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研究生:黃興洋
研究生(外文):Hsing-Yang Huang
論文名稱:AAC音訊編碼器設計與實現
論文名稱(外文):Design and Implementation of AAC audio coder
指導教授:顧孟愷
指導教授(外文):Mong-Kai Ku
學位類別:碩士
校院名稱:國立臺灣大學
系所名稱:資訊工程學研究所
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:2004
畢業學年度:92
語文別:英文
論文頁數:64
中文關鍵詞:AACFFTO2DFTMDCT位元分配模組雜訊估測心理聲學模型濾波器
外文關鍵詞:O2DFTBit Allocation ModuleAACPsychoacoustic ModelMDCTFilter BankFFTNoise Estimation
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AAC在所有音訊壓縮標準中提供了最高的壓縮率與品質。然而AAC編碼器的複雜度也很高。這個高複雜度主要來自於在濾波器、心理聲學模型與位元分配模組中所執行的大量運算。在這篇論文中,我們探討了快速演算法並且嘗試化減濾波器與位元分配模組的複雜度。快速的MDCT演算法利用MDCT與O2DFT (Odd Time Odd Frequency DFT)的關係,將MDCT的計算化減為N/4點的FFT計算。在位元分配模組的部分,我們應用快速初始整體增益值搜尋與雜訊估測方法發展一個新的位元分配模組。新的位元分配模組將雙迴圈架構化減為單一迴圈架構。它的複雜度相較於AAC標準中所提的位元分配模組低釵h。這些演算法可以有效地化減濾波器與位元分配模組的複雜度。相較於知名的編碼器FAAC,我們的編碼器節省了50%的編碼時間而仍然維持好的編碼品質。
AAC provides the highest compression rate and quality among all audio coding standards. However, the complexity of AAC encoder is also very high. The high complexity mainly comes from the great amount of operations performed in filter bank, psychoacoustic model, and bit allocation module. In this thesis, we study fast algorithms and try to reduce the complexities of filter bank and bit allocation module. The fast MDCT algorithm takes advantage of the relationship between MDCT and O2DFT (Odd Time Odd Frequency DFT), reduces the MDCT computation to N/4 points FFT. As for bit allocation module, we apply fast initial gain search method and noise estimation method to develop a new bit allocation module. New bit allocation module reduces the two nested loop structure to one single loop. The complexity is much lower comparing to the bit allocation module proposed in AAC standard. These algorithms efficiently reduce the complexities of filter bank and bit allocation module. Comparing to the famous coder FAAC, our coder saves 50% encoding time and still maintains good quality.
Contents i
Abstract i
Contents ii
List of Figures iv
List of Tables v
Chapter 1 Introduction 1
1.1 Motivation and Objective ...................... 1
1.2 Thesis Organization ............................2
Chapter 2 Psychoacoustics 4
2.1 Absolute Threshold of Hearing ..................4
2.2 Critical Bands .................................6
2.3 Simultaneous Masking ...........................8
2.4 Nonsimultaneous Masking ........................9
2.5 Masking Threshold .............................10
2.6 Psychoacoustic Model ..........................11
Chapter 3 AAC Algorithms 19
3.1 Filter Bank ...................................19
3.1.1 Window Shape Adaptation .....................20
3.1.2 Block Switching .............................21
3.2 Bit Allocation ................................24
3.2.1 Nonuniform Quantization .....................24
3.2.2 Scalefactor Band and Noise Shaping ..........25
3.2.3 Inner Loop ..................................25
3.2.4 Outer Loop ..................................26
3.3 Noiseless Coding ..............................29
3.3.1 Spectrum Clipping ...........................29
3.3.2 Sectioning ..................................29
3.3.3 Grouping and Interleaving ...................30
3.3.4 Scalefactors ................................31
3.3.5 Huffman Coding ..............................32
Chapter 4 Implementation 35
4.1 Fast MDCT .....................................35
4.2 New Bit Allocation Module .....................40
4.2.1 Fast Initial Gain Search ....................40
4.2.2 Noise Estimation and Stepsize Control .......43
Chapter 5 Results and Discussions 47
5.1 Test Environment ..............................47
5.2 Fast MDCT .....................................49
5.3 Fast Initial Gain Search ......................51
5.4 New Bit Allocation Module .....................53
5.5 Encoder Performance ...........................56
Chapter 6 Conclusion 58
References 59
Appendix 62
[1] ISO/IEC 13818-7, “Information Technology – Generic Coding of Moving Pictures and Associated Audio Information – Part 7: Advanced Audio Coding (AAC)”, 1997.
[2] ISO/IEC FCD 14496-3, “Information Technology – Coding of Audiovisual Objects – Part 3 Audio”, 1998.
[3] ISO/IEC 11172-3, “Coding of Moving Pictures and Associated Audio for Digital Storage Media at up to 1.5Mbit/s”, 1992.
[4] T. Painter, “Perceptual Coding of Digital Audio,” Proceedings of IEEE, Vol. 88, No. 4, April 2000.
[5] M. Bosi. et al, “ISO/IEC MPEG-2 Advanced Audio Coding,” Journal of the Audio Engineering Society, Vol.45, NO.10, October 1997, pp. 789-813
[6] E. Zwicker and H. Fastl, Psychoacoustic, Facts and Models (Springer, Berlin, Heidelberg, 1990).
[7] M. Schroeder, B. S. Atal, and J. L. Hall, “Optimizing Digital Speech Coders by Exploiting Masking Properties of The Human Ear,” J. Acoustic. Soc. Amer., pp 1647-1652, Dec. 1979.
[8] J. Johnston, “Estimation of Perceptual Entropy Using Noise Masking Criteria,” in Proc. ICASSP-88, May 1988, pp. 2524-2527
[9] B. Lincoln, “An Experimental High Fidelity Perceptual Audio Coder,” Project in MUS420 Win 97, Center for Computer Research in Music and Acoustics, Standford University, CA 94305-8180, 1998. http://ccrma-www.standford.edu/~bosse/proj/proj.html
[10] T. Y. Chang, “Research and Implementation of MP3 Encoding Algorithm,” Master’s thesis, National Chiao Tung University, Hsinchu, Taiwan, ROC, 2002.
[11] C. Y. Lee, “Analysis and Investigation of MP3 Audio Encoder with Implementation,” National Taiwan University, Taipei, Taiwan, ROC, 2003
[12] J. P. Princen, A. W. Johnston, and A. B. Bradley, “Subband/Transform Coding Using Filter Bank Designs Based on Time Domain Aliasing Cancellation,” in Proc. ICASSP-87, pp. 2161-2164.
[13] A. Sugiyama. et al., “Adaptive Transform Coding with an Adaptive Block Size (ATC-ABS),” in Proc. ICASSP 90, pp. 1093-1096, May 1990.
[14] S. R. Quackenbush and J. D. Johnston, “Noiseless Coding of Quantized Spectral Components in MPEG-2 Advanced Audio Coding”, IEEE ASSP Workshop on Applications of Signal Processing to Audio and Acoustics, 1997.
[15] R. Gluth, “Regular FFT-Related Transform Kernels for DCT/DST-based polyphase filter banks,” ICASSP 91, pp. 2205-2208 vol.3
[16] G. Bonnerot, M. Bellanger, “Odd-Time Odd-Frequency Discrete Fourier Transform for Symmetric Real-Valued Series,” IEEE Proceedings, March 1976, pp.392-393.
[17] V. Britanak and K. R. Rao, “A New Fast Algorithm for The Unified Forward and Inverse MDCT/MDST Computation,” Signal Processing, 2002, pp.443-459.
[18] K. R. Rao and P. C. Yip, Discrete Cosine and Sine Transforms, in: The Transform and Data Compression Handbook, CRC Press, Boca Raton, FL, 2001, pp. 117-195.
[19] C. K. Yang, “Investigation and Simplified Design of MP3 Audio Encoder,” Master’s thesis, National Chiao Tung University, Hsinchu, Taiwan, ROC, 2002.
[20] C. A. Serantes, A. S. Pena and N. G. Prelcic; “A Fast Noise-Scaling Algorithm For Uniform Quantization in Audio Coding Schemes,” Vol. 1 pp. 339~343, ICASSP 97.
[21] C. T. Chen, “A New Bit Allocation for MPEG Layer III,” Master’s thesis, National Chiao Tung University, Hsinchu, Taiwan, ROC, 1998.
[22] F. M. Chang, “A Study on the Fast Algorithm in MPEG-2/4 Advanced Audio Coding,” Master’s thesis, National Taipei University of Technology, Taipei, Taiwan, ROC, 2004.
[23]http://psplab.csie.nctu.edu.tw/projects/index.pl/testingsong14.html
[24] http://lame.sourceforge.net/
[25] http://www.audiocoding.com
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