本論文之目的旨在利用正弦激勵模式配合聲響心理學相關知識，開發一種能兼顧高音質與高壓縮比的2.4kb/s語音編碼技術，並針對巴克頻譜參數進行強健性向量量化設計以對抗通道雜訊。在語音壓縮部分，我們參考人耳在不同頻率與不同響度下之非線性響度聽覺響應，利用巴克頻譜進行正弦諧波振幅參數編碼。至於解碼部分，由於諧波個數會隨著音高變化，諧波振幅與巴克頻譜間並不存在唯一對映關係。為解決此問題，我們以知覺線性預估模型建立頻譜包絡線並由巴克頻譜求得其模型參數，進而還原取得諧波振幅。為加強相位的精確預估，本論文也提出非因果全極相位模型，經實驗證實能更有效模擬聲道相位。有關巴克頻譜的強健性向量量化設計，碼書的索引指定可有效的改善通道雜訊的干擾現象，但過去研究皆只局限在非記憶性二元對稱通道。本論文首先建立有限狀態馬可夫隨機過程之記憶性通道模型在碼書索引指定問題上的數學推導，再參考基因法則建立一適用於函數最佳化的隨機搜尋演算機制，以供雜訊干擾時設計通道錯誤模型與向量量化碼書索引指定之用。實驗結果顯示提出的正弦波參數模型可以改善合成聲音品質，而且其巴克頻譜參數碼書可以有效對抗通道雜訊。

This study focuses on two issues: perceptual enhancement of sinusoidal transform coding (STC) and optimal index assignment of vector quantization (VQ), to design a 2.4 kb/s speech coder that achieves high robustness against channel errors. STC attempts to model speech waveform as the sum of sinusoids whose frequencies, amplitudes, and phases are chosen to make the reconstruction a best fit to the original speech. The first part of this study focuses on quality enhancement of STC via the development of new parametric models. The benefits of the Bark spectrum are explored for use in the design of perceptual coding of the sine-wave amplitudes. In comparison to existing STC based on cepstral representation, the Bark-based amplitude coder is preferred because of its ability to achieve a uniform perceptual fit across the spectrum. One enhancement that further improves phase accuracy is the use of a noncausal all-pole vocal system that better matched the maximum-phase nature of differentiated glottal pulses. The next step of the present investigation was concerned with transmission of vector-quantized Bark spectrum over a noisy channel. We formulated the channel-robust VQ design as a combinatorial optimization problem leading to a search for the minimum distortion index assignment. To better track the statistical dependencies between error sequences, we propose to incorporate Markov characterization of the channel into the VQ design. Simulation results indicate that the global explorative properties of genetic algorithms make them very effective at finding the optimal index assignment and by using this index assignment the vector quantizer can be developed to respond to various channel conditions.

Cover
Abstract (in Chinese)
Abstract (in English)
Acknowledgements
Contents
List of Tables
List of Figures
1 Introduction
2 Sinusoidal Representation of Speech
2.1 The Sinusoidal Transform System
2.2 Models for Sine-Wave Parameters
2.3 Adaptive Postfilter
3 Quality Enhancement of Sinusoidal Transform Vocoders
3.1 Noncausall All-Pole Modeling of Vocal System
3.2 Perceptual Coding of Sine-Wave Amplitudes
3.3 Results and Discussions
4 Robust Vector Quantization of Bark Spectrum
4.1 Vector Quantization
4.2 Markov Characterization of Error Sequences
4.3 Minimum Distortion Index Assignment
4.4 Results and Discussions
5 Conclusions
Bibliograph
Vita
Publication List

[1] R. J. McAulay and T. F. Quatieri. "Low rate speech coding based on a sinusoidal model". In S. Furui and M. M. Sondhi, editors, Advances in Speech Signal Processing, chapter 6. Marcel Dekker, New York, 1992.
[2] E. Singer, R. J. McAulay, R. B. Dunn, and T. F. Quatieri. "Low rate coding of the spectral envelope using channel gains". In Proc. IEEE Int. Conf. Acoust., Speech and Signal Proc., pages 769-772, 1995.
[3] R. J. McAulay and T. F. Quatieri. "Sinusoidal coding". In W. B. Kleijn and K. K. Paliwal, editors, Speech Coding and Synthesis, chapter 4. Elsevier, Amsterdam, 1995.
[4] R. J. McAulay and T. F. Quatieri. "Magnitude-only reconstruction using a sinusoidal speech model". In Int. Conf. Acoust., Speech and Signal Proc., pages 27.6.1-27.6.4, San Diego, GA, March 1984.
[5] R. J. McAulay and T. F. Quatieri. "Sine-wave phase coding at low data rates". In Int. Conf. Acoust., Speech and Signal Proc., pages 577-580, 1991.
[6] R. J. McAulay, T. M. Parks, T. F. Quatieri, and M. Sabin. "Sine-wave amplitude coding at low data rates". In B. S. Atal, V. Cuperman, and A. Gersho, editors, Advances in Speech Coding. Kluwer Academic Publishers, Boston/Dordrecht/London, 1991.
[7] R. J. McAulay and T. F. Quatieri. "Pitch estimation and voicing detection based on a sinusoidal model". In Proc. IEEE Int. Conf. Acoust., Speech and Signal Proc., pages 249-252, Albuquerque, NM, Apr. 3-6 1990.
[8] R. J. McAulay and T. F. Quatieri. "The application of subband coding to improve quality and robustness of the sinusoidal transform coder". In Int. Conf. Acoust., Speech and Signal Proc., pages 439-442, 1993.
[9] R. J. McAulay and T. Champion. "Improved interoperable 2.4 kb/s LPC using sinusoidal transform coder techniques". In Proc. IEEE Int. Conf. Acoust., Speech and Signal Proc., pages 641-643, 1990.
[10] R. J. McAulay and T. F. Quatieri. "Multirate sinusoidal transform coding at rates from 2.4 kbps to 8 kbps". In Int. Conf. Acoust., Speech and Signal Proc., pages 38.7.1-38.7.4, 1987.
[11] R. J. McAulay and T. F. Quatieri. "The sinusoidal transform coder at 2400 b/s". In IEEE Military Communications Conf., pages 378-380, 1992.
[12] T. F. Quatieri and R. J. McAulay. "Phase coherence in speech reconstruction for enhancement and coding applications". In Int. Conf. Acoust., Speech and Signal Proc., pages 207-210, 1989.
[13] R. J. McAulay and T. F. Quatieri. "Mid-rate coding based on a sinusoidal representation of speech". In Int. Conf. Acoust., Speech and Signal Proc., pages 25.3.1-25.3.4, 1985.
[14] J. S. Marques, L. B. Almeida, and J. M. Tribolet. "Harmonic coding at 4.8 kb/s". In Int. Conf. Acoust., Speech and Signal Proc., pages 17-20, 1990.
[15] S. Ahmadi and A. S. Spanias. "A new sinusoidal phase modeling algorithm". In Int. Conf. Acoust., Speech and Signal Proc., pages 1675-1678, 1997.
[16] S. Ahmadi and A. S. Spnias. "A new phase model for sinusoidal transform coding of speech". IEEE Trans. on Speech and Audio Processing, 6(5):495-501, Sep. 1998.
[17] T. G. Champion, R. J. McAulay, and T. F. Quatieri. "High-order allpole modelling of the spectral envelope". In Int. Conf. Acoust., Speech and Signal Proc., pages I-529-I-532, 1994.
[18] E. Zwicker and H. Fastl. Psychoacoustics. Springer-Verlag, Berlin, 1990.
[19] S. Wang, A. Sekey, and A. Gersho. "An objective measure for predicting subjective quality of speech coders". IEEE J. Select. Areas Commun., 10(5):819-829, June 1992.
[20] V. Oppenheim and R. W. Schafer. Discrete-Time Signal Processing. Prentice Hall, Englewood Cliffs, NJ, 1989.
[21] X. Sun, F. Plante, B. M. Cheetham, and K. W. Wong. "Spectral envelope and phase optimisation for sinusoidal speech coding". In Proc. IEEE Workshop on Speech Coding for Telecommunications., pages 75-76, Annapolis, USA, Sept. 1995.
[22] X. Sun, F. Plante, B. M. Cheetham, and K. W. Wong. "Phase modeling of speech excitation for low bit-rate sinusoidal transform coding". In Int. Conf. Acoust., Speech and Signal Proc., pages 1691-1694, 1997.
[23] W. R. Gardner and B. D. Rao. "Noncausal all-pole modeling of voiced speech". IEEE Trans. Speech and Audio Processing, 5(1):1-10, 1997.
[24] E. Rosenberg. "Effect of glottal pulse shape on the quality of natural vowels". J. Acoust. Soc. Amer., 49(2):583-590, 1971.
[25] R. M. Gray and D. L. Neuhoff. "Quantization". IEEE trans. on Inform. Theory, 44(6):2325-2383, Oct. 1998.
[26] R. M. Gray. "Vector quantization". IEEE Acoust. Speech Signal Processing Mag., 1:4-29, Apr. 1984.
[27] J. R. B. De Marca and N. S. Jayant. "An algorithm for assigning binary indices to the codevectors of a multidimensional quantizer". In Proc. IEEE Int. Comm. Conf., pages 1128-1132, Seattle, WA, June 1987.
[28] Mehes and K. Zeger. "Affine index assignment for binary lattice quantization with channel noise". In IEEE International Symposium on Information Theory, page 377, 1995.
[29] K. A. Zeger and A. Gersho. "Vector quantizer design for memoryless noisy channels". In IEEE International Conference on Communications, pages 1593-1597, 1988.
[30] N. T. Cheng and N. K. Kingsbury. "Robust zero-redundancy vector quantization for noisy channels". In IEEE International Conference on Communications, pages 1338-1342, June 1989.
[31] K. Zeger and A. Gersho. "Pseudo-gray coding". IEEE Trans. on Communications, 38(12):2147-2158, December 1990.
[32] N. Farvardin. "A study of vector quantization for noisy channels". IEEE Trans. on Inform. Theory, 36(4):799-809, July 1990.
[33] M. R. Soleymani and S. D. Morgera. "Image coding for noisy channels". In Int. Conf. Acoust., Speech and Signal Proc., pages 2785-2788, 1991.
[34] P. Knagenhjelm. "How good is your index assignment". In Int. Conf. Acoust., Speech and Signal Proc., pages II-423-II-426, 1993.
[35] K. Kuo, C. H. Lin, and J. H. Leu. "Noise reduction and concealment for block encoded images". IEEE trans. on Consumer Electronics Letters, 40(3):514-520, Aug. 1994.
[36] E. A. Riskin, R. Ladner, R. Y. Wang, and L. E. Atlas. "Index assignment for progressive transmission of full-search vector quantization". IEEE Trans. on Image processing, 3(3):307-312, May 1994.
[37] S. W. McLaughlin, D. L. Neuhoff, and J. J. Ashley. "Optimal binary index assignment for a class of equiprobable scalar and vector quantizers". IEEE Trans. on Inform. Theory, 41(6):2031-2037, Nov. 1995.
[38] S. Gadkari and K. Rose. "Robust vector quantization by transmission energy allocation". Electronics Letters, 32(16):1451-1453, Aug. 1996.
[39] J. S. Pan, F. R. McInnes, and M. A. Jack. "Application of parallel genetic algorithm and property of multiple global optima to VQ codevector index assignment for noisy channels". Electronics Letters, 32(4):296-297, Feb. 1996.
[40] J. S. Pan and S. C. Chu. "Non-redundant VQ channel coding using tabu search strategy". Electronics Letters, 32(17):1545-1546, August 1996.
[41] J. S. Pan, F. R. McInnes, and M. A. Jack. "VQ codevector index assignment using genetic algorithms for noisy channels". In Proc. of Fourth Int. Conf. on Spoken Language Processing. ICSLP''96, pages 295-298, Philadelphia, PA, USA, Oct. 3-6 1996.
[42] P. Knagenhjelm and E. Agrell. "The hadamard transform - a tool for index assignment". IEEE Trans. Inform. Theory, 42(4):1139-1151, July 1996.
[43] S. Leung and L. W. Chan. "Transmission of vector quantized data over a noisy channel". IEEE Trans. Neural Networks, 8(3):582-589, May 1997.
[44] S. Leung and L. W. Chan. "An error control scheme for transmission of vector quantization data over noisy channels". IEEE Trans. Signal Processing, 46(10):2767-2780, October 1998.
[45] P. Hedelin, P. Knagenhjelm, and M. Skoglund. "Vector quantization for speech transmission". In W. B. Kleijn and K. K. Paliwal, editors, Speech Coding and Synthesis. Amsterdam, The Netherlands: Elsevier, 1995.
[46] P. Hedelin, P. Knagenhjelm, and M. Skoglund. "Theory for transmission of vector quantization data". In W. B. Kleijn and K. K. Paliwal, editors, Speech Coding and Synthesis. Amsterdam, The Netherlands: Elsevier, 1995.
[47] W. Turin and M. Sondhi. "Modeling error source in digital channels". IEEE J. Select. Areas Commun., 11(3):340-347, April 1993.
[48] N. Gilbert. "Capacity of a burst-noise channel". The Bell System Technical Journal, 39:1253-1265, Sept. 1960.
[49] D. Fritchman. "A binary channel characterization using partitioned Markov chains". IEEE Trans. Inform. Theory, IT-13(2):221-227, Apr. 1967.
[50] L. R. Rabiner and B. H. Juang. "An introduction to hidden markov models". IEEE ASSP Magazine, pages 4-16, January 1986.
[51] E. Goldberg. Genetic Algorithm in Search, Optimization and Machine Learning. Addison-Wesley, New York, 1989.
[52] J. H. Holland. Adaptation in Natural and Artificial Systems. University of Michigan Press, Ann Arbor MI, 1975.
[53] Z. Michalewicz. Genetic Algorithms $+$ Data Structures $=$ Evolution programs. Berlin Spring-Verlag, New York, 1994.
[54] R. J. McAulay and T. F. Quatieri. "Speech analysis-synthesis based on a sinusoidal representation". IEEE Trans. Acoust., Speech and Signal Proc., ASSP-34(4):744-754, 1986.
[55] T. F. Quatieri and R. J. McAulay. "Speech transformations based on a sinusoidal representation". IEEE Trans. Acoust., Speech and Signal Proc., ASSP-34(6):1449-1464, 1986.
[56] T. F. Quatieri and R. J. McAulay. "Speech transformations based on a sinusoidal representation". In Int. Conf. Acoust., Speech and Signal Proc., pages 489-490, 1985.
[57] T. F. Quatieri and R. J. McAulay. "Shape invariant time-scale and pitch modification of speech". IEEE Trans. Signal Proc., 40(3):497-510, March 1986.
[58] P. Neuburg. "Simple pitch-dependent algorithm for high-quality speech rate changing". J. Acoust. Soc. Amer., 63(2):624-625, Feb. 1971.
[59] S. Roucos and A. M. Wilgus. "High quality time-scale modification for speech". In Int. Conf. Acoust., Speech and Signal Proc., pages 493-496, Tampa, FL, March 1985.
[60] M. R. Portnoff. "Time-scale modification of speech based on short-time fourier analysis". IEEE Trans. Acoust., Speech and Signal Proc., ASSP-29(3):374-390, June 1981.
[61] J. L. Wayman and D. L. Wilson. "Some improvements on the synchronized-overlap-add method of time scale modification for use in real-time speech compression and noise filtering". IEEE Trans. Acoust., Speech and Signal Proc., ASSP-36(1):139-140, Jan. 1988.
[62] Fairbanks, W. L. Everitt, and R. P. Jaeger. "Method for time or frequency compression-expansion of speech". IRE Trans. Professional Group on Audio, AU-2:7-12, Jan.-Feb. 1954.
[63] J. Makhoul and A. El-Jaroudi. "Time-scale modification in medeum to low rate speech coding". In Int. Conf. Acoust., Speech and Signal Proc., pages 170-1708, Tokyo, 1986.
[64] T. E. Tremain. "The government standard linear predictive codng algorithm: LPC10". Speech Technology, pages 40-49, April 1982.
[65] S. Y. Kwon and A. J. Goldberg. "An enhanced LPC vocoder with no voiced/unvoiced switch". IEEE Trans. on Acoustics, Speech and Signal Processing, ASSP-32(4):851-858, 1984.
[66] V. McCree. "A new LPC vocoder model for low bit rate speech coding". PhD thesis, Georgia Inst. Technol., Atlanta, GA, Aug. 1992.
[67] V. McCree and T. P. Barnwell III. "A mixed excitation LPC vocoder model for low bit rate speech coding". IEEE Trans. on Speech and Audio Processing, 3(4):242-250, July 1995.
[68] V. McCree, K. Truong, E. B. George, T. P. Barnwell, and V. Viswanathan. "A 2.4 kbit/s MELP coder candidate for the new U. S. federal standard". In Int. Conf. Acoust., Speech and Signal Proc., pages 200-203, 1996.
[69] M. A. Kohler. "A comparison of the new 2400 bps melp federal standard with other standard coders". In Int. Conf. Acoust., Speech and Signal Proc., pages 1587-1590, 1997.
[70] L. M. Supplee, R. P. Chon, and J. S. Collura. "MELP: the new federal standard at 2400 bps". In Int. Conf. Acoust., Speech and Signal Proc., pages 1591-1594, 1997.
[71] J. P. Campell, T. E. Tremain, and V. C. Welch. "The federal standard 1016 4800 bps CELP voice coder". Digital Signal Processing, 1(3):145-155, 1989.
[72] J. P. Campell, T. E. Tremain, and V. C. Welch. "The federal standard 1016 4800 bps voice coder: CELP". Speech Technology, pages 58-63, Apr./May 1990.
[73] W. Griffin and J. S. Lim. "Multiband excitation vocoder. IEEE Trans. on Acoustics, Speech and Signal Processing", ASSP-36:1223-1235, Aug. 1988.
[74] M. S. Brandstein, J. C. Hardwick, and J. S. Lim. "The multi-band excitation speech coder". In B. S. Atal, V. Cuperman, and A. Gersho, editors, Advances in Speech Coding. Kluwer Academic Publishers, Boston/Dordrecht/London, 1991.
[75] R. J. McAulay and T. F. Quatieri. "Computationally efficient sine-wave synthesis and its application to sinusoidal transform coding". In Int. Conf. Acoust., Speech and Signal Proc., pages 370-373, 1988.
[76] B. George and M. J. T. Smith. "Speech analysis/synthesis and modification using an analysis-by-synthesis/overlap-add sinusoidal model". IEEE Trans. on Speech and Audio processing, 5(5):389-406, 1997.
[77] M. W. Macon and M. A. Clements. "Sinusoidal modeling and modification of unvoiced speech". IEEE Trans. on Speech and Audio Processing, 5(6):557-560, Nov. 1998.
[78] D. B. Paul. "The spectral envelope estimation vocoder". IEEE trans. on Acoustics, Speech and Signal Processing, ASSP-29:786-794, 1981.
[79] R. J. McAulay and T. F. Quatieri. "Phase modelling and its application to sinusoidal transform coding". In Int. Conf. Acoust., Speech and Signal Proc., pages 33.9.1-33.9.3, Tokyo, 1986.
[80] M. J. Sabin. "DPCM coding of spectral amplitudes without positive slope overload". IEEE Trans. on Signal Processing, 39(3):756-758, March 1993.
[81] L. R. Rabiner and R. W. Schafer. Digital Processing of Speech Signals. Prentice Hall, Englewoods Cliffs, 1978.
[82] Fourcin. "Speech processing by man and machine-Group report". In T. Bullock, editor, Recognition of Complex Acoustic Signal. Life Sciences Res. Rep. 5 of the Dahlem Workshops, Berlin, Germany, 1977.
[83] Sekey and B. Hanson. "Improved one-Bark bandwidth auditory filter". J. Acoust. Soc. Am., 75(6):1902-1904, June 1984.
[84] T. Watanabe and S. Hayashi. "An objective measure based on an auditory model for assessing low-rate coded speech". IEICE Trans. Inf. and Syst., E78-D(6):751-757, June 1995.
[85] R. A. W. Bladon. "Modeling the judgment of vowel quality differences". J. Acoust. Soc. Am., 69(5):1414-1422, May 1981.
[86] S. Itahashi and S. Yokoyama. "Automatic formant extraction ultilizing mel scale and equal loudness contour". In Int. Conf. Acoust., Speech and Signal Proc., pages 310-313, 1976.
[87] Hermansky. "Perceptual linear predictive (PLP) analysis of speech". J. Acoust. Soc. Am., 87(4):1738-1752, Apr. 1990.
[88] S. Furui. Digital Speech Processing, Synthesis, and Recognition. Marcel Dekker, New York and Basel, 1989.
[89] J. Makhoul. "Linear prediction: a tutorial review". Proc. IEEE, 63:561-580, 1975.
[90] IEEE69 "IEEE recommended practice for speech quality measurements". IEEE trans. Aud. Electroacoust., pages 227-246, Sept. 1969.
[91] W. Voiers. "Diagnostic acceptability measure for speech communication systems". In Proc. IEEE Int. Conf. Acoust., Speech, and Signal process, pages 204-207, May 1977.
[92] R. F. Kubichek. "Mel-cepstral distance measure for objective speech quality assessment". In Proc. IEEE Pacific Rim Conf. Commun., Computation, and Signal Proc., pages 125-128, 1993.
[93] N. Kitawaki, H. Nagabuchi, and K. Itoh. "Objective quality evaluation for low-bit-rate speech coding systems". IEEE J. Select. Areas Commun., 6(2):242-248, February 1988.
[94] Y. Linde, A. Buzo, and R. M. Gray. "An algorithm for vector quantizer design". IEEE Trans. Commun., COM-28:84-95, Jan. 1980.
[95] W. Turin. Performance Analysis of Digital Transmission Systems. Computer Science Press, New York, 1990.
[96] L. R. Rabiner. Fundamentals of Speech Recognition. Prentice Hall, Englewood Cliffs, New Jersey, 1993.
[97] S. Srinivas and K. S. Shanmugan. "Characterization of bursty channels using markov models". In IEEE International Conference on Communications: ICC''93, pages 1615 -1619, Geneva, 1993.
[98] J.N.L. Brummer. "Characterization of digital channels using hidden markov models". In IEEE Proceedings of the 1992 South African symposium on communications and signal processing: COMSIG''92, pages 183-188, Sept. 1992.
[99] T. Suematsu and H. Imai. "Estimation method for three state compound channel model". Electronics Letters, 29(1):96-98, Jan. 1993.
[100] J. Y. Chouinard, M. Lecours, and G. Y. Delisle. "Estimation of gilbert''s and fritchman''s models parameters using the gradient method for digital mobil radio channels". IEEE Trans. on Vehicular Technology, 37(3):158-166, August 1988.
[101] S. E. Levinson, L. R. Rabiner, and M. M. Sondhi. "An introduction to the application of the theory of probabilistic functions of a Markov process to automatic speech recognition". The Bell System Technical Journal, 62(4):1035-1074, part 1, Apr. 1983.
[102] M. Gen and R. Cheng. Genetic Algorithms and Engineering Design. John Wiley & Sons, New York, 1997.
[103] J. E. Baker. "Adaptive selection methods for genetic algorithms". In J. J. Grefenstette, editor, Proc. of the First International Conference on Genetic Algorithms and Their Applications, pages 101-111, Lawrence Erlbaum Associates, Hillsdale, NJ, 1985.
[104] J. E. Baker. "Reducing bias and inefficiency in the selection algorithm". In J. J. Grefenstette, editor, Proc. of the 2nd International Conference on Genetic Algorithms and Their Applications, pages 14-21, Lawrence Erlbaum Associates, Hillsdale, NJ, 1987.
[105] D. M. Tate and A. E. Smith. "Expected allele converage and the role of mutation in genetic algorithms". In Proc. of the First International Conference on Genetic Algorithms, pages 31-37, 1993.
[106] Y. Hussain and N. Farvardin. "Finite-state vector quantization over noisy channels and its application to lsp parameters". In Int. Conf. Acoust., Speech and Signal Proc., pages II-133-II-136, 1992. endthebibliography