly devoted to the development of a multiple-channel sound wave transmission model associated with a multiple-speaker and multiple-microphone environment for the purpose of evaluating the sound transmission characteristics in an enclosed space, which leads to the construction of a multiple-channel audio adaptation system that employs the adaptive filter techniques to achieve prescribed signal characteristics at multiple reception points through the use of automatic adaptation. This multiple-channel audio adaptation system inserts an independent adaptive FIR filter between each signal source and each speaker to compensate and to adjust the characteristics of the corresponding signal at each receiving end. As the susceptibility of the human ears is clearly frequency dependent, the cost function used in the adaptive filters of the multiple-channel audio adaptation system has taken into account of the static psychoacoustic model that applies frequency-dependent weighting to respective audio components, which produces a cost evaluation scheme similar to the human ears and enhances the overall system performance.
Based upon the sound transmission model and the error criteria we have developed, a number of applicable system functionalities of the multiple-channel audio adaptation system have been simulated and investigated by using the Matlab tool. These application areas include: (1) systems that compensate for the signal transmission characteristics measured at multiple reception points; (2) systems capable of eliminating the crosstalk at multiple reception points; (3) systems that cancel the noise components at multiple reception points. Moreover, systems with all three of the above functionalities can be implemented and have been verified by simulation. In addition, the system model developed in this thesis is applicable to many additional application areas with minor modifications.
[1] R. Rabenstein and A. Zayati, “A Direct Method to Computational Method,” IEEE International Conference on Acoustics, Speech, and Signal Processing, pp. 957-960, vol. 2, March 1999.
[2] A. Kelloniemi, V. Valimaki, and L. Savioja, “Simulation of Room Acoustics Using 2-D Digital Waveguide Meshes,” IEEE International Conference on Acoustics, Speech and Signal Processding, pp. 313-316, vol. 5, May 2006.
[3] R. de Vries, A. P. Berkhoff, C. H. Slump, and O. E. Herrmann, “Digital Compensation of Nonlinear Distortion in Loudspeakers,” IEEE International Conference on Acoustics, Speech, and Signal Processing, pp. 165-168, vol. 1, April 1993.
[4] W. Frank, R. Reger, and U. Appel, “Loudspeaker Nonlinearities- Analysis and Compensation”, Conf. Record 26th, Asilomar Conference on Signals, Systems and Computers, Pacific Grove, CA, pp. 756-760, October 1992.
[5] http://hostilefork.com/2007/12/16/perfect-reconstruction-equalizer/.
[6] Y. Haneda, S. Makino, and Y. Kaneda, “Multiple-Point Equali- zation of Room Transfer Functions by Using Common Acoustical Poles,” IEEE Transactions on Speech and Audio Processing, vol. 5, no. 4, July 1997.
[7] M. R. Petraglia, R. G. Alves, and P. S. R. Diniz, “New Structures for Adaptive Filtering in Subbands with Critical Sampling,” IEEE Transaction on Signal Processing, vol. 48, no. 12, December 2000.
[8] B. S. Atal and M. R. Schroeder, “Apparent Sound Source Translator,” U. S. Patent 3 236 949, 1962.
[9] M. Miyoshi and Y. Kaneda, “Inverse Filtering of Room Acoustics,” IEEE Transactions on Acoustics, Speech, Signal Processing, vol. 36, pp. 145-152, 1988.
[10] S. J. Elliott, I. M. Stothers, and P. A. Nelson, “A Multiple Error LMS Algorithm and Its Application to the Active Control of Sound and Vibration,” IEEE Transactions on Acoustics, Speech and Signal Processing, vol. ASSP-35, no. 10, October 1987.
[11] D. R. Morgan, “An Analysis of Multiple Correlation Cancellation Loops with a Filter in the Auxiliary Path,” IEEE Transactions on Acoustics, Speech and Signal Processing, vol. 28, pp. 454-467, 1980.
[12] P. A. Nelson, H. Hamada, and S. J. Elliott, “Adaptive Inverse Filters for Stereophonic Sound Reproduction,” IEEE Transactions on Signal Processing, vol. 40, no. 7, July 1992.
[13] J. Lim and C. Kyriakakis, “Multirate Adaptive Filtering for Immer- sive Audio,” Proc. IEEE ICASSP, Salt Lake City, UT, vol. 5, pp. 3357-3360, May 07-11, 2001.
[14] B. Gardner and K. Martin, “HRTF Measurements of a KEMAR Dummy-Head Microphone,” Technical Report #280 MIT Media Lab Perceptual Computing, May 1994.
[15] T. D. Rossing, Handbook of Acoustics, Springer, 2007.
[16] J. Huopaniemi, L. Savioja and M. Karjalainen, “Modeling of Reflections and Air Absorption in Acoustical Spaces a Digital Filter Design Approach,” in Proceedings of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA'97), New Paltz, New York, USA, October 19-22, 1997.
[17] Y. Huang, J. Benesty and J. Chen, “Generalized Crosstalk Cancel- lation and Equalization Using Multiple Loudspeakers for 3D Sound Reproduction at the Ears of Multiple Listeners,” IEEE International Conference on Acoustics, Speech, and Signal Processing, pp. 405-408, March 31-April 4, 2008.
[18] D. D. Rife and John Vanderkooy, “Transfer-Function Measurement with Maximum-Length Sequences,” J. Audio Eng. Soc., vol. 37, no. 6, June 1989.
[19] P. Alfke, “Efficient Shift Registers, LFSR Counters, and Long Pseudo-Random Sequence Generators,” Xilinx Application Note: XAPP052, July 1996.
[20] H. Fletcher and W. A. Munson , “Loudness, Its Definition, Measure and Calculation,” J. Acoust. Soc. Am., vol. 5, pp. 82-108, 1933.
[21] http://www.zainea.com/physiologicalsound.htm.
[22] J. W. Adams, “FIR Digital Filters with Least-Squares Stopbands Subject to Peak-Gain Constraints,” IEEE Transactions on Circuits and Systems, vol. 29, no. 4, April 1991.
[23] A. H.Sayed, Fundamentals of Adaptive Filtering, Wiley-IEEE Press, 2003.
[24] S. Haykin, Adaptive Filter Theory, 4th ed., Prentice-Hall, 2002.
[25] J. Huopaniemi, “Virtual Acoustics and 3-D Sound in Multimedia Signal Processing,” Dr. Eng. thesis, Helsinki University of Technology Laboratory of Acoustics and Audio Signal Processing, 1999.
[26] S. J. Elliott, I. M. Stothers, and P. A. Nelson, “A Multiple Error LMS Algorithm and Its Application to the Active Control of Sound and Vibration,” IEEE Transactions on Acoustics, Speech and Signal Processing, vol. ASSP-35, no. 10, October 1987.
[27] S.J. Elliott and P.A. Nelson, “Active Noise Control,” IEEE Signal Processing Magazine, October 1993.
[28] A. Mouchtaris, P. Reveliotis and C. Kyriakakis, “Inverse Filter Design for Immersive Audio Rendering Over Loudspeakers,” IEEE Transactions on Multimedia, vol. 2, no. 2, June 2000.
[29] 張晉維,主動式噪音控制耳機之設計與實現,國立台灣科技大學電子工程系碩士學位論文,一月,2009。
[30] TMS320C6000 Optimizing Complier v6.0 Beta User Guide, Texas Instruments, Literature Number: SPRU187N, July 2005.
[31] TMS320C6000 Programmer’s Guide, Texas Instruments, Literature Number: SPRU198G, August 2002.
[32] TMS320C672x DSP Real-Time Interrupt Reference Guide, Texas Instruments, Literature Number: SPRU717, April 2005.
[33] TMS320C6727B, Floating-Point Digital Signal Processors, Texas Instruments, Literature Number: SPRS370E, September 2006, Revised July 2008.
[34] TMS320C67x DSP Library Programmer’s Reference Guide, Texas Instruments, Literature Number: SPU657, February 2003.