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研究生: 林仁欣
Jen-Hsin Lin
論文名稱: 多層漸進式零樹小波分頻音訊壓縮技術
Scalable Audio Compression Using Wavelet Packet Decomposition and Embedded Zero Tree Coding
指導教授: 張寶基
Pao-Chi Chang
口試委員:
學位類別: 碩士
Master
系所名稱: 資訊電機學院 - 電機工程學系
Department of Electrical Engineering
畢業學年度: 88
語文別: 中文
論文頁數: 90
中文關鍵詞: 多媒體樂音壓縮小波封包人耳聲學模型多重解析
外文關鍵詞: Multimedia, audio, scalable, EZW, psychoacoustic, multi-layer
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  • 在網路的多媒體應用,將成為截取量最大的資料。而依不同的多媒體應用特性,如傳輸頻寬,傳輸即時性,有其不同的網路需求,而為了使多媒體應用達到有效率傳輸的目的,需要特別地對多媒體訊號作適當的壓縮與處理。因此,本論文發展一套多層式樂音編碼技術,來配合不同網路及其頻寬。本論文樂音壓縮標準的架構,以有多重解析度分析的小波轉換與小波封包為基礎,發展出多重解析層的樂音壓縮,並且包含重疊音框以消除區塊效應,再依音框做小波封包處理,然後優先萃取重要小波頻帶係數編碼,並利用熵編碼加以量化編碼。而進一步再利用漸進式零樹編碼、並配合人耳聲學模型之零樹搜尋編碼,再以熵編碼做進一步的編碼。本論文的壓縮分成三層,因此可針對網路之不同傳輸頻寬來選擇其適當之位元率來傳輸,其位元率分別為16 Kbps、32 Kbps、64 Kbps。在聽覺音質方面,雖然是低位元率,但是可保持一定的樂音品質,比現行在網路傳輸之相同位元率之壓縮方式優良。尤其在非常低位元率的壓縮時,本論文所提出的壓縮方式有很不錯的表現。


    Multimedia transmission over Internet is getting popular and increasingly important. In particular, scalable coding is desirable for heterogeneous network with varies bandwidths. In this work, we propose a scalable embedded zero tree wavelet packet (EZWP) audio coding system that is a scalable audio compression system using wavelet packet decomposition and embedded zero-tree coding. We focus on multi-layer low bitrate coding which delivers high perceptual quality. In the base layer, the overlapped audio segment is first transformed by wavelet packet. Then the local significant coefficients are extracted, quantized, and coded by variable length coding. In the enhancement layer and the full band layer, the residual signal that is the difference between the original and the output of the previous layer is coded via EZW with psychoacoustic model and arithmetic coding. The target bit rates for three layers are 16, 32, and 64 Kbps, respectively. The performance of the proposed coding system is only slightly inferior to MPEG-1 layer 3 while it provides bitrate scalability. Therefore, it is suitable for multimedia distribution over Internet that is composed of heterogeneous networks.

    第一章 緒論 1.1 音訊壓縮簡介 1.2 研究動機與目的 1.3 系統架構 1.4 論文架構 第二章 小波分析技術 2.1 小波轉換 ( Wavelet Transform ) 2.1.1小波分解與離散小波轉換 2.1.2多重解析度分析 2.2小波濾波器 2.3小波封包 ( Wavelet Packet ) 2.4小波分頻架構 2.5 漸進式零樹編碼 ( Embeded Zero-Tree Coding ) 2.5.1 零樹搜尋法則 2.6 連續近似量化 2.7 算術編碼 ( Arithmetic Coding ) 第三章 音訊編碼技術 3.1 一般音訊壓縮編解碼器結構 3.2 人耳聲學模型 3.2.1 基本原理與其應用 3.2.2 雜訊對單頻音的遮蔽效應 3.2.3頻音對單頻音的遮蔽效應 3.2.4 時間軸上的遮蔽效應 3.2.5模型公式 3.2.6訊號之各頻帶的最小遮蔽臨界值 3.3 MPEG音訊編碼器家族 3.4 杜比 ( Dolby ) AC 3音訊編碼器 第四章 多層漸進式人耳聲學零樹編碼系統 4.1 小波樹狀結構與係數分組 4.2 多層分解編碼 4.2.1 基礎層編碼 4.3.2 加強層編碼 4.3.3 全頻層編碼 第五章 實驗結果與討論 5.1 小波分頻濾波器組合成品質 5.2 基礎層小波分頻合成品質 5.3 加強層小波分頻合成品質 5.4 全頻層小波分頻合成品質 第六章 結論 第七章 未來展望

    [1]Andreas S. Spanias, “Speech Coding : A tutorial Review”, Proceedings Of The IEEE, VOL. 82, NO. 10, October 1994, pp 1541-1582.
    [2]Painter, T.; Spanias, A., “A review of algorithms for perceptual coding of digital audio signals” , Digital Signal Processing Proceedings, 1997. DSP 97., 1997 13th International Conference, VOL 1 , 1997 , pp 179 —208.
    [3]ISO/IEC 11172-3 : “Information technology - Coding of moving pictures and associatedaudio for digital storage media at up to about 1.5 Mbit/s - Part 3: Audio".1992 (“MPEG-1”).
    [4]James C. Mckinney, Robert Hopkins, “Digital Audio Compression Standard (AC3)”, 1994.
    [5]M. Sablatash and T. Cooklev, ”Compression of High-Quality Audio Signals, Including Recent Methods Using Wavelet Packets,” Digital Signal Processing, vol. 6, no. 10, 1996, pp. 96-107.
    [6]Y. Karelic and D. Malah, “Compression of High-Quality Audio Signals Using Adaptive Filterbanks and A Zero-Tree Coder,” Electrical and Electronics Engineers in Israel, 1995.
    [7]P. Srinivasan and L. H. Jamieson, “High-Quality Audio Compression Using an Adaptive Wavelet Packet Decomposition and Psychoacoustic Modeling,” IEEE Trans. on Signal Processing, vol. 46, no. 4, April 1998, pp. 1085-1093.
    [8]S. Boland and M. Deriche, “Audio Coding Using The Wavelet Packet Transform and A combined Scalar-Vector Quantization,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1996, pp. 1041-1044.
    [9]X. Xiong and Z. Eryuan, “Digital Audio Codec Based on the Improved Optimization Algorithm of Adaptive Wavelets and Dynamic Bit Allocation Scheme,” proceeding of ICSP’96, pp. 1523-1526.
    [10]P. Philippe, F. Moreau de Saint-Martin, M. Lever, and J. Soumagne, “Optimal Wavelet Packets for Low-Delay Audio Coding,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1996, pp. 550-553.
    [11]M. T. Chou, “Audio Compression Using Wavelet Packets And A Zero-Tree Coder With Psychoacoustic Modeling”, 碩士論文, 中央大學, 1999
    [12]C. S. Burrus, R. A. Gopinath, and H. Guo, “Introdution to Wavelets and Wavelet Transforms,” 1998.
    [13]P. E. Kudumakis and M. B. Sandler, “Wavelet Packet Based Scalable Audio Coding,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1996, pp. 41-44.
    [14]W. K. Dobson, J. J. Yang, K. J. Smart, and F. K. Guo, “High Quality Low Complexity Scalable Wavelet Audio Coding,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1997, pp. 327-330.
    [15]P. Philippe, F. Moreau de Saint-Martin, and L. Mainard, “On The Choice of Wavelet Filters for Audio Compression,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1995, pp. 1045-1048.
    [16]P. E. Kudumakis and M. B. Sandler, “On The Performance of Wavelets for Low Bit Rate Coding of Audio Signals,” in Proc. Int. Conf. Acoust., Speech, Signal Process. 1995, pp. 3087-3090.
    [17]I. Daubechies, "Ten Lectures on Wavelets," no. 61 in CBMS-NSF Series in Applied Mathematics, SIAM, Philadelphia, 1992.
    [18]R. R. Coifman and M. V. Wickerhauser, "Entropy-based algorithms for best basis selection," IEEE Trans. Information Theory, vol. 38, pp. 713-718, March, 1992.
    [19]J. M. Shapiro, "Embedded Image Coding Using Zerotrees of Wavelet Coefficients," IEEE Trans. Signal Processing, Spec. Issue Wavelets Signal Processing, vol. 41, pp. 3445-3462, Dec. 1993.
    [20]Rissanen, J., Langdon, G.G. “Arithmetic Coding”, IBM J. Res. Dev. 23,2, Mar. 1979, pp149-162
    [21]Ian H.Witten, Radford M. Neal, John G. Cleary, “Arithmetic Coding For Data Compressoni”, Communication of the ACM, Vol. 30, No 6, June 1987, pp 520-540
    [22]C Todd, “A Digital Audio System for Broadcast and Prerecorded Media”, in Proc. 75th Conv. Aud. Eng. Soc., preprint #, Mar. 1984.
    [23]E.F. Schroder and W. Voessing, “High Quality Digital Audio Encoding With 3.0 Bits/Sample Using Adaptive Transform Coding”, in Proc. 80th Conv. Aud. Eng. Soc., preprint #2321, Mar. 1986.
    [24]G. Theile, et al., “Low-Bit Rate Coding of High Quality Audio Signals”, in Proc. 82nd Conv. Aud. Eng. Soc., preprint #2423, Mar. 1987.
    [25]K. Brandenburg, “OCF — A New Coding Algorithm for High Quality Sound Signals”, in Proc. ICASSP-87, May 1987, pp. 5.1.1-5.1.4.
    [26]J. Johnston, “Transform Coding of Audio Signals Using Perceptual Noise Criteria”, IEEE J. Sel. Areas in Comm., Feb. 1988, pp. 314-23.
    [27]W-Y Chan and A. Gersho, “High Fidelity Audio Transform Coding With Vector Quantization”, in Proc. ICASSP-90, May 1990, pp. 1109-1112.
    [28]K. Brandenburg and J.D. Johnston, “Second Generation Perceptual Audio Coding: The Hybrid Coder”, in Proc. 88th Conv. Aud. Eng. Soc., preprint #2937, Mar. 1990.
    [29]K. Brandenburg, et al, “Adaptive Spectral Entropy Coding of High Quality Music Signals”, in Proc. 90th Conv. Aud. Eng. Soc., preprint #3011, Feb. 1991.
    [30]Y.F. Dehery, et al, “A MUSICAM Source Codec for Digital Audio Broadcasting and Storage”, in proc. ICASSP-91, May 1991, pp. 3605-3608.
    [31]M. Iwadare, et al., “A 128 kb/s Hi-Fi Audio CODEC Based on Adaptive Transform Coding With Adaptiv Block Size MDCT”, IEEE J. Sel. Areas in Comm., Jan. 1992, pp. 138-144.
    [32]E. Zwicker and H. Fastl, Psychoacoustics, Facts and Models (Springer, Berlin, Heidelberg, 1990).
    [33]D. Y. Pan, “A Tutorial on MPEG/Audio Compression,” IEEE Multimedia pp. 60-74, 1995.
    [34]ISO/IEC 13818-3 : “Information technology — Generic Coding of Moving Pictures and Associated Audio , Part 3: Audio".1994 (“MPEG-2”).
    [35]C. Todd, et. Al., “AC-3 : Flexible Perceptual Coding for Audio Transmission and Storage”, in Proc. 96th Conv. Aud. Eng. Soc., preprint #3796, Feb. 1994
    [36]United States Advanced Television Systems Committee (ATSC), Audio Specialist Group (T3/S7) Doc. A/52, “Digital Audio Compression Standard (AC-3)”, Nov. 1994.
    [37] Martin Isenburg, “Transmission of Multimedia Data Over Lossy Networks”, Internaltional Computer Science Institute at Berkeley, USA, Aug. 1996.

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