資料隱藏應用於零樹小波分頻壓縮系統之音訊封包遺失回復技術

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連登科(Teng-Ko Lien ) 查詢紙本館藏

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電機工程研究所

論文名稱

資料隱藏應用於零樹小波分頻壓縮系統之音訊封包遺失回復技術
(Packet Loss Recovery Using Data Hiding for Embedded Zero Tree Wavelet Packet Audio Coding System)

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摘要(中)

隨著網路的快速發展，即時性多媒體的應用也愈加廣泛。對於網際網路多媒體即時資料的應用，封包的遺失是一個嚴重的問題，因此需要特別地對即時性多媒體訊號封包作適當的保護處理，降低封包遺失對播放品質的影響。本論文發展資料隱藏應用於零樹小波分頻壓縮系統之音訊封包遺失回復技術，使得重建樂音在不同的封包遺失情形下，皆能有最佳之樂音品質。
本論文以Embedded Zero-tree Wavelet Packet (EZWP) 作為樂音壓縮系統，其以小波封包(Wavelet Packet)分頻方式，結合人耳聲學模型與零樹編碼。利用每個音框開始的前幾次掃描係數當作重要部分，最後一次掃描的更精細值當作不重要部分。基本的想法是將音框資料的重要部分，隱藏於其他音框不重要的部分，並發展區塊資料隱藏法，以降低資料隱藏對於壓縮音質的影響。當封包發生遺失時，可以在解碼端萃取隱藏資料，回復遺失封包的樂音波形。利用EZW 漸進式編碼的特性，非常適合用於資料隱藏，應用於遺失封包恢復技術並不會增加傳輸位元率，並相容位元流語法，且不會嚴重降低原始之音訊封包品質。實驗結果顯示在封包遺失率達到 8% 的情況下，平均有效提升約1.2dB SSNR值; 客觀的聽覺測試的結果，亦可證明論文提出的方法能有效改善封包遺失所造成的影響。

摘要(英)

Multimedia transmission over Internet is getting popular and increasingly important. Because some networks do not provide quality of service (QoS), the received packet may get lost. As a result, audio quality is often degraded due to packet loss. In this thesis, we propose a packet loss recovery technique using data hiding method applied to Embedded Zero-tree Wavelet Packet (EZWP) audio coding system. The EZWP audio coding system uses wavelet packet decomposition and embedded zero-tree coding based on the psychoacoustic model. We choose several beginning scans of EZW coefficients in each frame as the most significant data, and choose the value list in the last scan as the least significant data. The basic idea is to hide the most significant data into the least significant data. In order to reduce the damage to audio quality, we develop block data hiding method. When packets are lost, we can extract the hided data to reconstruct the most important part of lost audio signals. The hided bitstreams are compatible to the encoding syntax. The data hiding method will not increase bitrate but only slightly degrade the reconstructed audio quality. In simulations, the proposed recovery strategy is able to improve SSNR by 1.2 dB on average. The subjective evaluation tests show that the proposed recovery provides smooth audio signals even at a loss rate of 8%.

關鍵字(中)

★ 小波分頻
★ 資料隱藏
★ 遺失封包回復
★ 零樹編碼
★ 音訊編碼

關鍵字(英)

★ audio coding
★ data hinding
★ packet loss recovery
★ wavelet packet
★ zero tree coding

論文目次

摘要I
AbstractII
目錄III
圖目VI
表目XI
第一章緒論1
1.1簡介1
1.2研究動機與目的3
1.3系統架構4
1.4論文架構5
第二章網際網路環境簡介與數位浮水印7
2.1網際網路環境7
2.2傳輸的問題9
2.3封包遺失的解決方法11
2.3.1傳送端封包遺失的解決方法12
2.3.1接收端封包遺失的解決方法14
2.4數位浮水印15
第三章音訊編碼技術21
3.1音訊壓縮編解碼器結構23
3.2人耳聲學模型24
3.2.1基本原理與其應用24
3.2.2雜訊對單頻音的遮蔽效應26
3.2.3頻音對單頻音的遮蔽效應30
3.2.4時間軸上的遮蔽效應31
3.2.5模型公式32
3.2.6訊號之各頻帶的最小遮蔽臨界值36
3.3 音訊編碼器家族39
3.3.1 MPEG音訊編碼器家族39
3.3.2杜比 ( Dolby ) AC 3音訊編碼器42
3.4 MPEG 1 Audio 聲學模型44
3.5 小波轉換與零樹編碼49
3.6零樹搜尋法則54
3.7漸進式與交錯式的碼傳送方式58
3.8連續近似量化60
第四章資料隱藏應用於遺失封包回復技術63
4.1 資料隱藏法65
4.1.1隱藏資料位元位置分析65
4.1.2 Value list資料隱藏法67
4.1.3 Value list區塊資料隱藏法69
4.2 保護資料分析72
4.2.1單層封包遺失回復72
4.2.2小波係數平滑化77
4.2.3多層封包遺失回復78
4.3 資料隱藏代價評估82
第五章實驗結果及討論85
5.1單層封包遺失回復品質評估85
5.1.1單層封包遺失回復客觀品質評估86
5.1.2單層封包遺失回復主觀品質評估92
5.2多層封包遺失回復品質評估94
5.2.1單層封包遺失回復客觀品質評估94
5.2.2多層封包遺失回復主觀品質評估110
第六章結論112
參考文獻113

參考文獻

[1]Paxson, V., "Measurements and Analysis of End-to-End Internet Dynamics," PhD thesis, Computer Science Division, University of California, Berkeley, April 1997.
[2]IVS "INRIA Videoconferencing System"
http://www.inria.fr/rodeo/ivs.html
[3]RAT "Robust Audio Tool"
http://www.cs.ucl.ac.uk/mice/rat/index.html
[4]VIC "Video Conferencing Tool"
http://www-nrg.ee.lbl.gov/vic/
[5]Braden, R., Clark, D., and Shenker, S., "Integrated services in the internet architecture: an overview," Request for Comments (Informational) RFC 1633, Internet Engineering Task Force, June 1994.
[6]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 (MPEG-1) ," 1992.
[7]James, C., Mckinney, Robert Hopkins, "Digital Audio Compression Standard (AC3) ," 1994.
[8]Sablatash, M., and Cooklev, T., "Compression of High-Quality Audio Signals, Including Recent Methods Using Wavelet Packets," Digital Signal Processing, vol. 6, no. 10, 1996, Page(s). 96-107.
[9]Karelic, Y., and Malah, D., "Compression of High-Quality Audio Signals Using Adaptive Filterbanks and A Zero-Tree Coder," Electrical and Electronics Engineers in Israel, 1995.
[10]Srinivasan, P., and Jamieson, L. H., "High-Quality Audio Compression Using an Adaptive Wavelet Packet Decomposition and Psychoacoustic Modeling," IEEE Trans. on Signal Processing, vol. 46, no. 4, April 1998, Page(s). 1085-1093.
[11]Chou, M.-T., "Audio Compression Using Wavelet Packets And A Zero-Tree Coder With Psychoacoustic Modeling," Mater thesis, National Central University, Taiwan, 1999.
[12]Lin, R.-S., "Scalable Audio Compression Using Wavelet Packets Decomposition and Embedded Zero Tree Coding," Mater thesis, National Central University, Taiwan, 2000.
[13]Martin Isenburg, "Transmission of Multimedia Data Over Lossy Networks," International Computer Science Institute at Berkeley, USA, Aug. 1996.
[14]Bolot, J., Crepin, H., and Vega Garcia, A., "Analysis of Audio Packet Loss in the Internet," Proc. 1995 Workshop on Network and Operating System Supageort for Audio and Video, 1995, Page(s). 163-174.
[15]Perkins, C., Hodson, O., Hardman, V., "A survey of packet loss recovery techniques for streaming audio," IEEE Network, vol. 12 5, Sept.-Oct. 1998, Page(s). 40-48.
[16]Hardman, V., Sasse, M.A., Handley, M., Watson, A., "Reliable audio for use over the Internet," INET'95 conference.
[17]Carle, G., Biersack, E.W., "Survey of error recovery techniques for IP-based audio-visual multicast apagelications," IEEE Network, vol. 11 6, Nov.-Dec. 1997, Page(s). 24-36.
[18]http://www-sop.inria.fr/rodeo/personnel/Thierry.Turletti/audio/cost.html
[19]Chua, R., "The ICSI audio laboratory," International Computer Science Institute, May 1996.
[20]Fingscheidt, T., Vary, P., and Andonegui, J.A., "Robust speech decoding: can error concealment be better than error correction," Speech and Signal Processing, 1998. Proceedings of the 1998 IEEE International Conference, vol. 1, May 1998, Page(s). 373-376.
[21]Miller, P.T., and Licklider, J. C. R., "The intelligibility of interrupted speech," j. Acoust. Soc. Amer, vol 22, no 2, 1950, Page(s). 167-73.
[22]Boney, L., Tewfik A., and Hamdy, K., "Digital watermarks for audio signals," Processdings of the 1996 international conference on multimedia computing and systems.
[23]Tilki, J.F., and Beex, A.A., "Encoding a hidden auxiliary channel onto a digital audio signal using psychoacoustic masking," engineering new century , IEEE 1997.
[24]Wang, Ye, "A new watermarking method of digital audio content for copyright protection," proceedings of ICSP’98 fourth international conference on signal processing, 1998.
[25]Todd, C, "A Digital Audio System for Broadcast and Prerecorded Media," Proc. 75th Conv. Aud. Eng. Soc., Mar. 1984.
[26]Schroder, E.F., and Voessing, W., "High Quality Digital Audio Encoding With 3.0 Bits/Sample Using Adaptive Transform Coding," Proc. 80th Conv. Aud. Eng. Soc., preprint #2321, Mar. 1986.
[27]Theile, G., "Low-Bit Rate Coding of High Quality Audio Signals," Proc. 82nd Conv. Aud. Eng. Soc., preprint #2423, Mar. 1987.
[28]Brandenburg, K., "OCF - A New Coding Algorithm for High Quality Sound Signals," Proc. ICASSP-87, May 1987, Page(s). 5.1.1-5.1.4.
[29]Johnston, J., "Transform Coding of Audio Signals Using Perceptual Noise Criteria," IEEE J. Sel. Areas in Comm., Feb. 1988, Page(s). 314-23.
[30]W.-Y., Chan, and Gersho, A., "High Fidelity Audio Transform Coding With Vector Quantization," Proc. ICASSP-90, May 1990, Page(s). 1109-1112.
[31]Brandenburg, K., and Johnston, J.D., "Second Generation Perceptual Audio Coding: The Hybrid Coder," Proc. 88th Conv. Aud. Eng. Soc., preprint #2937, Mar. 1990.
[32]Brandenburg, K, et al, "Adaptive Spectral Entropy Coding of High Quality Music Signals," Proc. 90th Conv. Aud. Eng. Soc., preprint #3011, Feb. 1991.
[33]Dehery, Y.F., et al, "A MUSICAM Source Codec for Digital Audio Broadcasting and Storage," proc. ICASSP-91, May 1991, Page(s). 3605-3608.
[34]Iwadare, M., et al., "A 128 kb/s Hi-Fi Audio CODEC Based on Adaptive Transform Coding With Adaptive Block Size MDCT," IEEE J. Sel. Areas in Comm, Jan. 1992. , Page(s). 138-144.
[35]Zwicker, E., and Fastl, H., Psychoacoustics Facts and Models, Springer, Berlin, Heidelberg, 1990.
[36]Pan, D.Y., "A Tutorial on MPEG/Audio Compression," IEEE Multimedia, 1995, Page(s). 60-74.
[37]ISO/IEC 13818-3 : "Information technology - Generic Coding of Moving Pictures and Associated Audio , Part 3: MPEG-2 BC," 1994.
[38]ISO/IEC 13818-7 : "Information Technology - Generic Coding of Moving Pictures and Associated Audio, Part 7: MPEG-2 AAC," 1997.
[39]Shlien, S., "Guide to MPEG-1 audio standard," Broadcasting, IEEE Transactions, vol. 40 4, Dec. 1994, Page(s). 206-218.
[40]http://www.cl.cam.ac.uk/~fapage2/software/mpeg/
[41]Shapiro, J.M., "Embedded Image Coding Using Zerotrees of Wavelet Coefficients," IEEE Trans. Signal Processing, Spec. Issue Wavelets Signal Processing, vol. 41, Dec. 1993, Page(s). 3445-3462.
[42]Bolot, J.C., Vega-Garcia, A., "Control mechanisms for packet audio in the Internet," INFOCOM '96. Fifteenth Annual Joint Conference of the IEEE Computer Societies. Networking the Next Generation, Proceedings IEEE, vol. 1, 1996, Page(s). 232-239.

指導教授

張寶基(Pao-Chi Chang)

審核日期

2001-6-14

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